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Developer Documentation
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<h1>
Developer Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Developers-Guide" href="#Developers-Guide">1 Developers Guide</a>
<ul class="no-bullet">
<li><a name="toc-Notes-for-external-developers" href="#Notes-for-external-developers">1.1 Notes for external developers</a></li>
<li><a name="toc-Contributing" href="#Contributing">1.2 Contributing</a></li>
<li><a name="toc-Coding-Rules-1" href="#Coding-Rules-1">1.3 Coding Rules</a>
<ul class="no-bullet">
<li><a name="toc-Code-formatting-conventions" href="#Code-formatting-conventions">1.3.1 Code formatting conventions</a></li>
<li><a name="toc-Comments" href="#Comments">1.3.2 Comments</a></li>
<li><a name="toc-C-language-features" href="#C-language-features">1.3.3 C language features</a></li>
<li><a name="toc-Naming-conventions" href="#Naming-conventions">1.3.4 Naming conventions</a></li>
<li><a name="toc-Miscellaneous-conventions" href="#Miscellaneous-conventions">1.3.5 Miscellaneous conventions</a></li>
<li><a name="toc-Editor-configuration" href="#Editor-configuration">1.3.6 Editor configuration</a></li>
</ul></li>
<li><a name="toc-Development-Policy" href="#Development-Policy">1.4 Development Policy</a></li>
<li><a name="toc-Submitting-patches-1" href="#Submitting-patches-1">1.5 Submitting patches</a></li>
<li><a name="toc-New-codecs-or-formats-checklist" href="#New-codecs-or-formats-checklist">1.6 New codecs or formats checklist</a></li>
<li><a name="toc-patch-submission-checklist" href="#patch-submission-checklist">1.7 patch submission checklist</a></li>
<li><a name="toc-Patch-review-process" href="#Patch-review-process">1.8 Patch review process</a></li>
<li><a name="toc-Regression-tests-1" href="#Regression-tests-1">1.9 Regression tests</a>
<ul class="no-bullet">
<li><a name="toc-Adding-files-to-the-fate_002dsuite-dataset" href="#Adding-files-to-the-fate_002dsuite-dataset">1.9.1 Adding files to the fate-suite dataset</a></li>
<li><a name="toc-Visualizing-Test-Coverage" href="#Visualizing-Test-Coverage">1.9.2 Visualizing Test Coverage</a></li>
<li><a name="toc-Using-Valgrind" href="#Using-Valgrind">1.9.3 Using Valgrind</a></li>
</ul></li>
<li><a name="toc-Release-process-1" href="#Release-process-1">1.10 Release process</a>
<ul class="no-bullet">
<li><a name="toc-Criteria-for-Point-Releases-1" href="#Criteria-for-Point-Releases-1">1.10.1 Criteria for Point Releases</a></li>
<li><a name="toc-Release-Checklist" href="#Release-Checklist">1.10.2 Release Checklist</a></li>
</ul></li>
</ul></li>
</ul>
</div>
<a name="Developers-Guide"></a>
<h2 class="chapter">1 Developers Guide<span class="pull-right"><a class="anchor hidden-xs" href="#Developers-Guide" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Developers-Guide" aria-hidden="true">TOC</a></span></h2>
<a name="Notes-for-external-developers"></a>
<h3 class="section">1.1 Notes for external developers<span class="pull-right"><a class="anchor hidden-xs" href="#Notes-for-external-developers" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Notes-for-external-developers" aria-hidden="true">TOC</a></span></h3>
<p>This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in <samp>doc/examples</samp> and in the source code to
see how the public API is employed.
</p>
<p>You can use the FFmpeg libraries in your commercial program, but you
are encouraged to <em>publish any patch you make</em>. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
</p>
<p>For more detailed legal information about the use of FFmpeg in
external programs read the <samp>LICENSE</samp> file in the source tree and
consult <a href="http://ffmpeg.org/legal.html">http://ffmpeg.org/legal.html</a>.
</p>
<a name="Contributing"></a>
<h3 class="section">1.2 Contributing<span class="pull-right"><a class="anchor hidden-xs" href="#Contributing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Contributing" aria-hidden="true">TOC</a></span></h3>
<p>There are 3 ways by which code gets into ffmpeg.
</p><ul>
<li> Submitting Patches to the main developer mailing list
see <a href="#Submitting-patches">Submitting patches</a> for details.
</li><li> Directly committing changes to the main tree.
</li><li> Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
</li></ul>
<p>Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the <a href="#Coding-Rules">Coding Rules</a>.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
</p>
<a name="Coding-Rules"></a><a name="Coding-Rules-1"></a>
<h3 class="section">1.3 Coding Rules<span class="pull-right"><a class="anchor hidden-xs" href="#Coding-Rules-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Coding-Rules-1" aria-hidden="true">TOC</a></span></h3>
<a name="Code-formatting-conventions"></a>
<h4 class="subsection">1.3.1 Code formatting conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Code-formatting-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code-formatting-conventions" aria-hidden="true">TOC</a></span></h4>
<p>There are the following guidelines regarding the indentation in files:
</p>
<ul>
<li> Indent size is 4.
</li><li> The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
</li><li> You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
</li></ul>
<p>The presentation is one inspired by &rsquo;indent -i4 -kr -nut&rsquo;.
</p>
<p>The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
</p>
<a name="Comments"></a>
<h4 class="subsection">1.3.2 Comments<span class="pull-right"><a class="anchor hidden-xs" href="#Comments" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Comments" aria-hidden="true">TOC</a></span></h4>
<p>Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
</p>
<p>Avoid Qt-style and similar Doxygen syntax with <code>!</code> in it, i.e. replace
<code>//!</code> with <code>///</code> and similar. Also @ syntax should be employed
for markup commands, i.e. use <code>@param</code> and not <code>\param</code>.
</p>
<div class="example">
<pre class="example">/**
* @file
* MPEG codec.
* @author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar {
int var1; /**&lt; var1 description */
int var2; ///&lt; var2 description
/** var3 description */
int var3;
} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @param my_parameter description of my_parameter
* @return return value description
*/
int myfunc(int my_parameter)
...
</pre></div>
<a name="C-language-features"></a>
<h4 class="subsection">1.3.3 C language features<span class="pull-right"><a class="anchor hidden-xs" href="#C-language-features" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-C-language-features" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
</p>
<ul>
<li> the &lsquo;<samp>inline</samp>&rsquo; keyword;
</li><li> &lsquo;<samp>//</samp>&rsquo; comments;
</li><li> designated struct initializers (&lsquo;<samp>struct s x = { .i = 17 };</samp>&rsquo;)
</li><li> compound literals (&lsquo;<samp>x = (struct s) { 17, 23 };</samp>&rsquo;)
</li></ul>
<p>These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
</p>
<p>All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
</p>
<ul>
<li> mixing statements and declarations;
</li><li> &lsquo;<samp>long long</samp>&rsquo; (use &lsquo;<samp>int64_t</samp>&rsquo; instead);
</li><li> &lsquo;<samp>__attribute__</samp>&rsquo; not protected by &lsquo;<samp>#ifdef __GNUC__</samp>&rsquo; or similar;
</li><li> GCC statement expressions (&lsquo;<samp>(x = ({ int y = 4; y; })</samp>&rsquo;).
</li></ul>
<a name="Naming-conventions"></a>
<h4 class="subsection">1.3.4 Naming conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Naming-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Naming-conventions" aria-hidden="true">TOC</a></span></h4>
<p>All names should be composed with underscores (_), not CamelCase. For example,
&lsquo;<samp>avfilter_get_video_buffer</samp>&rsquo; is an acceptable function name and
&lsquo;<samp>AVFilterGetVideo</samp>&rsquo; is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
</p>
<p>There are the following conventions for naming variables and functions:
</p>
<ul>
<li> For local variables no prefix is required.
</li><li> For file-scope variables and functions declared as <code>static</code>, no prefix
is required.
</li><li> For variables and functions visible outside of file scope, but only used
internally by a library, an <code>ff_</code> prefix should be used,
e.g. &lsquo;<samp>ff_w64_demuxer</samp>&rsquo;.
</li><li> For variables and functions visible outside of file scope, used internally
across multiple libraries, use <code>avpriv_</code> as prefix, for example,
&lsquo;<samp>avpriv_aac_parse_header</samp>&rsquo;.
</li><li> Each library has its own prefix for public symbols, in addition to the
commonly used <code>av_</code> (<code>avformat_</code> for libavformat,
<code>avcodec_</code> for libavcodec, <code>swr_</code> for libswresample, etc).
Check the existing code and choose names accordingly.
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
<code>lib&lt;name&gt;/lib&lt;name&gt;.v</code> files.
</li></ul>
<p>Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in <code>_t</code> are reserved by
<a href="http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02">POSIX</a>.
Also avoid names starting with <code>__</code> or <code>_</code> followed by an uppercase
letter as they are reserved by the C standard. Names starting with <code>_</code>
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with <code>_</code> altogether.
</p>
<a name="Miscellaneous-conventions"></a>
<h4 class="subsection">1.3.5 Miscellaneous conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Miscellaneous-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Miscellaneous-conventions" aria-hidden="true">TOC</a></span></h4>
<ul>
<li> fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
</li><li> Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don&rsquo;t make the code easier to understand.
</li></ul>
<a name="Editor-configuration"></a>
<h4 class="subsection">1.3.6 Editor configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Editor-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Editor-configuration" aria-hidden="true">TOC</a></span></h4>
<p>In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your <samp>.vimrc</samp>:
</p><div class="example">
<pre class="example">&quot; indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
&quot; Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
&quot; Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
&quot; Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@&lt;!$/
</pre></div>
<p>For Emacs, add these roughly equivalent lines to your <samp>.emacs.d/init.el</samp>:
</p><div class="example">
<pre class="example">(c-add-style &quot;ffmpeg&quot;
'(&quot;k&amp;r&quot;
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style &quot;ffmpeg&quot;)
</pre></div>
<a name="Development-Policy"></a>
<h3 class="section">1.4 Development Policy<span class="pull-right"><a class="anchor hidden-xs" href="#Development-Policy" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development-Policy" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Contributions should be licensed under the
<a href="http://www.gnu.org/licenses/lgpl-2.1.html">LGPL 2.1</a>,
including an &quot;or any later version&quot; clause, or, if you prefer
a gift-style license, the
<a href="http://opensource.org/licenses/isc-license.txt">ISC</a> or
<a href="http://mit-license.org/">MIT</a> license.
<a href="http://www.gnu.org/licenses/gpl-2.0.html">GPL 2</a> including
an &quot;or any later version&quot; clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
</li><li> You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers&rsquo;
work.
</li><li> The commit message should have a short first line in the form of
a &lsquo;<samp>topic: short description</samp>&rsquo; as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
</li><li> You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
</li><li> Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
</li><li> Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
<p>Note: Redundant code can be removed.
</p>
</li><li> Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
</li><li> We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
<p>NOTE: If you had to put if(){ .. } over a large (&gt; 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
</p>
</li><li> Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as &quot;fixed!&quot; or &quot;Changed it.&quot; are unacceptable.
Recommended format:
<div class="example">
<pre class="example">area changed: Short 1 line description
details describing what and why and giving references.
</pre></div>
</li><li> Make sure the author of the commit is set correctly. (see git commit &ndash;author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
</li><li> When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
</li><li> Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
</li><li> Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
</li><li> Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
</li><li> Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
</li><li> Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
</li><li> Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
</li><li> Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
</li><li> Make sure that no parts of the codebase that you maintain are missing from the
<samp>MAINTAINERS</samp> file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help
finding a new maintainer and also don&rsquo;t forget updating the <samp>MAINTAINERS</samp> file.
</li></ol>
<p>We think our rules are not too hard. If you have comments, contact us.
</p>
<a name="Submitting-patches"></a><a name="Submitting-patches-1"></a>
<h3 class="section">1.5 Submitting patches<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-patches-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-patches-1" aria-hidden="true">TOC</a></span></h3>
<p>First, read the <a href="#Coding-Rules">Coding Rules</a> above if you did not yet, in particular
the rules regarding patch submission.
</p>
<p>When you submit your patch, please use <code>git format-patch</code> or
<code>git send-email</code>. We cannot read other diffs :-)
</p>
<p>Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
</p>
<p>Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
</p>
<p>Run the <a href="#Regression-tests">Regression tests</a> before submitting a patch in order to verify
it does not cause unexpected problems.
</p>
<p>It also helps quite a bit if you tell us what the patch does (for example
&rsquo;replaces lrint by lrintf&rsquo;), and why (for example &rsquo;*BSD isn&rsquo;t C99 compliant
and has no lrint()&rsquo;)
</p>
<p>Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
</p>
<p>Patches should be posted to the
<a href="http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel">ffmpeg-devel</a>
mailing list. Use <code>git send-email</code> when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission.
</p>
<p>Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
</p>
<p>Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
</p>
<a name="New-codecs-or-formats-checklist"></a>
<h3 class="section">1.6 New codecs or formats checklist<span class="pull-right"><a class="anchor hidden-xs" href="#New-codecs-or-formats-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-New-codecs-or-formats-checklist" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Did you use av_cold for codec initialization and close functions?
</li><li> Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
</li><li> Did you bump the minor version number (and reset the micro version
number) in <samp>libavcodec/version.h</samp> or <samp>libavformat/version.h</samp>?
</li><li> Did you register it in <samp>allcodecs.c</samp> or <samp>allformats.c</samp>?
</li><li> Did you add the AVCodecID to <samp>avcodec.h</samp>?
When adding new codec IDs, also add an entry to the codec descriptor
list in <samp>libavcodec/codec_desc.c</samp>.
</li><li> If it has a FourCC, did you add it to <samp>libavformat/riff.c</samp>,
even if it is only a decoder?
</li><li> Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you&rsquo;re just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
</li><li> Did you add an entry to the table of supported formats or codecs in
<samp>doc/general.texi</samp>?
</li><li> Did you add an entry in the Changelog?
</li><li> If it depends on a parser or a library, did you add that dependency in
configure?
</li><li> Did you <code>git add</code> the appropriate files before committing?
</li><li> Did you make sure it compiles standalone, i.e. with
<code>configure --disable-everything --enable-decoder=foo</code>
(or <code>--enable-demuxer</code> or whatever your component is)?
</li></ol>
<a name="patch-submission-checklist"></a>
<h3 class="section">1.7 patch submission checklist<span class="pull-right"><a class="anchor hidden-xs" href="#patch-submission-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-patch-submission-checklist" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Does <code>make fate</code> pass with the patch applied?
</li><li> Was the patch generated with git format-patch or send-email?
</li><li> Did you sign off your patch? (git commit -s)
See <a href="http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches">http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches</a> for the meaning
of sign off.
</li><li> Did you provide a clear git commit log message?
</li><li> Is the patch against latest FFmpeg git master branch?
</li><li> Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
</li><li> Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
</li><li> If the change is to speed critical code, did you benchmark it?
</li><li> If you did any benchmarks, did you provide them in the mail?
</li><li> Have you checked that the patch does not introduce buffer overflows or
other security issues?
</li><li> Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
<a href="http://caca.zoy.org/wiki/zzuf">zzuf</a>. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
</li><li> Does the patch not mix functional and cosmetic changes?
</li><li> Did you add tabs or trailing whitespace to the code? Both are forbidden.
</li><li> Is the patch attached to the email you send?
</li><li> Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
</li><li> If the patch fixes a bug, did you provide a verbose analysis of the bug?
</li><li> If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples &gt;100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
</li><li> Did you provide a verbose summary about what the patch does change?
</li><li> Did you provide a verbose explanation why it changes things like it does?
</li><li> Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
</li><li> Did you provide an example so we can verify the new feature added by the
patch easily?
</li><li> If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
</li><li> You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
</li><li> Lines with similar content should be aligned vertically when doing so
improves readability.
</li><li> Consider to add a regression test for your code.
</li><li> If you added YASM code please check that things still work with &ndash;disable-yasm
</li><li> Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like <code>av_malloc()</code>
are notoriously left unchecked, which is a serious problem.
</li><li> Test your code with valgrind and or Address Sanitizer to ensure it&rsquo;s free
of leaks, out of array accesses, etc.
</li></ol>
<a name="Patch-review-process"></a>
<h3 class="section">1.8 Patch review process<span class="pull-right"><a class="anchor hidden-xs" href="#Patch-review-process" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patch-review-process" aria-hidden="true">TOC</a></span></h3>
<p>All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
</p>
<p>We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
</p>
<p>If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
</p>
<p>When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
</p>
<a name="Regression-tests"></a><a name="Regression-tests-1"></a>
<h3 class="section">1.9 Regression tests<span class="pull-right"><a class="anchor hidden-xs" href="#Regression-tests-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Regression-tests-1" aria-hidden="true">TOC</a></span></h3>
<p>Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
</p>
<p>Running &rsquo;make fate&rsquo; accomplishes this, please see <a href="fate.html">fate.html</a> for details.
</p>
<p>[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
</p>
<a name="Adding-files-to-the-fate_002dsuite-dataset"></a>
<h4 class="subsection">1.9.1 Adding files to the fate-suite dataset<span class="pull-right"><a class="anchor hidden-xs" href="#Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">TOC</a></span></h4>
<p>When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
</p>
<a name="Visualizing-Test-Coverage"></a>
<h4 class="subsection">1.9.2 Visualizing Test Coverage<span class="pull-right"><a class="anchor hidden-xs" href="#Visualizing-Test-Coverage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Visualizing-Test-Coverage" aria-hidden="true">TOC</a></span></h4>
<p>The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools <code>gcov</code>/<code>lcov</code>. This involves
the following steps:
</p>
<ol>
<li> Configure to compile with instrumentation enabled:
<code>configure --toolchain=gcov</code>.
</li><li> Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
</li><li> Run <code>make lcov</code> to generate coverage data in HTML format.
</li><li> View <code>lcov/index.html</code> in your preferred HTML viewer.
</li></ol>
<p>You can use the command <code>make lcov-reset</code> to reset the coverage
measurements. You will need to rerun <code>make lcov</code> after running a
new test.
</p>
<a name="Using-Valgrind"></a>
<h4 class="subsection">1.9.3 Using Valgrind<span class="pull-right"><a class="anchor hidden-xs" href="#Using-Valgrind" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-Valgrind" aria-hidden="true">TOC</a></span></h4>
<p>The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
<code>--toolchain=valgrind-memcheck</code> or <code>--toolchain=valgrind-massif</code>
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the <strong>memcheck</strong> or the
<strong>massif</strong> tool of the valgrind suite.
</p>
<p>In case you need finer control over how valgrind is invoked, use the
<code>--target-exec='valgrind &lt;your_custom_valgrind_options&gt;</code> option in
your configure line instead.
</p>
<a name="Release-process"></a><a name="Release-process-1"></a>
<h3 class="section">1.10 Release process<span class="pull-right"><a class="anchor hidden-xs" href="#Release-process-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-process-1" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg maintains a set of <strong>release branches</strong>, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a <strong>release
manager</strong> prepares, tests and publishes tarballs on the
<a href="http://ffmpeg.org">http://ffmpeg.org</a> website.
</p>
<p>There are two kinds of releases:
</p>
<ol>
<li> <strong>Major releases</strong> always include the latest and greatest
features and functionality.
</li><li> <strong>Point releases</strong> are cut from <strong>release</strong> branches,
which are named <code>release/X</code>, with <code>X</code> being the release
version number.
</li></ol>
<p>Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been <strong>compiled</strong> against
previous versions of <strong>the same release series</strong> in any case!
</p>
<p>However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the <strong>ffmpeg-devel</strong> mailing list in time to allow forward planning.
</p>
<a name="Criteria-for-Point-Releases"></a><a name="Criteria-for-Point-Releases-1"></a>
<h4 class="subsection">1.10.1 Criteria for Point Releases<span class="pull-right"><a class="anchor hidden-xs" href="#Criteria-for-Point-Releases-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Criteria-for-Point-Releases-1" aria-hidden="true">TOC</a></span></h4>
<p>Changes that match the following criteria are valid candidates for
inclusion into a point release:
</p>
<ol>
<li> Fixes a security issue, preferably identified by a <strong>CVE
number</strong> issued by <a href="http://cve.mitre.org/">http://cve.mitre.org/</a>.
</li><li> Fixes a documented bug in <a href="https://trac.ffmpeg.org">https://trac.ffmpeg.org</a>.
</li><li> Improves the included documentation.
</li><li> Retains both source code and binary compatibility with previous
point releases of the same release branch.
</li></ol>
<p>The order for checking the rules is (1 OR 2 OR 3) AND 4.
</p>
<a name="Release-Checklist"></a>
<h4 class="subsection">1.10.2 Release Checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Release-Checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-Checklist" aria-hidden="true">TOC</a></span></h4>
<p>The release process involves the following steps:
</p>
<ol>
<li> Ensure that the <samp>RELEASE</samp> file contains the version number for
the upcoming release.
</li><li> Add the release at <a href="https://trac.ffmpeg.org/admin/ticket/versions">https://trac.ffmpeg.org/admin/ticket/versions</a>.
</li><li> Announce the intent to do a release to the mailing list.
</li><li> Make sure all relevant security fixes have been backported. See
<a href="https://ffmpeg.org/security.html">https://ffmpeg.org/security.html</a>.
</li><li> Ensure that the FATE regression suite still passes in the release
branch on at least <strong>i386</strong> and <strong>amd64</strong>
(cf. <a href="#Regression-tests">Regression tests</a>).
</li><li> Prepare the release tarballs in <code>bz2</code> and <code>gz</code> formats, and
supplementing files that contain <code>gpg</code> signatures
</li><li> Publish the tarballs at <a href="http://ffmpeg.org/releases">http://ffmpeg.org/releases</a>. Create and
push an annotated tag in the form <code>nX</code>, with <code>X</code>
containing the version number.
</li><li> Propose and send a patch to the <strong>ffmpeg-devel</strong> mailing list
with a news entry for the website.
</li><li> Publish the news entry.
</li><li> Send announcement to the mailing list.
</li></ol>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

44
Externals/ffmpeg/dev/doc/examples/Makefile vendored
View File

@ -1,44 +0,0 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

23
Externals/ffmpeg/dev/doc/examples/README vendored
View File

@ -1,23 +0,0 @@
FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then just run "make examples".
This will build the examples using the FFmpeg build system. You can clean those
examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make.

134
Externals/ffmpeg/dev/doc/examples/avio_reading.c vendored
View File

@ -1,134 +0,0 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

665
Externals/ffmpeg/dev/doc/examples/decoding_encoding.c vendored
View File

@ -1,665 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavcodec API use example.
*
* @example decoding_encoding.c
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(c->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i=0; i<decoded_frame->nb_samples; i++)
for (ch=0; ch<c->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&decoded_frame);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1,25};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
c->pix_fmt, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
fflush(stdout);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for (;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
exit(1);
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&frame);
printf("\n");
}
int main(int argc, char **argv)
{
const char *output_type;
/* register all the codecs */
avcodec_register_all();
if (argc < 2) {
printf("usage: %s output_type\n"
"API example program to decode/encode a media stream with libavcodec.\n"
"This program generates a synthetic stream and encodes it to a file\n"
"named test.h264, test.mp2 or test.mpg depending on output_type.\n"
"The encoded stream is then decoded and written to a raw data output.\n"
"output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
output_type = argv[1];
if (!strcmp(output_type, "h264")) {
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.pcm", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");
} else {
fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
output_type);
return 1;
}
return 0;
}

386
Externals/ffmpeg/dev/doc/examples/demuxing_decoding.c vendored
View File

@ -1,386 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* The different ways of decoding and managing data memory. You are not
* supposed to support all the modes in your application but pick the one most
* appropriate to your needs. Look for the use of api_mode in this example to
* see what are the differences of API usage between them */
enum {
API_MODE_OLD = 0, /* old method, deprecated */
API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */
API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
};
static int api_mode = API_MODE_OLD;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
/* If we use the new API with reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the decoders, with or without reference counting */
if (api_mode == API_MODE_NEW_API_REF_COUNT)
av_dict_set(&opts, "refcounted_frames", "1", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
"input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call. If unset, it's using\n"
"the classic old method.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5) {
const char *mode = argv[1] + strlen("-refcount=");
if (!strcmp(mode, "old")) api_mode = API_MODE_OLD;
else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT;
else {
fprintf(stderr, "unknow mode '%s'\n", mode);
exit(1);
}
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
ret = av_image_alloc(video_dst_data, video_dst_linesize,
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
/* When using the new API, you need to use the libavutil/frame.h API, while
* the classic frame management is available in libavcodec */
if (api_mode == API_MODE_OLD)
frame = avcodec_alloc_frame();
else
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_close(video_dec_ctx);
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
if (api_mode == API_MODE_OLD)
avcodec_free_frame(&frame);
else
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}

185
Externals/ffmpeg/dev/doc/examples/extract_mvs.c vendored
View File

@ -1,185 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
static const char *src_filename = NULL;
static int video_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
src_filename = argv[1];
av_register_all();
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
}
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

365
Externals/ffmpeg/dev/doc/examples/filter_audio.c vendored
View File

@ -1,365 +0,0 @@
/*
* copyright (c) 2013 Andrew Kelley
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include <inttypes.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink)
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
AVFilter *volume;
AVFilterContext *aformat_ctx;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
static int get_input(AVFrame *frame, int frame_num)
{
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float*)frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
int main(int argc, char *argv[])
{
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
return 1;
}
duration = atof(argv[1]);
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}

280
Externals/ffmpeg/dev/doc/examples/filtering_audio.c vendored
View File

@ -1,280 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet0, packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index) {
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
if (packet.size <= 0)
av_free_packet(&packet0);
} else {
/* discard non-wanted packets */
av_free_packet(&packet0);
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

262
Externals/ffmpeg/dev/doc/examples/filtering_video.c vendored
View File

@ -1,262 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

56
Externals/ffmpeg/dev/doc/examples/metadata.c vendored
View File

@ -1,56 +0,0 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}

670
Externals/ffmpeg/dev/doc/examples/muxing.c vendored
View File

@ -1,670 +0,0 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, *codec);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = ost->st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->st->codec;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->st->codec;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i, ret;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(pict);
if (ret < 0)
exit(1);
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->st->codec;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
c = ost->st->codec;
frame = get_video_frame(ost);
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* a hack to avoid data copy with some raw video muxers */
AVPacket pkt;
av_init_packet(&pkt);
if (!frame)
return 1;
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = ost->st->index;
pkt.data = (uint8_t *)frame;
pkt.size = sizeof(AVPicture);
pkt.pts = pkt.dts = frame->pts;
av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
AVPacket pkt = { 0 };
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_close(ost->st->codec);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
if (argc > 3 && !strcmp(argv[2], "-flags")) {
av_dict_set(&opt, argv[2]+1, argv[3], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}

165
Externals/ffmpeg/dev/doc/examples/remuxing.c vendored
View File

@ -1,165 +0,0 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
in_filename = argv[1];
out_filename = argv[2];
av_register_all();
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *in_stream = ifmt_ctx->streams[i];
AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
if (ret < 0) {
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
out_stream->codec->codec_tag = 0;
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_free_packet(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

214
Externals/ffmpeg/dev/doc/examples/resampling_audio.c vendored
View File

@ -1,214 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

140
Externals/ffmpeg/dev/doc/examples/scaling_video.c vendored
View File

@ -1,140 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

755
Externals/ffmpeg/dev/doc/examples/transcode_aac.c vendored
View File

@ -1,755 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 48000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
/**
* Convert an error code into a text message.
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static const char *get_error_text(const int error)
{
static char error_buffer[255];
av_strerror(error, error_buffer, sizeof(error_buffer));
return error_buffer;
}
/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodec *input_codec;
int error;
/** Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
filename, get_error_text(error));
*input_format_context = NULL;
return error;
}
/** Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = (*input_format_context)->streams[0]->codec;
return 0;
}
/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/** Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
filename, get_error_text(error));
return error;
}
/** Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/** Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/** Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/** Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/** Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/** Save the encoder context for easiert access later. */
*output_codec_context = stream->codec;
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
(*output_codec_context)->channels = OUTPUT_CHANNELS;
(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
(*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
(*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
return 0;
cleanup:
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/** Initialize one data packet for reading or writing. */
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/** Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/** Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int init_fifo(AVAudioFifo **fifo)
{
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/** Write the header of the output file container. */
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/** Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
get_error_text(error));
return error;
}
}
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&input_packet);
return error;
}
/**
* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again.
*/
if (*finished && *data_present)
*finished = 0;
av_free_packet(&input_packet);
return 0;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
get_error_text(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/** Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Add converted input audio samples to the FIFO buffer for later processing. */
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/** Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/** Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
/** Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/** Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/**
* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error.
*/
if (*finished && !data_present) {
ret = 0;
goto cleanup;
}
/** If there is decoded data, convert and store it */
if (data_present) {
/** Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/** Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/** Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
get_error_text(error));
av_frame_free(frame);
return error;
}
return 0;
}
/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/** Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
/** Write one audio frame from the temporary packet to the output file. */
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
av_free_packet(&output_packet);
}
return 0;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/** Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/** Write the trailer of the output file container. */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/** Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/** Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo))
goto cleanup;
/** Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
while (1) {
/** Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/**
* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
if (finished)
break;
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
if (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/** Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_close(output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

583
Externals/ffmpeg/dev/doc/examples/transcoding.c vendored
View File

@ -1,583 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream;
AVCodecContext *codec_ctx;
stream = ifmt_ctx->streams[i];
codec_ctx = stream->codec;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* Open decoder */
ret = avcodec_open2(codec_ctx,
avcodec_find_decoder(codec_ctx->codec_id), NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = in_stream->codec;
enc_ctx = out_stream->codec;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
enc_ctx->pix_fmt = encoder->pix_fmts[0];
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_free_packet(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_free_packet(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_close(ifmt_ctx->streams[i]->codec);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
avcodec_close(ofmt_ctx->streams[i]->codec);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

719
Externals/ffmpeg/dev/doc/faq.html vendored
View File

@ -1,719 +0,0 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
FFmpeg FAQ
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
FFmpeg FAQ
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-General-Questions" href="#General-Questions">1 General Questions</a>
<ul class="no-bullet">
<li><a name="toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?</a></li>
<li><a name="toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?</a></li>
<li><a name="toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e">1.3 I cannot read this file although this format seems to be supported by ffmpeg.</a></li>
<li><a name="toc-Which-codecs-are-supported-by-Windows_003f" href="#Which-codecs-are-supported-by-Windows_003f">1.4 Which codecs are supported by Windows?</a></li>
</ul></li>
<li><a name="toc-Compilation" href="#Compilation">2 Compilation</a>
<ul class="no-bullet">
<li><a name="toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code></a></li>
<li><a name="toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?</a></li>
<li><a name="toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f">2.3 How do I make <code>pkg-config</code> find my libraries?</a></li>
<li><a name="toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f">2.4 How do I use <code>pkg-config</code> when cross-compiling?</a></li>
</ul></li>
<li><a name="toc-Usage" href="#Usage">3 Usage</a>
<ul class="no-bullet">
<li><a name="toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f">3.1 ffmpeg does not work; what is wrong?</a></li>
<li><a name="toc-How-do-I-encode-single-pictures-into-movies_003f" href="#How-do-I-encode-single-pictures-into-movies_003f">3.2 How do I encode single pictures into movies?</a></li>
<li><a name="toc-How-do-I-encode-movie-to-single-pictures_003f" href="#How-do-I-encode-movie-to-single-pictures_003f">3.3 How do I encode movie to single pictures?</a></li>
<li><a name="toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?</a></li>
<li><a name="toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f">3.5 How can I read from the standard input or write to the standard output?</a></li>
<li><a name="toc-_002df-jpeg-doesn_0027t-work_002e" href="#g_t_002df-jpeg-doesn_0027t-work_002e">3.6 -f jpeg doesn&rsquo;t work.</a></li>
<li><a name="toc-Why-can-I-not-change-the-frame-rate_003f" href="#Why-can-I-not-change-the-frame-rate_003f">3.7 Why can I not change the frame rate?</a></li>
<li><a name="toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f">3.8 How do I encode Xvid or DivX video with ffmpeg?</a></li>
<li><a name="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f">3.9 Which are good parameters for encoding high quality MPEG-4?</a></li>
<li><a name="toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?</a></li>
<li><a name="toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?</a></li>
<li><a name="toc-How-can-I-read-DirectShow-files_003f" href="#How-can-I-read-DirectShow-files_003f">3.12 How can I read DirectShow files?</a></li>
<li><a name="toc-How-can-I-join-video-files_003f" href="#How-can-I-join-video-files_003f">3.13 How can I join video files?</a></li>
<li><a name="toc-How-can-I-concatenate-video-files_003f" href="#How-can-I-concatenate-video-files_003f">3.14 How can I concatenate video files?</a>
<ul class="no-bullet">
<li><a name="toc-Concatenating-using-the-concat-filter" href="#Concatenating-using-the-concat-filter">3.14.1 Concatenating using the concat <em>filter</em></a></li>
<li><a name="toc-Concatenating-using-the-concat-demuxer" href="#Concatenating-using-the-concat-demuxer">3.14.2 Concatenating using the concat <em>demuxer</em></a></li>
<li><a name="toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029">3.14.3 Concatenating using the concat <em>protocol</em> (file level)</a></li>
<li><a name="toc-Concatenating-using-raw-audio-and-video" href="#Concatenating-using-raw-audio-and-video">3.14.4 Concatenating using raw audio and video</a></li>
</ul></li>
<li><a name="toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.</a></li>
<li><a name="toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f">3.16 Why does FFmpeg not see the subtitles in my VOB file?</a></li>
<li><a name="toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?</a></li>
</ul></li>
<li><a name="toc-Development" href="#Development">4 Development</a>
<ul class="no-bullet">
<li><a name="toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?</a></li>
<li><a name="toc-Can-you-support-my-C-compiler-XXX_003f" href="#Can-you-support-my-C-compiler-XXX_003f">4.2 Can you support my C compiler XXX?</a></li>
<li><a name="toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f">4.3 Is Microsoft Visual C++ supported?</a></li>
<li><a name="toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f">4.4 Can you add automake, libtool or autoconf support?</a></li>
<li><a name="toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f">4.5 Why not rewrite FFmpeg in object-oriented C++?</a></li>
<li><a name="toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f">4.6 Why are the ffmpeg programs devoid of debugging symbols?</a></li>
<li><a name="toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f">4.7 I do not like the LGPL, can I contribute code under the GPL instead?</a></li>
<li><a name="toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.</a></li>
<li><a name="toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.</a></li>
<li><a name="toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope</a></li>
<li><a name="toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?</a></li>
<li><a name="toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?</a></li>
<li><a name="toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?</a></li>
<li><a name="toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.</a></li>
<li><a name="toc-Why-is-make-fate-not-running-all-tests_003f" href="#Why-is-make-fate-not-running-all-tests_003f">4.15 Why is <code>make fate</code> not running all tests?</a></li>
<li><a name="toc-Why-is-make-fate-not-finding-the-samples_003f" href="#Why-is-make-fate-not-finding-the-samples_003f">4.16 Why is <code>make fate</code> not finding the samples?</a></li>
</ul></li>
</ul>
</div>
<a name="General-Questions"></a>
<h2 class="chapter">1 General Questions<span class="pull-right"><a class="anchor hidden-xs" href="#General-Questions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-General-Questions" aria-hidden="true">TOC</a></span></h2>
<a name="Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f"></a>
<h3 class="section">1.1 Why doesn&rsquo;t FFmpeg support feature [xyz]?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-doesn_0027t-FFmpeg-support-feature-_005bxyz_005d_003f" aria-hidden="true">TOC</a></span></h3>
<p>Because no one has taken on that task yet. FFmpeg development is
driven by the tasks that are important to the individual developers.
If there is a feature that is important to you, the best way to get
it implemented is to undertake the task yourself or sponsor a developer.
</p>
<a name="FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f"></a>
<h3 class="section">1.2 FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-does-not-support-codec-XXX_002e-Can-you-include-a-Windows-DLL-loader-to-support-it_003f" aria-hidden="true">TOC</a></span></h3>
<p>No. Windows DLLs are not portable, bloated and often slow.
Moreover FFmpeg strives to support all codecs natively.
A DLL loader is not conducive to that goal.
</p>
<a name="I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e"></a>
<h3 class="section">1.3 I cannot read this file although this format seems to be supported by ffmpeg.<span class="pull-right"><a class="anchor hidden-xs" href="#I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-cannot-read-this-file-although-this-format-seems-to-be-supported-by-ffmpeg_002e" aria-hidden="true">TOC</a></span></h3>
<p>Even if ffmpeg can read the container format, it may not support all its
codecs. Please consult the supported codec list in the ffmpeg
documentation.
</p>
<a name="Which-codecs-are-supported-by-Windows_003f"></a>
<h3 class="section">1.4 Which codecs are supported by Windows?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-codecs-are-supported-by-Windows_003f" aria-hidden="true">TOC</a></span></h3>
<p>Windows does not support standard formats like MPEG very well, unless you
install some additional codecs.
</p>
<p>The following list of video codecs should work on most Windows systems:
</p><dl compact="compact">
<dt><samp>msmpeg4v2</samp></dt>
<dd><p>.avi/.asf
</p></dd>
<dt><samp>msmpeg4</samp></dt>
<dd><p>.asf only
</p></dd>
<dt><samp>wmv1</samp></dt>
<dd><p>.asf only
</p></dd>
<dt><samp>wmv2</samp></dt>
<dd><p>.asf only
</p></dd>
<dt><samp>mpeg4</samp></dt>
<dd><p>Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
</p></dd>
<dt><samp>mpeg1video</samp></dt>
<dd><p>.mpg only
</p></dd>
</dl>
<p>Note, ASF files often have .wmv or .wma extensions in Windows. It should also
be mentioned that Microsoft claims a patent on the ASF format, and may sue
or threaten users who create ASF files with non-Microsoft software. It is
strongly advised to avoid ASF where possible.
</p>
<p>The following list of audio codecs should work on most Windows systems:
</p><dl compact="compact">
<dt><samp>adpcm_ima_wav</samp></dt>
<dt><samp>adpcm_ms</samp></dt>
<dt><samp>pcm_s16le</samp></dt>
<dd><p>always
</p></dd>
<dt><samp>libmp3lame</samp></dt>
<dd><p>If some MP3 codec like LAME is installed.
</p></dd>
</dl>
<a name="Compilation"></a>
<h2 class="chapter">2 Compilation<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation" aria-hidden="true">TOC</a></span></h2>
<a name="error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027"></a>
<h3 class="section">2.1 <code>error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'</code><span class="pull-right"><a class="anchor hidden-xs" href="#error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-error_003a-can_0027t-find-a-register-in-class-_0027GENERAL_005fREGS_0027-while-reloading-_0027asm_0027" aria-hidden="true">TOC</a></span></h3>
<p>This is a bug in gcc. Do not report it to us. Instead, please report it to
the gcc developers. Note that we will not add workarounds for gcc bugs.
</p>
<p>Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
<a href="http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203">http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203</a>.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
</p>
<a name="I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f"></a>
<h3 class="section">2.2 I have installed this library with my distro&rsquo;s package manager. Why does <code>configure</code> not see it?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-installed-this-library-with-my-distro_0027s-package-manager_002e-Why-does-configure-not-see-it_003f" aria-hidden="true">TOC</a></span></h3>
<p>Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
</p>
<p>To build FFmpeg, you need to install the development package. It is usually
called <samp>libfoo-dev</samp> or <samp>libfoo-devel</samp>. You can remove it after the
build is finished, but be sure to keep the main package.
</p>
<a name="How-do-I-make-pkg_002dconfig-find-my-libraries_003f"></a>
<h3 class="section">2.3 How do I make <code>pkg-config</code> find my libraries?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-make-pkg_002dconfig-find-my-libraries_003f" aria-hidden="true">TOC</a></span></h3>
<p>Somewhere along with your libraries, there is a <samp>.pc</samp> file (or several)
in a <samp>pkgconfig</samp> directory. You need to set environment variables to
point <code>pkg-config</code> to these files.
</p>
<p>If you need to <em>add</em> directories to <code>pkg-config</code>&rsquo;s search list
(typical use case: library installed separately), add it to
<code>$PKG_CONFIG_PATH</code>:
</p>
<div class="example">
<pre class="example">export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
</pre></div>
<p>If you need to <em>replace</em> <code>pkg-config</code>&rsquo;s search list
(typical use case: cross-compiling), set it in
<code>$PKG_CONFIG_LIBDIR</code>:
</p>
<div class="example">
<pre class="example">export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
</pre></div>
<p>If you need to know the library&rsquo;s internal dependencies (typical use: static
linking), add the <code>--static</code> option to <code>pkg-config</code>:
</p>
<div class="example">
<pre class="example">./configure --pkg-config-flags=--static
</pre></div>
<a name="How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f"></a>
<h3 class="section">2.4 How do I use <code>pkg-config</code> when cross-compiling?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-use-pkg_002dconfig-when-cross_002dcompiling_003f" aria-hidden="true">TOC</a></span></h3>
<p>The best way is to install <code>pkg-config</code> in your cross-compilation
environment. It will automatically use the cross-compilation libraries.
</p>
<p>You can also use <code>pkg-config</code> from the host environment by
specifying explicitly <code>--pkg-config=pkg-config</code> to <code>configure</code>.
In that case, you must point <code>pkg-config</code> to the correct directories
using the <code>PKG_CONFIG_LIBDIR</code>, as explained in the previous entry.
</p>
<p>As an intermediate solution, you can place in your cross-compilation
environment a script that calls the host <code>pkg-config</code> with
<code>PKG_CONFIG_LIBDIR</code> set. That script can look like that:
</p>
<div class="example">
<pre class="example">#!/bin/sh
PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
export PKG_CONFIG_LIBDIR
exec /usr/bin/pkg-config &quot;$@&quot;
</pre></div>
<a name="Usage"></a>
<h2 class="chapter">3 Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Usage" aria-hidden="true">TOC</a></span></h2>
<a name="ffmpeg-does-not-work_003b-what-is-wrong_003f"></a>
<h3 class="section">3.1 ffmpeg does not work; what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-ffmpeg-does-not-work_003b-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>
<p>Try a <code>make distclean</code> in the ffmpeg source directory before the build.
If this does not help see
(<a href="http://ffmpeg.org/bugreports.html">http://ffmpeg.org/bugreports.html</a>).
</p>
<a name="How-do-I-encode-single-pictures-into-movies_003f"></a>
<h3 class="section">3.2 How do I encode single pictures into movies?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-single-pictures-into-movies_003f" aria-hidden="true">TOC</a></span></h3>
<p>First, rename your pictures to follow a numerical sequence.
For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
</pre></div>
<p>Notice that &lsquo;<samp>%d</samp>&rsquo; is replaced by the image number.
</p>
<p><samp>img%03d.jpg</samp> means the sequence <samp>img001.jpg</samp>, <samp>img002.jpg</samp>, etc.
</p>
<p>Use the <samp>-start_number</samp> option to declare a starting number for
the sequence. This is useful if your sequence does not start with
<samp>img001.jpg</samp> but is still in a numerical order. The following
example will start with <samp>img100.jpg</samp>:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
</pre></div>
<p>If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match <code>*jpg</code> to the <samp>/tmp</samp> directory in the sequence of
<samp>img001.jpg</samp>, <samp>img002.jpg</samp> and so on.
</p>
<div class="example">
<pre class="example">x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s &quot;$i&quot; /tmp/img&quot;$counter&quot;.jpg; x=$(($x+1)); done
</pre></div>
<p>If you want to sequence them by oldest modified first, substitute
<code>$(ls -r -t *jpg)</code> in place of <code>*jpg</code>.
</p>
<p>Then run:
</p>
<div class="example">
<pre class="example">ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
</pre></div>
<p>The same logic is used for any image format that ffmpeg reads.
</p>
<p>You can also use <code>cat</code> to pipe images to ffmpeg:
</p>
<div class="example">
<pre class="example">cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
</pre></div>
<a name="How-do-I-encode-movie-to-single-pictures_003f"></a>
<h3 class="section">3.3 How do I encode movie to single pictures?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-movie-to-single-pictures_003f" aria-hidden="true">TOC</a></span></h3>
<p>Use:
</p>
<div class="example">
<pre class="example">ffmpeg -i movie.mpg movie%d.jpg
</pre></div>
<p>The <samp>movie.mpg</samp> used as input will be converted to
<samp>movie1.jpg</samp>, <samp>movie2.jpg</samp>, etc...
</p>
<p>Instead of relying on file format self-recognition, you may also use
</p><dl compact="compact">
<dt><samp>-c:v ppm</samp></dt>
<dt><samp>-c:v png</samp></dt>
<dt><samp>-c:v mjpeg</samp></dt>
</dl>
<p>to force the encoding.
</p>
<p>Applying that to the previous example:
</p><div class="example">
<pre class="example">ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
</pre></div>
<p>Beware that there is no &quot;jpeg&quot; codec. Use &quot;mjpeg&quot; instead.
</p>
<a name="Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f"></a>
<h3 class="section">3.4 Why do I see a slight quality degradation with multithreaded MPEG* encoding?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-do-I-see-a-slight-quality-degradation-with-multithreaded-MPEG_002a-encoding_003f" aria-hidden="true">TOC</a></span></h3>
<p>For multithreaded MPEG* encoding, the encoded slices must be independent,
otherwise thread n would practically have to wait for n-1 to finish, so it&rsquo;s
quite logical that there is a small reduction of quality. This is not a bug.
</p>
<a name="How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f"></a>
<h3 class="section">3.5 How can I read from the standard input or write to the standard output?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-from-the-standard-input-or-write-to-the-standard-output_003f" aria-hidden="true">TOC</a></span></h3>
<p>Use <samp>-</samp> as file name.
</p>
<a name="g_t_002df-jpeg-doesn_0027t-work_002e"></a>
<h3 class="section">3.6 -f jpeg doesn&rsquo;t work.<span class="pull-right"><a class="anchor hidden-xs" href="#_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_002df-jpeg-doesn_0027t-work_002e" aria-hidden="true">TOC</a></span></h3>
<p>Try &rsquo;-f image2 test%d.jpg&rsquo;.
</p>
<a name="Why-can-I-not-change-the-frame-rate_003f"></a>
<h3 class="section">3.7 Why can I not change the frame rate?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-can-I-not-change-the-frame-rate_003f" aria-hidden="true">TOC</a></span></h3>
<p>Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v command line option.
</p>
<a name="How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f"></a>
<h3 class="section">3.8 How do I encode Xvid or DivX video with ffmpeg?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-encode-Xvid-or-DivX-video-with-ffmpeg_003f" aria-hidden="true">TOC</a></span></h3>
<p>Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
standard (note that there are many other coding formats that use this
same standard). Thus, use &rsquo;-c:v mpeg4&rsquo; to encode in these formats. The
default fourcc stored in an MPEG-4-coded file will be &rsquo;FMP4&rsquo;. If you want
a different fourcc, use the &rsquo;-vtag&rsquo; option. E.g., &rsquo;-vtag xvid&rsquo; will
force the fourcc &rsquo;xvid&rsquo; to be stored as the video fourcc rather than the
default.
</p>
<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f"></a>
<h3 class="section">3.9 Which are good parameters for encoding high quality MPEG-4?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d4_003f" aria-hidden="true">TOC</a></span></h3>
<p>&rsquo;-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2&rsquo;,
things to try: &rsquo;-bf 2&rsquo;, &rsquo;-flags qprd&rsquo;, &rsquo;-flags mv0&rsquo;, &rsquo;-flags skiprd&rsquo;.
</p>
<a name="Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f"></a>
<h3 class="section">3.10 Which are good parameters for encoding high quality MPEG-1/MPEG-2?<span class="pull-right"><a class="anchor hidden-xs" href="#Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Which-are-good-parameters-for-encoding-high-quality-MPEG_002d1_002fMPEG_002d2_003f" aria-hidden="true">TOC</a></span></h3>
<p>&rsquo;-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2&rsquo;
but beware the &rsquo;-g 100&rsquo; might cause problems with some decoders.
Things to try: &rsquo;-bf 2&rsquo;, &rsquo;-flags qprd&rsquo;, &rsquo;-flags mv0&rsquo;, &rsquo;-flags skiprd.
</p>
<a name="Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f"></a>
<h3 class="section">3.11 Interlaced video looks very bad when encoded with ffmpeg, what is wrong?<span class="pull-right"><a class="anchor hidden-xs" href="#Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Interlaced-video-looks-very-bad-when-encoded-with-ffmpeg_002c-what-is-wrong_003f" aria-hidden="true">TOC</a></span></h3>
<p>You should use &rsquo;-flags +ilme+ildct&rsquo; and maybe &rsquo;-flags +alt&rsquo; for interlaced
material, and try &rsquo;-top 0/1&rsquo; if the result looks really messed-up.
</p>
<a name="How-can-I-read-DirectShow-files_003f"></a>
<h3 class="section">3.12 How can I read DirectShow files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-read-DirectShow-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-read-DirectShow-files_003f" aria-hidden="true">TOC</a></span></h3>
<p>If you have built FFmpeg with <code>./configure --enable-avisynth</code>
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
</p>
<p>Just create an &quot;input.avs&quot; text file with this single line ...
</p><div class="example">
<pre class="example">DirectShowSource(&quot;C:\path to your file\yourfile.asf&quot;)
</pre></div>
<p>... and then feed that text file to ffmpeg:
</p><div class="example">
<pre class="example">ffmpeg -i input.avs
</pre></div>
<p>For ANY other help on AviSynth, please visit the
<a href="http://www.avisynth.org/">AviSynth homepage</a>.
</p>
<a name="How-can-I-join-video-files_003f"></a>
<h3 class="section">3.13 How can I join video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-join-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-join-video-files_003f" aria-hidden="true">TOC</a></span></h3>
<p>To &quot;join&quot; video files is quite ambiguous. The following list explains the
different kinds of &quot;joining&quot; and points out how those are addressed in
FFmpeg. To join video files may mean:
</p>
<ul>
<li> To put them one after the other: this is called to <em>concatenate</em> them
(in short: concat) and is addressed
<a href="#How-can-I-concatenate-video-files">in this very faq</a>.
</li><li> To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
<em>multiplex</em> them together (in short: mux), and is done by simply
invoking ffmpeg with several <samp>-i</samp> options.
</li><li> For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
<em>merge</em> them, and can be done using the
<a href="http://ffmpeg.org/ffmpeg-filters.html#amerge"><code>amerge</code></a> filter.
</li><li> For audio, to play one on top of the other: this is called to <em>mix</em>
them, and can be done by first merging them into a single stream and then
using the <a href="http://ffmpeg.org/ffmpeg-filters.html#pan"><code>pan</code></a> filter to mix
the channels at will.
</li><li> For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
<a href="http://ffmpeg.org/ffmpeg-filters.html#overlay"><code>overlay</code></a> video filter.
</li></ul>
<a name="How-can-I-concatenate-video-files"></a><a name="How-can-I-concatenate-video-files_003f"></a>
<h3 class="section">3.14 How can I concatenate video files?<span class="pull-right"><a class="anchor hidden-xs" href="#How-can-I-concatenate-video-files_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-can-I-concatenate-video-files_003f" aria-hidden="true">TOC</a></span></h3>
<p>There are several solutions, depending on the exact circumstances.
</p>
<a name="Concatenating-using-the-concat-filter"></a>
<h4 class="subsection">3.14.1 Concatenating using the concat <em>filter</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-filter" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-filter" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg has a <a href="http://ffmpeg.org/ffmpeg-filters.html#concat"><code>concat</code></a> filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
</p>
<a name="Concatenating-using-the-concat-demuxer"></a>
<h4 class="subsection">3.14.2 Concatenating using the concat <em>demuxer</em><span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-demuxer" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-demuxer" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg has a <a href="http://www.ffmpeg.org/ffmpeg-formats.html#concat"><code>concat</code></a> demuxer which you can use when you want to avoid a re-encode and
your format doesn&rsquo;t support file level concatenation.
</p>
<a name="Concatenating-using-the-concat-protocol-_0028file-level_0029"></a>
<h4 class="subsection">3.14.3 Concatenating using the concat <em>protocol</em> (file level)<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-the-concat-protocol-_0028file-level_0029" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg has a <a href="http://ffmpeg.org/ffmpeg-protocols.html#concat"><code>concat</code></a> protocol designed specifically for that, with examples in the
documentation.
</p>
<p>A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files containing them.
</p>
<p>Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble <code>cat</code> command (or the
equally humble <code>copy</code> under Windows), and finally transcoding back to your
format of choice.
</p>
<div class="example">
<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg &gt; intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
</pre></div>
<p>Additionally, you can use the <code>concat</code> protocol instead of <code>cat</code> or
<code>copy</code> which will avoid creation of a potentially huge intermediate file.
</p>
<div class="example">
<pre class="example">ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:&quot;intermediate1.mpg|intermediate2.mpg&quot; -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
</pre></div>
<p>Note that you may need to escape the character &quot;|&quot; which is special for many
shells.
</p>
<p>Another option is usage of named pipes, should your platform support it:
</p>
<div class="example">
<pre class="example">mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg &lt; /dev/null &amp;
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg &lt; /dev/null &amp;
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
</pre></div>
<a name="Concatenating-using-raw-audio-and-video"></a>
<h4 class="subsection">3.14.4 Concatenating using raw audio and video<span class="pull-right"><a class="anchor hidden-xs" href="#Concatenating-using-raw-audio-and-video" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Concatenating-using-raw-audio-and-video" aria-hidden="true">TOC</a></span></h4>
<p>Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
<code>tail</code> as seen below. Note that when piping through <code>tail</code> you
must use command grouping, <code>{ ;}</code>, to background properly.
</p>
<p>For example, let&rsquo;s say we want to concatenate two FLV files into an
output.flv file:
</p>
<div class="example">
<pre class="example">mkfifo temp1.a
mkfifo temp1.v
mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - &gt; temp1.a &lt; /dev/null &amp;
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - &gt; temp2.a &lt; /dev/null &amp;
ffmpeg -i input1.flv -an -f yuv4mpegpipe - &gt; temp1.v &lt; /dev/null &amp;
{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - &lt; /dev/null | tail -n +2 &gt; temp2.v ; } &amp;
cat temp1.a temp2.a &gt; all.a &amp;
cat temp1.v temp2.v &gt; all.v &amp;
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
</pre></div>
<a name="Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e"></a>
<h3 class="section">3.15 Using <samp>-f lavfi</samp>, audio becomes mono for no apparent reason.<span class="pull-right"><a class="anchor hidden-xs" href="#Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-_002df-lavfi_002c-audio-becomes-mono-for-no-apparent-reason_002e" aria-hidden="true">TOC</a></span></h3>
<p>Use <samp>-dumpgraph -</samp> to find out exactly where the channel layout is
lost.
</p>
<p>Most likely, it is through <code>auto-inserted aresample</code>. Try to understand
why the converting filter was needed at that place.
</p>
<p>Just before the output is a likely place, as <samp>-f lavfi</samp> currently
only support packed S16.
</p>
<p>Then insert the correct <code>aformat</code> explicitly in the filtergraph,
specifying the exact format.
</p>
<div class="example">
<pre class="example">aformat=sample_fmts=s16:channel_layouts=stereo
</pre></div>
<a name="Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f"></a>
<h3 class="section">3.16 Why does FFmpeg not see the subtitles in my VOB file?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-does-FFmpeg-not-see-the-subtitles-in-my-VOB-file_003f" aria-hidden="true">TOC</a></span></h3>
<p>VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initially detected.
</p>
<p>Some applications, including the <code>ffmpeg</code> command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.
</p>
<p>The size of the initial scan is controlled by two options: <code>probesize</code>
(default ~5 Mo) and <code>analyzeduration</code> (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
</p>
<a name="Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f"></a>
<h3 class="section">3.17 Why was the <code>ffmpeg</code> <samp>-sameq</samp> option removed? What to use instead?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-was-the-ffmpeg-_002dsameq-option-removed_003f-What-to-use-instead_003f" aria-hidden="true">TOC</a></span></h3>
<p>The <samp>-sameq</samp> option meant &quot;same quantizer&quot;, and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
&quot;same quality&quot; and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
</p>
<p>Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
<samp>-qscale</samp> and <samp>-qmax</samp>, but you should peruse the documentation
of the encoder you chose.
</p>
<a name="Development"></a>
<h2 class="chapter">4 Development<span class="pull-right"><a class="anchor hidden-xs" href="#Development" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development" aria-hidden="true">TOC</a></span></h2>
<a name="Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f"></a>
<h3 class="section">4.1 Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Are-there-examples-illustrating-how-to-use-the-FFmpeg-libraries_002c-particularly-libavcodec-and-libavformat_003f" aria-hidden="true">TOC</a></span></h3>
<p>Yes. Check the <samp>doc/examples</samp> directory in the source
repository, also available online at:
<a href="https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples">https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples</a>.
</p>
<p>Examples are also installed by default, usually in
<code>$PREFIX/share/ffmpeg/examples</code>.
</p>
<p>Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
examine the source code for one of the many open source projects that
already incorporate FFmpeg at (<a href="projects.html">projects.html</a>).
</p>
<a name="Can-you-support-my-C-compiler-XXX_003f"></a>
<h3 class="section">4.2 Can you support my C compiler XXX?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-support-my-C-compiler-XXX_003f" aria-hidden="true">TOC</a></span></h3>
<p>It depends. If your compiler is C99-compliant, then patches to support
it are likely to be welcome if they do not pollute the source code
with <code>#ifdef</code>s related to the compiler.
</p>
<a name="Is-Microsoft-Visual-C_002b_002b-supported_003f"></a>
<h3 class="section">4.3 Is Microsoft Visual C++ supported?<span class="pull-right"><a class="anchor hidden-xs" href="#Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Is-Microsoft-Visual-C_002b_002b-supported_003f" aria-hidden="true">TOC</a></span></h3>
<p>Yes. Please see the <a href="platform.html">Microsoft Visual C++</a>
section in the FFmpeg documentation.
</p>
<a name="Can-you-add-automake_002c-libtool-or-autoconf-support_003f"></a>
<h3 class="section">4.4 Can you add automake, libtool or autoconf support?<span class="pull-right"><a class="anchor hidden-xs" href="#Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Can-you-add-automake_002c-libtool-or-autoconf-support_003f" aria-hidden="true">TOC</a></span></h3>
<p>No. These tools are too bloated and they complicate the build.
</p>
<a name="Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f"></a>
<h3 class="section">4.5 Why not rewrite FFmpeg in object-oriented C++?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-not-rewrite-FFmpeg-in-object_002doriented-C_002b_002b_003f" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read <a href="http://www.tux.org/lkml/#s15">&quot;Programming Religion&quot;</a>.
</p>
<a name="Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f"></a>
<h3 class="section">4.6 Why are the ffmpeg programs devoid of debugging symbols?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-are-the-ffmpeg-programs-devoid-of-debugging-symbols_003f" aria-hidden="true">TOC</a></span></h3>
<p>The build process creates <code>ffmpeg_g</code>, <code>ffplay_g</code>, etc. which
contain full debug information. Those binaries are stripped to create
<code>ffmpeg</code>, <code>ffplay</code>, etc. If you need the debug information, use
the *_g versions.
</p>
<a name="I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f"></a>
<h3 class="section">4.7 I do not like the LGPL, can I contribute code under the GPL instead?<span class="pull-right"><a class="anchor hidden-xs" href="#I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-do-not-like-the-LGPL_002c-can-I-contribute-code-under-the-GPL-instead_003f" aria-hidden="true">TOC</a></span></h3>
<p>Yes, as long as the code is optional and can easily and cleanly be placed
under #if CONFIG_GPL without breaking anything. So, for example, a new codec
or filter would be OK under GPL while a bug fix to LGPL code would not.
</p>
<a name="I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e"></a>
<h3 class="section">4.8 I&rsquo;m using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C-application-but-the-linker-complains-about-missing-symbols-from-the-libraries-themselves_002e" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: <code>-lavdevice</code> must come before
<code>-lavformat</code>, <code>-lavutil</code> must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.
</p>
<p>An easy way to get the full list of required libraries in dependency order
is to use <code>pkg-config</code>.
</p>
<div class="example">
<pre class="example">c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
</pre></div>
<p>See <samp>doc/example/Makefile</samp> and <samp>doc/example/pc-uninstalled</samp> for
more details.
</p>
<a name="I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e"></a>
<h3 class="section">4.9 I&rsquo;m using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-FFmpeg-from-within-my-C_002b_002b-application-but-the-linker-complains-about-missing-symbols-which-seem-to-be-available_002e" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg is a pure C project, so to use the libraries within your C++ application
you need to explicitly state that you are using a C library. You can do this by
encompassing your FFmpeg includes using <code>extern &quot;C&quot;</code>.
</p>
<p>See <a href="http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3">http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3</a>
</p>
<a name="I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope"></a>
<h3 class="section">4.10 I&rsquo;m using libavutil from within my C++ application but the compiler complains about &rsquo;UINT64_C&rsquo; was not declared in this scope<span class="pull-right"><a class="anchor hidden-xs" href="#I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I_0027m-using-libavutil-from-within-my-C_002b_002b-application-but-the-compiler-complains-about-_0027UINT64_005fC_0027-was-not-declared-in-this-scope" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg is a pure C project using C99 math features, in order to enable C++
to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
</p>
<a name="I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f"></a>
<h3 class="section">4.11 I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?<span class="pull-right"><a class="anchor hidden-xs" href="#I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-I-have-a-file-in-memory-_002f-a-API-different-from-_002aopen_002f_002aread_002f-libc-how-do-I-use-it-with-libavformat_003f" aria-hidden="true">TOC</a></span></h3>
<p>You have to create a custom AVIOContext using <code>avio_alloc_context</code>,
see <samp>libavformat/aviobuf.c</samp> in FFmpeg and <samp>libmpdemux/demux_lavf.c</samp> in MPlayer or MPlayer2 sources.
</p>
<a name="Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f"></a>
<h3 class="section">4.12 Where is the documentation about ffv1, msmpeg4, asv1, 4xm?<span class="pull-right"><a class="anchor hidden-xs" href="#Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Where-is-the-documentation-about-ffv1_002c-msmpeg4_002c-asv1_002c-4xm_003f" aria-hidden="true">TOC</a></span></h3>
<p>see <a href="http://www.ffmpeg.org/~michael/">http://www.ffmpeg.org/~michael/</a>
</p>
<a name="How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f"></a>
<h3 class="section">4.13 How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?<span class="pull-right"><a class="anchor hidden-xs" href="#How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-How-do-I-feed-H_002e263_002dRTP-_0028and-other-codecs-in-RTP_0029-to-libavcodec_003f" aria-hidden="true">TOC</a></span></h3>
<p>Even if peculiar since it is network oriented, RTP is a container like any
other. You have to <em>demux</em> RTP before feeding the payload to libavcodec.
In this specific case please look at RFC 4629 to see how it should be done.
</p>
<a name="AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e"></a>
<h3 class="section">4.14 AVStream.r_frame_rate is wrong, it is much larger than the frame rate.<span class="pull-right"><a class="anchor hidden-xs" href="#AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVStream_002er_005fframe_005frate-is-wrong_002c-it-is-much-larger-than-the-frame-rate_002e" aria-hidden="true">TOC</a></span></h3>
<p><code>r_frame_rate</code> is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then <code>r_frame_rate</code>
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see <code>AVStream.avg_frame_rate</code>.
</p>
<a name="Why-is-make-fate-not-running-all-tests_003f"></a>
<h3 class="section">4.15 Why is <code>make fate</code> not running all tests?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-running-all-tests_003f" aria-hidden="true">TOC</a></span></h3>
<p>Make sure you have the fate-suite samples and the <code>SAMPLES</code> Make variable
or <code>FATE_SAMPLES</code> environment variable or the <code>--samples</code>
<code>configure</code> option is set to the right path.
</p>
<a name="Why-is-make-fate-not-finding-the-samples_003f"></a>
<h3 class="section">4.16 Why is <code>make fate</code> not finding the samples?<span class="pull-right"><a class="anchor hidden-xs" href="#Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Why-is-make-fate-not-finding-the-samples_003f" aria-hidden="true">TOC</a></span></h3>
<p>Do you happen to have a <code>~</code> character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace <code>~</code> by the full path.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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FFmpeg Automated Testing Environment
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FFmpeg Automated Testing Environment
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<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Introduction" href="#Introduction">1 Introduction</a></li>
<li><a name="toc-Using-FATE-from-your-FFmpeg-source-directory" href="#Using-FATE-from-your-FFmpeg-source-directory">2 Using FATE from your FFmpeg source directory</a></li>
<li><a name="toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server">3 Submitting the results to the FFmpeg result aggregation server</a></li>
<li><a name="toc-FATE-makefile-targets-and-variables" href="#FATE-makefile-targets-and-variables">4 FATE makefile targets and variables</a>
<ul class="no-bullet">
<li><a name="toc-Makefile-targets" href="#Makefile-targets">4.1 Makefile targets</a></li>
<li><a name="toc-Makefile-variables" href="#Makefile-variables">4.2 Makefile variables</a></li>
<li><a name="toc-Examples" href="#Examples">4.3 Examples</a></li>
</ul></li>
</ul>
</div>
<a name="Introduction"></a>
<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>
<p>FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
</p>
<p>The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg&rsquo;s
FATE server.
</p>
<p>In any way you can have a look at the publicly viewable FATE results
by visiting this website:
</p>
<p><a href="http://fate.ffmpeg.org/">http://fate.ffmpeg.org/</a>
</p>
<p>This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
</p>
<p>The second part of this document describes how you can run FATE to
submit your results to FFmpeg&rsquo;s FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
</p>
<p>In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
</p>
<a name="Using-FATE-from-your-FFmpeg-source-directory"></a>
<h2 class="chapter">2 Using FATE from your FFmpeg source directory<span class="pull-right"><a class="anchor hidden-xs" href="#Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-FATE-from-your-FFmpeg-source-directory" aria-hidden="true">TOC</a></span></h2>
<p>If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
</p>
<div class="example">
<pre class="example">make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
</pre></div>
<p>The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
&lsquo;&ndash;samples=&lt;path to the samples directory&gt;&rsquo;. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
</p>
<div class="example">
<pre class="example">./configure --samples=fate-suite/
make fate-rsync
make fate
</pre></div>
<p>Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
</p>
<div class="example">
<pre class="example">FATE_SAMPLES=fate-suite/ make fate
</pre></div>
<div class="info">
<p>Do not put a &rsquo;~&rsquo; character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
</p></div>
<p>To use a custom wrapper to run the test, pass <samp>--target-exec</samp> to
<code>configure</code> or set the <var>TARGET_EXEC</var> Make variable.
</p>
<a name="Submitting-the-results-to-the-FFmpeg-result-aggregation-server"></a>
<h2 class="chapter">3 Submitting the results to the FFmpeg result aggregation server<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-the-results-to-the-FFmpeg-result-aggregation-server" aria-hidden="true">TOC</a></span></h2>
<p>To submit your results to the server you should run fate through the
shell script <samp>tests/fate.sh</samp> from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
</p>
<div class="example">
<pre class="example">tests/fate.sh /path/to/fate_config
</pre></div>
<p>A configuration file template with comments describing the individual
configuration variables can be found at <samp>doc/fate_config.sh.template</samp>.
</p>
<p>The mentioned configuration template is also available here:
</p><pre class="verbatim">slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv=&quot;ssh -T fate@fate.ffmpeg.org&quot; # command to submit report
comment= # optional description
build_only= # set to &quot;yes&quot; for a compile-only instance that skips tests
# the following are optional and map to configure options
arch=
cpu=
cross_prefix=
as=
cc=
ld=
target_os=
sysroot=
target_exec=
target_path=
target_samples=
extra_cflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above
#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'
</pre>
<p>Create a configuration that suits your needs, based on the configuration
template. The &lsquo;slot&rsquo; configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern &lt;arch&gt;-&lt;os&gt;-&lt;compiler&gt;-&lt;compiler version&gt;. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
</p>
<p>For your first test runs the &lsquo;fate_recv&rsquo; variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
</p>
<ul>
<li> configure.log
</li><li> compile.log
</li><li> test.log
</li><li> report
</li><li> version
</li></ul>
<p>When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address <a href="mailto:fate-admin@ffmpeg.org">fate-admin@ffmpeg.org</a>.
</p>
<p>Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server&rsquo;s fingerprint is:
</p>
<dl compact="compact">
<dt><samp>RSA</samp></dt>
<dd><p>d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
</p></dd>
<dt><samp>ECDSA</samp></dt>
<dd><p>76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
</p></dd>
</dl>
<p>If you have problems connecting to the FATE server, it may help to try out
the <code>ssh</code> command with one or more <samp>-v</samp> options. You should
get detailed output concerning your SSH configuration and the authentication
process.
</p>
<p>The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
</p>
<a name="FATE-makefile-targets-and-variables"></a>
<h2 class="chapter">4 FATE makefile targets and variables<span class="pull-right"><a class="anchor hidden-xs" href="#FATE-makefile-targets-and-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FATE-makefile-targets-and-variables" aria-hidden="true">TOC</a></span></h2>
<a name="Makefile-targets"></a>
<h3 class="section">4.1 Makefile targets<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-targets" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-targets" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><samp>fate-rsync</samp></dt>
<dd><p>Download/synchronize sample files to the configured samples directory.
</p>
</dd>
<dt><samp>fate-list</samp></dt>
<dd><p>Will list all fate/regression test targets.
</p>
</dd>
<dt><samp>fate</samp></dt>
<dd><p>Run the FATE test suite (requires the fate-suite dataset).
</p></dd>
</dl>
<a name="Makefile-variables"></a>
<h3 class="section">4.2 Makefile variables<span class="pull-right"><a class="anchor hidden-xs" href="#Makefile-variables" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Makefile-variables" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><samp>V</samp></dt>
<dd><p>Verbosity level, can be set to 0, 1 or 2.
</p><ul>
<li> 0: show just the test arguments
</li><li> 1: show just the command used in the test
</li><li> 2: show everything
</li></ul>
</dd>
<dt><samp>SAMPLES</samp></dt>
<dd><p>Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
</p>
</dd>
<dt><samp>THREADS</samp></dt>
<dd><p>Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
</p>
</dd>
<dt><samp>THREAD_TYPE</samp></dt>
<dd><p>Specify which threading strategy test, either <var>slice</var> or <var>frame</var>,
by default <var>slice+frame</var>
</p>
</dd>
<dt><samp>CPUFLAGS</samp></dt>
<dd><p>Specify CPU flags.
</p>
</dd>
<dt><samp>TARGET_EXEC</samp></dt>
<dd><p>Specify or override the wrapper used to run the tests.
The <var>TARGET_EXEC</var> option provides a way to run FATE wrapped in
<code>valgrind</code>, <code>qemu-user</code> or <code>wine</code> or on remote targets
through <code>ssh</code>.
</p>
</dd>
<dt><samp>GEN</samp></dt>
<dd><p>Set to <var>1</var> to generate the missing or mismatched references.
</p></dd>
</dl>
<a name="Examples"></a>
<h3 class="section">4.3 Examples<span class="pull-right"><a class="anchor hidden-xs" href="#Examples" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Examples" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
</pre></div>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
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<title>
FFmpeg Bitstream Filters Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
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<h1>
FFmpeg Bitstream Filters Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-Bitstream-Filters" href="#Bitstream-Filters">2 Bitstream Filters</a>
<ul class="no-bullet">
<li><a name="toc-aac_005fadtstoasc" href="#aac_005fadtstoasc">2.1 aac_adtstoasc</a></li>
<li><a name="toc-chomp" href="#chomp">2.2 chomp</a></li>
<li><a name="toc-dump_005fextra" href="#dump_005fextra">2.3 dump_extra</a></li>
<li><a name="toc-h264_005fmp4toannexb" href="#h264_005fmp4toannexb">2.4 h264_mp4toannexb</a></li>
<li><a name="toc-imxdump" href="#imxdump">2.5 imxdump</a></li>
<li><a name="toc-mjpeg2jpeg" href="#mjpeg2jpeg">2.6 mjpeg2jpeg</a></li>
<li><a name="toc-mjpega_005fdump_005fheader" href="#mjpega_005fdump_005fheader">2.7 mjpega_dump_header</a></li>
<li><a name="toc-movsub" href="#movsub">2.8 movsub</a></li>
<li><a name="toc-mp3_005fheader_005fdecompress" href="#mp3_005fheader_005fdecompress">2.9 mp3_header_decompress</a></li>
<li><a name="toc-noise" href="#noise">2.10 noise</a></li>
<li><a name="toc-remove_005fextra" href="#remove_005fextra">2.11 remove_extra</a></li>
</ul></li>
<li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>This document describes the bitstream filters provided by the
libavcodec library.
</p>
<p>A bitstream filter operates on the encoded stream data, and performs
bitstream level modifications without performing decoding.
</p>
<a name="Bitstream-Filters"></a>
<h2 class="chapter">2 Bitstream Filters<span class="pull-right"><a class="anchor hidden-xs" href="#Bitstream-Filters" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Bitstream-Filters" aria-hidden="true">TOC</a></span></h2>
<p>When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option <code>--list-bsfs</code>.
</p>
<p>You can disable all the bitstream filters using the configure option
<code>--disable-bsfs</code>, and selectively enable any bitstream filter using
the option <code>--enable-bsf=BSF</code>, or you can disable a particular
bitstream filter using the option <code>--disable-bsf=BSF</code>.
</p>
<p>The option <code>-bsfs</code> of the ff* tools will display the list of
all the supported bitstream filters included in your build.
</p>
<p>The ff* tools have a -bsf option applied per stream, taking a
comma-separated list of filters, whose parameters follow the filter
name after a &rsquo;=&rsquo;.
</p>
<div class="example">
<pre class="example">ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
</pre></div>
<p>Below is a description of the currently available bitstream filters,
with their parameters, if any.
</p>
<a name="aac_005fadtstoasc"></a>
<h3 class="section">2.1 aac_adtstoasc<span class="pull-right"><a class="anchor hidden-xs" href="#aac_005fadtstoasc" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-aac_005fadtstoasc" aria-hidden="true">TOC</a></span></h3>
<p>Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
bitstream filter.
</p>
<p>This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
ADTS header and removes the ADTS header.
</p>
<p>This is required for example when copying an AAC stream from a raw
ADTS AAC container to a FLV or a MOV/MP4 file.
</p>
<a name="chomp"></a>
<h3 class="section">2.2 chomp<span class="pull-right"><a class="anchor hidden-xs" href="#chomp" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-chomp" aria-hidden="true">TOC</a></span></h3>
<p>Remove zero padding at the end of a packet.
</p>
<a name="dump_005fextra"></a>
<h3 class="section">2.3 dump_extra<span class="pull-right"><a class="anchor hidden-xs" href="#dump_005fextra" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-dump_005fextra" aria-hidden="true">TOC</a></span></h3>
<p>Add extradata to the beginning of the filtered packets.
</p>
<p>The additional argument specifies which packets should be filtered.
It accepts the values:
</p><dl compact="compact">
<dt>&lsquo;<samp>a</samp>&rsquo;</dt>
<dd><p>add extradata to all key packets, but only if <var>local_header</var> is
set in the <samp>flags2</samp> codec context field
</p>
</dd>
<dt>&lsquo;<samp>k</samp>&rsquo;</dt>
<dd><p>add extradata to all key packets
</p>
</dd>
<dt>&lsquo;<samp>e</samp>&rsquo;</dt>
<dd><p>add extradata to all packets
</p></dd>
</dl>
<p>If not specified it is assumed &lsquo;<samp>k</samp>&rsquo;.
</p>
<p>For example the following <code>ffmpeg</code> command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the <code>libx264</code> encoder, but corrects them by adding
the header stored in extradata to the key packets:
</p><div class="example">
<pre class="example">ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
</pre></div>
<a name="h264_005fmp4toannexb"></a>
<h3 class="section">2.4 h264_mp4toannexb<span class="pull-right"><a class="anchor hidden-xs" href="#h264_005fmp4toannexb" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-h264_005fmp4toannexb" aria-hidden="true">TOC</a></span></h3>
<p>Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
</p>
<p>This is required by some streaming formats, typically the MPEG-2
transport stream format (&quot;mpegts&quot;).
</p>
<p>For example to remux an MP4 file containing an H.264 stream to mpegts
format with <code>ffmpeg</code>, you can use the command:
</p>
<div class="example">
<pre class="example">ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
</pre></div>
<a name="imxdump"></a>
<h3 class="section">2.5 imxdump<span class="pull-right"><a class="anchor hidden-xs" href="#imxdump" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-imxdump" aria-hidden="true">TOC</a></span></h3>
<p>Modifies the bitstream to fit in MOV and to be usable by the Final Cut
Pro decoder. This filter only applies to the mpeg2video codec, and is
likely not needed for Final Cut Pro 7 and newer with the appropriate
<samp>-tag:v</samp>.
</p>
<p>For example, to remux 30 MB/sec NTSC IMX to MOV:
</p>
<div class="example">
<pre class="example">ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
</pre></div>
<a name="mjpeg2jpeg"></a>
<h3 class="section">2.6 mjpeg2jpeg<span class="pull-right"><a class="anchor hidden-xs" href="#mjpeg2jpeg" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mjpeg2jpeg" aria-hidden="true">TOC</a></span></h3>
<p>Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
</p>
<p>MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
</p>
<div class="example">
<pre class="example">ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
</pre></div>
<p>Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
<a href="http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml">http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml</a>:
</p>
<p>Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that &quot;MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed &ndash; and *omitted* &ndash;
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won&rsquo;t have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec.&quot;
</p>
<p>This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
</p>
<div class="example">
<pre class="example">ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
</pre></div>
<a name="mjpega_005fdump_005fheader"></a>
<h3 class="section">2.7 mjpega_dump_header<span class="pull-right"><a class="anchor hidden-xs" href="#mjpega_005fdump_005fheader" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mjpega_005fdump_005fheader" aria-hidden="true">TOC</a></span></h3>
<a name="movsub"></a>
<h3 class="section">2.8 movsub<span class="pull-right"><a class="anchor hidden-xs" href="#movsub" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-movsub" aria-hidden="true">TOC</a></span></h3>
<a name="mp3_005fheader_005fdecompress"></a>
<h3 class="section">2.9 mp3_header_decompress<span class="pull-right"><a class="anchor hidden-xs" href="#mp3_005fheader_005fdecompress" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-mp3_005fheader_005fdecompress" aria-hidden="true">TOC</a></span></h3>
<a name="noise"></a>
<h3 class="section">2.10 noise<span class="pull-right"><a class="anchor hidden-xs" href="#noise" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-noise" aria-hidden="true">TOC</a></span></h3>
<p>Damages the contents of packets without damaging the container. Can be
used for fuzzing or testing error resilience/concealment.
</p>
<p>Parameters:
A numeral string, whose value is related to how often output bytes will
be modified. Therefore, values below or equal to 0 are forbidden, and
the lower the more frequent bytes will be modified, with 1 meaning
every byte is modified.
</p>
<div class="example">
<pre class="example">ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
</pre></div>
<p>applies the modification to every byte.
</p>
<a name="remove_005fextra"></a>
<h3 class="section">2.11 remove_extra<span class="pull-right"><a class="anchor hidden-xs" href="#remove_005fextra" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-remove_005fextra" aria-hidden="true">TOC</a></span></h3>
<a name="See-Also"></a>
<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="libavcodec.html">libavcodec</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
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<title>
FFmpeg Resampler Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
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<body>
<div style="width: 95%; margin: auto">
<h1>
FFmpeg Resampler Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-Resampler-Options" href="#Resampler-Options">2 Resampler Options</a></li>
<li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg resampler provides a high-level interface to the
libswresample library audio resampling utilities. In particular it
allows one to perform audio resampling, audio channel layout rematrixing,
and convert audio format and packing layout.
</p>
<a name="Resampler-Options"></a>
<h2 class="chapter">2 Resampler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Resampler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Resampler-Options" aria-hidden="true">TOC</a></span></h2>
<p>The audio resampler supports the following named options.
</p>
<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
FFmpeg tools, <var>option</var>=<var>value</var> for the aresample filter,
by setting the value explicitly in the
<code>SwrContext</code> options or using the <samp>libavutil/opt.h</samp> API for
programmatic use.
</p>
<dl compact="compact">
<dt><samp>ich, in_channel_count</samp></dt>
<dd><p>Set the number of input channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
<samp>in_channel_layout</samp> is set.
</p>
</dd>
<dt><samp>och, out_channel_count</samp></dt>
<dd><p>Set the number of output channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
<samp>out_channel_layout</samp> is set.
</p>
</dd>
<dt><samp>uch, used_channel_count</samp></dt>
<dd><p>Set the number of used input channels. Default value is 0. This option is
only used for special remapping.
</p>
</dd>
<dt><samp>isr, in_sample_rate</samp></dt>
<dd><p>Set the input sample rate. Default value is 0.
</p>
</dd>
<dt><samp>osr, out_sample_rate</samp></dt>
<dd><p>Set the output sample rate. Default value is 0.
</p>
</dd>
<dt><samp>isf, in_sample_fmt</samp></dt>
<dd><p>Specify the input sample format. It is set by default to <code>none</code>.
</p>
</dd>
<dt><samp>osf, out_sample_fmt</samp></dt>
<dd><p>Specify the output sample format. It is set by default to <code>none</code>.
</p>
</dd>
<dt><samp>tsf, internal_sample_fmt</samp></dt>
<dd><p>Set the internal sample format. Default value is <code>none</code>.
This will automatically be chosen when it is not explicitly set.
</p>
</dd>
<dt><samp>icl, in_channel_layout</samp></dt>
<dt><samp>ocl, out_channel_layout</samp></dt>
<dd><p>Set the input/output channel layout.
</p>
<p>See <a href="ffmpeg-utils.html#channel-layout-syntax">(ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual</a>
for the required syntax.
</p>
</dd>
<dt><samp>clev, center_mix_level</samp></dt>
<dd><p>Set the center mix level. It is a value expressed in deciBel, and must be
in the interval [-32,32].
</p>
</dd>
<dt><samp>slev, surround_mix_level</samp></dt>
<dd><p>Set the surround mix level. It is a value expressed in deciBel, and must
be in the interval [-32,32].
</p>
</dd>
<dt><samp>lfe_mix_level</samp></dt>
<dd><p>Set LFE mix into non LFE level. It is used when there is a LFE input but no
LFE output. It is a value expressed in deciBel, and must
be in the interval [-32,32].
</p>
</dd>
<dt><samp>rmvol, rematrix_volume</samp></dt>
<dd><p>Set rematrix volume. Default value is 1.0.
</p>
</dd>
<dt><samp>rematrix_maxval</samp></dt>
<dd><p>Set maximum output value for rematrixing.
This can be used to prevent clipping vs. preventing volumn reduction
A value of 1.0 prevents cliping.
</p>
</dd>
<dt><samp>flags, swr_flags</samp></dt>
<dd><p>Set flags used by the converter. Default value is 0.
</p>
<p>It supports the following individual flags:
</p><dl compact="compact">
<dt><samp>res</samp></dt>
<dd><p>force resampling, this flag forces resampling to be used even when the
input and output sample rates match.
</p></dd>
</dl>
</dd>
<dt><samp>dither_scale</samp></dt>
<dd><p>Set the dither scale. Default value is 1.
</p>
</dd>
<dt><samp>dither_method</samp></dt>
<dd><p>Set dither method. Default value is 0.
</p>
<p>Supported values:
</p><dl compact="compact">
<dt>&lsquo;<samp>rectangular</samp>&rsquo;</dt>
<dd><p>select rectangular dither
</p></dd>
<dt>&lsquo;<samp>triangular</samp>&rsquo;</dt>
<dd><p>select triangular dither
</p></dd>
<dt>&lsquo;<samp>triangular_hp</samp>&rsquo;</dt>
<dd><p>select triangular dither with high pass
</p></dd>
<dt>&lsquo;<samp>lipshitz</samp>&rsquo;</dt>
<dd><p>select lipshitz noise shaping dither
</p></dd>
<dt>&lsquo;<samp>shibata</samp>&rsquo;</dt>
<dd><p>select shibata noise shaping dither
</p></dd>
<dt>&lsquo;<samp>low_shibata</samp>&rsquo;</dt>
<dd><p>select low shibata noise shaping dither
</p></dd>
<dt>&lsquo;<samp>high_shibata</samp>&rsquo;</dt>
<dd><p>select high shibata noise shaping dither
</p></dd>
<dt>&lsquo;<samp>f_weighted</samp>&rsquo;</dt>
<dd><p>select f-weighted noise shaping dither
</p></dd>
<dt>&lsquo;<samp>modified_e_weighted</samp>&rsquo;</dt>
<dd><p>select modified-e-weighted noise shaping dither
</p></dd>
<dt>&lsquo;<samp>improved_e_weighted</samp>&rsquo;</dt>
<dd><p>select improved-e-weighted noise shaping dither
</p>
</dd>
</dl>
</dd>
<dt><samp>resampler</samp></dt>
<dd><p>Set resampling engine. Default value is swr.
</p>
<p>Supported values:
</p><dl compact="compact">
<dt>&lsquo;<samp>swr</samp>&rsquo;</dt>
<dd><p>select the native SW Resampler; filter options precision and cheby are not
applicable in this case.
</p></dd>
<dt>&lsquo;<samp>soxr</samp>&rsquo;</dt>
<dd><p>select the SoX Resampler (where available); compensation, and filter options
filter_size, phase_shift, filter_type &amp; kaiser_beta, are not applicable in this
case.
</p></dd>
</dl>
</dd>
<dt><samp>filter_size</samp></dt>
<dd><p>For swr only, set resampling filter size, default value is 32.
</p>
</dd>
<dt><samp>phase_shift</samp></dt>
<dd><p>For swr only, set resampling phase shift, default value is 10, and must be in
the interval [0,30].
</p>
</dd>
<dt><samp>linear_interp</samp></dt>
<dd><p>Use Linear Interpolation if set to 1, default value is 0.
</p>
</dd>
<dt><samp>cutoff</samp></dt>
<dd><p>Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
</p>
</dd>
<dt><samp>precision</samp></dt>
<dd><p>For soxr only, the precision in bits to which the resampled signal will be
calculated. The default value of 20 (which, with suitable dithering, is
appropriate for a destination bit-depth of 16) gives SoX&rsquo;s &rsquo;High Quality&rsquo;; a
value of 28 gives SoX&rsquo;s &rsquo;Very High Quality&rsquo;.
</p>
</dd>
<dt><samp>cheby</samp></dt>
<dd><p>For soxr only, selects passband rolloff none (Chebyshev) &amp; higher-precision
approximation for &rsquo;irrational&rsquo; ratios. Default value is 0.
</p>
</dd>
<dt><samp>async</samp></dt>
<dd><p>For swr only, simple 1 parameter audio sync to timestamps using stretching,
squeezing, filling and trimming. Setting this to 1 will enable filling and
trimming, larger values represent the maximum amount in samples that the data
may be stretched or squeezed for each second.
Default value is 0, thus no compensation is applied to make the samples match
the audio timestamps.
</p>
</dd>
<dt><samp>first_pts</samp></dt>
<dd><p>For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
This allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame&rsquo;s expected pts, so no padding or
trimming is done. For example, this could be set to 0 to pad the beginning with
silence if an audio stream starts after the video stream or to trim any samples
with a negative pts due to encoder delay.
</p>
</dd>
<dt><samp>min_comp</samp></dt>
<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger stretching/squeezing/filling or trimming of the
data to make it match the timestamps. The default is that
stretching/squeezing/filling and trimming is disabled
(<samp>min_comp</samp> = <code>FLT_MAX</code>).
</p>
</dd>
<dt><samp>min_hard_comp</samp></dt>
<dd><p>For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger adding/dropping samples to make it match the
timestamps. This option effectively is a threshold to select between
hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
all compensation is by default disabled through <samp>min_comp</samp>.
The default is 0.1.
</p>
</dd>
<dt><samp>comp_duration</samp></dt>
<dd><p>For swr only, set duration (in seconds) over which data is stretched/squeezed
to make it match the timestamps. Must be a non-negative double float value,
default value is 1.0.
</p>
</dd>
<dt><samp>max_soft_comp</samp></dt>
<dd><p>For swr only, set maximum factor by which data is stretched/squeezed to make it
match the timestamps. Must be a non-negative double float value, default value
is 0.
</p>
</dd>
<dt><samp>matrix_encoding</samp></dt>
<dd><p>Select matrixed stereo encoding.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt>&lsquo;<samp>none</samp>&rsquo;</dt>
<dd><p>select none
</p></dd>
<dt>&lsquo;<samp>dolby</samp>&rsquo;</dt>
<dd><p>select Dolby
</p></dd>
<dt>&lsquo;<samp>dplii</samp>&rsquo;</dt>
<dd><p>select Dolby Pro Logic II
</p></dd>
</dl>
<p>Default value is <code>none</code>.
</p>
</dd>
<dt><samp>filter_type</samp></dt>
<dd><p>For swr only, select resampling filter type. This only affects resampling
operations.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt>&lsquo;<samp>cubic</samp>&rsquo;</dt>
<dd><p>select cubic
</p></dd>
<dt>&lsquo;<samp>blackman_nuttall</samp>&rsquo;</dt>
<dd><p>select Blackman Nuttall Windowed Sinc
</p></dd>
<dt>&lsquo;<samp>kaiser</samp>&rsquo;</dt>
<dd><p>select Kaiser Windowed Sinc
</p></dd>
</dl>
</dd>
<dt><samp>kaiser_beta</samp></dt>
<dd><p>For swr only, set Kaiser Window Beta value. Must be an integer in the
interval [2,16], default value is 9.
</p>
</dd>
<dt><samp>output_sample_bits</samp></dt>
<dd><p>For swr only, set number of used output sample bits for dithering. Must be an integer in the
interval [0,64], default value is 0, which means it&rsquo;s not used.
</p>
</dd>
</dl>
<a name="See-Also"></a>
<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="libswresample.html">libswresample</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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<html>
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<head>
<meta charset="utf-8">
<title>
FFmpeg Scaler Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
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<body>
<div style="width: 95%; margin: auto">
<h1>
FFmpeg Scaler Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-Scaler-Options" href="#Scaler-Options">2 Scaler Options</a></li>
<li><a name="toc-See-Also" href="#See-Also">3 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">4 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg rescaler provides a high-level interface to the libswscale
library image conversion utilities. In particular it allows one to perform
image rescaling and pixel format conversion.
</p>
<a name="scaler_005foptions"></a><a name="Scaler-Options"></a>
<h2 class="chapter">2 Scaler Options<span class="pull-right"><a class="anchor hidden-xs" href="#Scaler-Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Scaler-Options" aria-hidden="true">TOC</a></span></h2>
<p>The video scaler supports the following named options.
</p>
<p>Options may be set by specifying -<var>option</var> <var>value</var> in the
FFmpeg tools. For programmatic use, they can be set explicitly in the
<code>SwsContext</code> options or through the <samp>libavutil/opt.h</samp> API.
</p>
<dl compact="compact">
<dd>
<a name="sws_005fflags"></a></dd>
<dt><samp>sws_flags</samp></dt>
<dd><p>Set the scaler flags. This is also used to set the scaling
algorithm. Only a single algorithm should be selected.
</p>
<p>It accepts the following values:
</p><dl compact="compact">
<dt>&lsquo;<samp>fast_bilinear</samp>&rsquo;</dt>
<dd><p>Select fast bilinear scaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>bilinear</samp>&rsquo;</dt>
<dd><p>Select bilinear scaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>bicubic</samp>&rsquo;</dt>
<dd><p>Select bicubic scaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>experimental</samp>&rsquo;</dt>
<dd><p>Select experimental scaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>neighbor</samp>&rsquo;</dt>
<dd><p>Select nearest neighbor rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>area</samp>&rsquo;</dt>
<dd><p>Select averaging area rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>bicublin</samp>&rsquo;</dt>
<dd><p>Select bicubic scaling algorithm for the luma component, bilinear for
chroma components.
</p>
</dd>
<dt>&lsquo;<samp>gauss</samp>&rsquo;</dt>
<dd><p>Select Gaussian rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>sinc</samp>&rsquo;</dt>
<dd><p>Select sinc rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>lanczos</samp>&rsquo;</dt>
<dd><p>Select lanczos rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>spline</samp>&rsquo;</dt>
<dd><p>Select natural bicubic spline rescaling algorithm.
</p>
</dd>
<dt>&lsquo;<samp>print_info</samp>&rsquo;</dt>
<dd><p>Enable printing/debug logging.
</p>
</dd>
<dt>&lsquo;<samp>accurate_rnd</samp>&rsquo;</dt>
<dd><p>Enable accurate rounding.
</p>
</dd>
<dt>&lsquo;<samp>full_chroma_int</samp>&rsquo;</dt>
<dd><p>Enable full chroma interpolation.
</p>
</dd>
<dt>&lsquo;<samp>full_chroma_inp</samp>&rsquo;</dt>
<dd><p>Select full chroma input.
</p>
</dd>
<dt>&lsquo;<samp>bitexact</samp>&rsquo;</dt>
<dd><p>Enable bitexact output.
</p></dd>
</dl>
</dd>
<dt><samp>srcw</samp></dt>
<dd><p>Set source width.
</p>
</dd>
<dt><samp>srch</samp></dt>
<dd><p>Set source height.
</p>
</dd>
<dt><samp>dstw</samp></dt>
<dd><p>Set destination width.
</p>
</dd>
<dt><samp>dsth</samp></dt>
<dd><p>Set destination height.
</p>
</dd>
<dt><samp>src_format</samp></dt>
<dd><p>Set source pixel format (must be expressed as an integer).
</p>
</dd>
<dt><samp>dst_format</samp></dt>
<dd><p>Set destination pixel format (must be expressed as an integer).
</p>
</dd>
<dt><samp>src_range</samp></dt>
<dd><p>Select source range.
</p>
</dd>
<dt><samp>dst_range</samp></dt>
<dd><p>Select destination range.
</p>
</dd>
<dt><samp>param0, param1</samp></dt>
<dd><p>Set scaling algorithm parameters. The specified values are specific of
some scaling algorithms and ignored by others. The specified values
are floating point number values.
</p>
</dd>
<dt><samp>sws_dither</samp></dt>
<dd><p>Set the dithering algorithm. Accepts one of the following
values. Default value is &lsquo;<samp>auto</samp>&rsquo;.
</p>
<dl compact="compact">
<dt>&lsquo;<samp>auto</samp>&rsquo;</dt>
<dd><p>automatic choice
</p>
</dd>
<dt>&lsquo;<samp>none</samp>&rsquo;</dt>
<dd><p>no dithering
</p>
</dd>
<dt>&lsquo;<samp>bayer</samp>&rsquo;</dt>
<dd><p>bayer dither
</p>
</dd>
<dt>&lsquo;<samp>ed</samp>&rsquo;</dt>
<dd><p>error diffusion dither
</p>
</dd>
<dt>&lsquo;<samp>a_dither</samp>&rsquo;</dt>
<dd><p>arithmetic dither, based using addition
</p>
</dd>
<dt>&lsquo;<samp>x_dither</samp>&rsquo;</dt>
<dd><p>arithmetic dither, based using xor (more random/less apparent patterning that
a_dither).
</p>
</dd>
</dl>
</dd>
</dl>
<a name="See-Also"></a>
<h2 class="chapter">3 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="libswscale.html">libswscale</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">4 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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<html>
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<title>
ffplay Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
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<link rel="stylesheet" type="text/css" href="style.min.css">
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<div style="width: 95%; margin: auto">
<h1>
ffplay Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Synopsis" href="#Synopsis">1 Synopsis</a></li>
<li><a name="toc-Description" href="#Description">2 Description</a></li>
<li><a name="toc-Options" href="#Options">3 Options</a>
<ul class="no-bullet">
<li><a name="toc-Stream-specifiers-1" href="#Stream-specifiers-1">3.1 Stream specifiers</a></li>
<li><a name="toc-Generic-options" href="#Generic-options">3.2 Generic options</a></li>
<li><a name="toc-AVOptions" href="#AVOptions">3.3 AVOptions</a></li>
<li><a name="toc-Main-options" href="#Main-options">3.4 Main options</a></li>
<li><a name="toc-Advanced-options" href="#Advanced-options">3.5 Advanced options</a></li>
<li><a name="toc-While-playing" href="#While-playing">3.6 While playing</a></li>
</ul></li>
<li><a name="toc-See-Also" href="#See-Also">4 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">5 Authors</a></li>
</ul>
</div>
<a name="Synopsis"></a>
<h2 class="chapter">1 Synopsis<span class="pull-right"><a class="anchor hidden-xs" href="#Synopsis" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Synopsis" aria-hidden="true">TOC</a></span></h2>
<p>ffplay [<var>options</var>] [<samp>input_file</samp>]
</p>
<a name="Description"></a>
<h2 class="chapter">2 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>FFplay is a very simple and portable media player using the FFmpeg
libraries and the SDL library. It is mostly used as a testbed for the
various FFmpeg APIs.
</p>
<a name="Options"></a>
<h2 class="chapter">3 Options<span class="pull-right"><a class="anchor hidden-xs" href="#Options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Options" aria-hidden="true">TOC</a></span></h2>
<p>All the numerical options, if not specified otherwise, accept a string
representing a number as input, which may be followed by one of the SI
unit prefixes, for example: &rsquo;K&rsquo;, &rsquo;M&rsquo;, or &rsquo;G&rsquo;.
</p>
<p>If &rsquo;i&rsquo; is appended to the SI unit prefix, the complete prefix will be
interpreted as a unit prefix for binary multiples, which are based on
powers of 1024 instead of powers of 1000. Appending &rsquo;B&rsquo; to the SI unit
prefix multiplies the value by 8. This allows using, for example:
&rsquo;KB&rsquo;, &rsquo;MiB&rsquo;, &rsquo;G&rsquo; and &rsquo;B&rsquo; as number suffixes.
</p>
<p>Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
the option name with &quot;no&quot;. For example using &quot;-nofoo&quot;
will set the boolean option with name &quot;foo&quot; to false.
</p>
<a name="Stream-specifiers"></a><a name="Stream-specifiers-1"></a>
<h3 class="section">3.1 Stream specifiers<span class="pull-right"><a class="anchor hidden-xs" href="#Stream-specifiers-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Stream-specifiers-1" aria-hidden="true">TOC</a></span></h3>
<p>Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) a given option belongs to.
</p>
<p>A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. <code>-codec:a:1 ac3</code> contains the
<code>a:1</code> stream specifier, which matches the second audio stream. Therefore, it
would select the ac3 codec for the second audio stream.
</p>
<p>A stream specifier can match several streams, so that the option is applied to all
of them. E.g. the stream specifier in <code>-b:a 128k</code> matches all audio
streams.
</p>
<p>An empty stream specifier matches all streams. For example, <code>-codec copy</code>
or <code>-codec: copy</code> would copy all the streams without reencoding.
</p>
<p>Possible forms of stream specifiers are:
</p><dl compact="compact">
<dt><samp><var>stream_index</var></samp></dt>
<dd><p>Matches the stream with this index. E.g. <code>-threads:1 4</code> would set the
thread count for the second stream to 4.
</p></dd>
<dt><samp><var>stream_type</var>[:<var>stream_index</var>]</samp></dt>
<dd><p><var>stream_type</var> is one of following: &rsquo;v&rsquo; for video, &rsquo;a&rsquo; for audio, &rsquo;s&rsquo; for subtitle,
&rsquo;d&rsquo; for data, and &rsquo;t&rsquo; for attachments. If <var>stream_index</var> is given, then it matches
stream number <var>stream_index</var> of this type. Otherwise, it matches all
streams of this type.
</p></dd>
<dt><samp>p:<var>program_id</var>[:<var>stream_index</var>]</samp></dt>
<dd><p>If <var>stream_index</var> is given, then it matches the stream with number <var>stream_index</var>
in the program with the id <var>program_id</var>. Otherwise, it matches all streams in the
program.
</p></dd>
<dt><samp>#<var>stream_id</var> or i:<var>stream_id</var></samp></dt>
<dd><p>Match the stream by stream id (e.g. PID in MPEG-TS container).
</p></dd>
<dt><samp>m:<var>key</var>[:<var>value</var>]</samp></dt>
<dd><p>Matches streams with the metadata tag <var>key</var> having the specified value. If
<var>value</var> is not given, matches streams that contain the given tag with any
value.
</p>
<p>Note that in <code>ffmpeg</code>, matching by metadata will only work properly for
input files.
</p></dd>
</dl>
<a name="Generic-options"></a>
<h3 class="section">3.2 Generic options<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-options" aria-hidden="true">TOC</a></span></h3>
<p>These options are shared amongst the ff* tools.
</p>
<dl compact="compact">
<dt><samp>-L</samp></dt>
<dd><p>Show license.
</p>
</dd>
<dt><samp>-h, -?, -help, --help [<var>arg</var>]</samp></dt>
<dd><p>Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
</p>
<p>Possible values of <var>arg</var> are:
</p><dl compact="compact">
<dt><samp>long</samp></dt>
<dd><p>Print advanced tool options in addition to the basic tool options.
</p>
</dd>
<dt><samp>full</samp></dt>
<dd><p>Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
</p>
</dd>
<dt><samp>decoder=<var>decoder_name</var></samp></dt>
<dd><p>Print detailed information about the decoder named <var>decoder_name</var>. Use the
<samp>-decoders</samp> option to get a list of all decoders.
</p>
</dd>
<dt><samp>encoder=<var>encoder_name</var></samp></dt>
<dd><p>Print detailed information about the encoder named <var>encoder_name</var>. Use the
<samp>-encoders</samp> option to get a list of all encoders.
</p>
</dd>
<dt><samp>demuxer=<var>demuxer_name</var></samp></dt>
<dd><p>Print detailed information about the demuxer named <var>demuxer_name</var>. Use the
<samp>-formats</samp> option to get a list of all demuxers and muxers.
</p>
</dd>
<dt><samp>muxer=<var>muxer_name</var></samp></dt>
<dd><p>Print detailed information about the muxer named <var>muxer_name</var>. Use the
<samp>-formats</samp> option to get a list of all muxers and demuxers.
</p>
</dd>
<dt><samp>filter=<var>filter_name</var></samp></dt>
<dd><p>Print detailed information about the filter name <var>filter_name</var>. Use the
<samp>-filters</samp> option to get a list of all filters.
</p></dd>
</dl>
</dd>
<dt><samp>-version</samp></dt>
<dd><p>Show version.
</p>
</dd>
<dt><samp>-formats</samp></dt>
<dd><p>Show available formats (including devices).
</p>
</dd>
<dt><samp>-devices</samp></dt>
<dd><p>Show available devices.
</p>
</dd>
<dt><samp>-codecs</samp></dt>
<dd><p>Show all codecs known to libavcodec.
</p>
<p>Note that the term &rsquo;codec&rsquo; is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
</p>
</dd>
<dt><samp>-decoders</samp></dt>
<dd><p>Show available decoders.
</p>
</dd>
<dt><samp>-encoders</samp></dt>
<dd><p>Show all available encoders.
</p>
</dd>
<dt><samp>-bsfs</samp></dt>
<dd><p>Show available bitstream filters.
</p>
</dd>
<dt><samp>-protocols</samp></dt>
<dd><p>Show available protocols.
</p>
</dd>
<dt><samp>-filters</samp></dt>
<dd><p>Show available libavfilter filters.
</p>
</dd>
<dt><samp>-pix_fmts</samp></dt>
<dd><p>Show available pixel formats.
</p>
</dd>
<dt><samp>-sample_fmts</samp></dt>
<dd><p>Show available sample formats.
</p>
</dd>
<dt><samp>-layouts</samp></dt>
<dd><p>Show channel names and standard channel layouts.
</p>
</dd>
<dt><samp>-colors</samp></dt>
<dd><p>Show recognized color names.
</p>
</dd>
<dt><samp>-sources <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></dt>
<dd><p>Show autodetected sources of the intput device.
Some devices may provide system-dependent source names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
</p><div class="example">
<pre class="example">ffmpeg -sources pulse,server=192.168.0.4
</pre></div>
</dd>
<dt><samp>-sinks <var>device</var>[,<var>opt1</var>=<var>val1</var>[,<var>opt2</var>=<var>val2</var>]...]</samp></dt>
<dd><p>Show autodetected sinks of the output device.
Some devices may provide system-dependent sink names that cannot be autodetected.
The returned list cannot be assumed to be always complete.
</p><div class="example">
<pre class="example">ffmpeg -sinks pulse,server=192.168.0.4
</pre></div>
</dd>
<dt><samp>-loglevel [repeat+]<var>loglevel</var> | -v [repeat+]<var>loglevel</var></samp></dt>
<dd><p>Set the logging level used by the library.
Adding &quot;repeat+&quot; indicates that repeated log output should not be compressed
to the first line and the &quot;Last message repeated n times&quot; line will be
omitted. &quot;repeat&quot; can also be used alone.
If &quot;repeat&quot; is used alone, and with no prior loglevel set, the default
loglevel will be used. If multiple loglevel parameters are given, using
&rsquo;repeat&rsquo; will not change the loglevel.
<var>loglevel</var> is a string or a number containing one of the following values:
</p><dl compact="compact">
<dt>&lsquo;<samp>quiet, -8</samp>&rsquo;</dt>
<dd><p>Show nothing at all; be silent.
</p></dd>
<dt>&lsquo;<samp>panic, 0</samp>&rsquo;</dt>
<dd><p>Only show fatal errors which could lead the process to crash, such as
and assert failure. This is not currently used for anything.
</p></dd>
<dt>&lsquo;<samp>fatal, 8</samp>&rsquo;</dt>
<dd><p>Only show fatal errors. These are errors after which the process absolutely
cannot continue after.
</p></dd>
<dt>&lsquo;<samp>error, 16</samp>&rsquo;</dt>
<dd><p>Show all errors, including ones which can be recovered from.
</p></dd>
<dt>&lsquo;<samp>warning, 24</samp>&rsquo;</dt>
<dd><p>Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
</p></dd>
<dt>&lsquo;<samp>info, 32</samp>&rsquo;</dt>
<dd><p>Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
</p></dd>
<dt>&lsquo;<samp>verbose, 40</samp>&rsquo;</dt>
<dd><p>Same as <code>info</code>, except more verbose.
</p></dd>
<dt>&lsquo;<samp>debug, 48</samp>&rsquo;</dt>
<dd><p>Show everything, including debugging information.
</p></dd>
</dl>
<p>By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
<code>AV_LOG_FORCE_NOCOLOR</code> or <code>NO_COLOR</code>, or can be forced setting
the environment variable <code>AV_LOG_FORCE_COLOR</code>.
The use of the environment variable <code>NO_COLOR</code> is deprecated and
will be dropped in a following FFmpeg version.
</p>
</dd>
<dt><samp>-report</samp></dt>
<dd><p>Dump full command line and console output to a file named
<code><var>program</var>-<var>YYYYMMDD</var>-<var>HHMMSS</var>.log</code> in the current
directory.
This file can be useful for bug reports.
It also implies <code>-loglevel verbose</code>.
</p>
<p>Setting the environment variable <code>FFREPORT</code> to any value has the
same effect. If the value is a &rsquo;:&rsquo;-separated key=value sequence, these
options will affect the report; option values must be escaped if they
contain special characters or the options delimiter &rsquo;:&rsquo; (see the
&ldquo;Quoting and escaping&rdquo; section in the ffmpeg-utils manual).
</p>
<p>The following options are recognized:
</p><dl compact="compact">
<dt><samp>file</samp></dt>
<dd><p>set the file name to use for the report; <code>%p</code> is expanded to the name
of the program, <code>%t</code> is expanded to a timestamp, <code>%%</code> is expanded
to a plain <code>%</code>
</p></dd>
<dt><samp>level</samp></dt>
<dd><p>set the log verbosity level using a numerical value (see <code>-loglevel</code>).
</p></dd>
</dl>
<p>For example, to output a report to a file named <samp>ffreport.log</samp>
using a log level of <code>32</code> (alias for log level <code>info</code>):
</p>
<div class="example">
<pre class="example">FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
</pre></div>
<p>Errors in parsing the environment variable are not fatal, and will not
appear in the report.
</p>
</dd>
<dt><samp>-hide_banner</samp></dt>
<dd><p>Suppress printing banner.
</p>
<p>All FFmpeg tools will normally show a copyright notice, build options
and library versions. This option can be used to suppress printing
this information.
</p>
</dd>
<dt><samp>-cpuflags flags (<em>global</em>)</samp></dt>
<dd><p>Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you&rsquo;re doing.
</p><div class="example">
<pre class="example">ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
</pre></div>
<p>Possible flags for this option are:
</p><dl compact="compact">
<dt>&lsquo;<samp>x86</samp>&rsquo;</dt>
<dd><dl compact="compact">
<dt>&lsquo;<samp>mmx</samp>&rsquo;</dt>
<dt>&lsquo;<samp>mmxext</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse2</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse2slow</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse3</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse3slow</samp>&rsquo;</dt>
<dt>&lsquo;<samp>ssse3</samp>&rsquo;</dt>
<dt>&lsquo;<samp>atom</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse4.1</samp>&rsquo;</dt>
<dt>&lsquo;<samp>sse4.2</samp>&rsquo;</dt>
<dt>&lsquo;<samp>avx</samp>&rsquo;</dt>
<dt>&lsquo;<samp>xop</samp>&rsquo;</dt>
<dt>&lsquo;<samp>fma4</samp>&rsquo;</dt>
<dt>&lsquo;<samp>3dnow</samp>&rsquo;</dt>
<dt>&lsquo;<samp>3dnowext</samp>&rsquo;</dt>
<dt>&lsquo;<samp>cmov</samp>&rsquo;</dt>
</dl>
</dd>
<dt>&lsquo;<samp>ARM</samp>&rsquo;</dt>
<dd><dl compact="compact">
<dt>&lsquo;<samp>armv5te</samp>&rsquo;</dt>
<dt>&lsquo;<samp>armv6</samp>&rsquo;</dt>
<dt>&lsquo;<samp>armv6t2</samp>&rsquo;</dt>
<dt>&lsquo;<samp>vfp</samp>&rsquo;</dt>
<dt>&lsquo;<samp>vfpv3</samp>&rsquo;</dt>
<dt>&lsquo;<samp>neon</samp>&rsquo;</dt>
</dl>
</dd>
<dt>&lsquo;<samp>PowerPC</samp>&rsquo;</dt>
<dd><dl compact="compact">
<dt>&lsquo;<samp>altivec</samp>&rsquo;</dt>
</dl>
</dd>
<dt>&lsquo;<samp>Specific Processors</samp>&rsquo;</dt>
<dd><dl compact="compact">
<dt>&lsquo;<samp>pentium2</samp>&rsquo;</dt>
<dt>&lsquo;<samp>pentium3</samp>&rsquo;</dt>
<dt>&lsquo;<samp>pentium4</samp>&rsquo;</dt>
<dt>&lsquo;<samp>k6</samp>&rsquo;</dt>
<dt>&lsquo;<samp>k62</samp>&rsquo;</dt>
<dt>&lsquo;<samp>athlon</samp>&rsquo;</dt>
<dt>&lsquo;<samp>athlonxp</samp>&rsquo;</dt>
<dt>&lsquo;<samp>k8</samp>&rsquo;</dt>
</dl>
</dd>
</dl>
</dd>
<dt><samp>-opencl_bench</samp></dt>
<dd><p>Benchmark all available OpenCL devices and show the results. This option
is only available when FFmpeg has been compiled with <code>--enable-opencl</code>.
</p>
</dd>
<dt><samp>-opencl_options options (<em>global</em>)</samp></dt>
<dd><p>Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with <code>--enable-opencl</code>.
</p>
<p><var>options</var> must be a list of <var>key</var>=<var>value</var> option pairs
separated by &rsquo;:&rsquo;. See the &ldquo;OpenCL Options&rdquo; section in the
ffmpeg-utils manual for the list of supported options.
</p></dd>
</dl>
<a name="AVOptions"></a>
<h3 class="section">3.3 AVOptions<span class="pull-right"><a class="anchor hidden-xs" href="#AVOptions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AVOptions" aria-hidden="true">TOC</a></span></h3>
<p>These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
<samp>-help</samp> option. They are separated into two categories:
</p><dl compact="compact">
<dt><samp>generic</samp></dt>
<dd><p>These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
</p></dd>
<dt><samp>private</samp></dt>
<dd><p>These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
</p></dd>
</dl>
<p>For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the <samp>id3v2_version</samp> private option of the MP3
muxer:
</p><div class="example">
<pre class="example">ffmpeg -i input.flac -id3v2_version 3 out.mp3
</pre></div>
<p>All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them.
</p>
<p>Note: the <samp>-nooption</samp> syntax cannot be used for boolean
AVOptions, use <samp>-option 0</samp>/<samp>-option 1</samp>.
</p>
<p>Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.
</p>
<a name="Main-options"></a>
<h3 class="section">3.4 Main options<span class="pull-right"><a class="anchor hidden-xs" href="#Main-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Main-options" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><samp>-x <var>width</var></samp></dt>
<dd><p>Force displayed width.
</p></dd>
<dt><samp>-y <var>height</var></samp></dt>
<dd><p>Force displayed height.
</p></dd>
<dt><samp>-s <var>size</var></samp></dt>
<dd><p>Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
</p></dd>
<dt><samp>-fs</samp></dt>
<dd><p>Start in fullscreen mode.
</p></dd>
<dt><samp>-an</samp></dt>
<dd><p>Disable audio.
</p></dd>
<dt><samp>-vn</samp></dt>
<dd><p>Disable video.
</p></dd>
<dt><samp>-sn</samp></dt>
<dd><p>Disable subtitles.
</p></dd>
<dt><samp>-ss <var>pos</var></samp></dt>
<dd><p>Seek to a given position in seconds.
</p></dd>
<dt><samp>-t <var>duration</var></samp></dt>
<dd><p>play &lt;duration&gt; seconds of audio/video
</p></dd>
<dt><samp>-bytes</samp></dt>
<dd><p>Seek by bytes.
</p></dd>
<dt><samp>-nodisp</samp></dt>
<dd><p>Disable graphical display.
</p></dd>
<dt><samp>-f <var>fmt</var></samp></dt>
<dd><p>Force format.
</p></dd>
<dt><samp>-window_title <var>title</var></samp></dt>
<dd><p>Set window title (default is the input filename).
</p></dd>
<dt><samp>-loop <var>number</var></samp></dt>
<dd><p>Loops movie playback &lt;number&gt; times. 0 means forever.
</p></dd>
<dt><samp>-showmode <var>mode</var></samp></dt>
<dd><p>Set the show mode to use.
Available values for <var>mode</var> are:
</p><dl compact="compact">
<dt>&lsquo;<samp>0, video</samp>&rsquo;</dt>
<dd><p>show video
</p></dd>
<dt>&lsquo;<samp>1, waves</samp>&rsquo;</dt>
<dd><p>show audio waves
</p></dd>
<dt>&lsquo;<samp>2, rdft</samp>&rsquo;</dt>
<dd><p>show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
</p></dd>
</dl>
<p>Default value is &quot;video&quot;, if video is not present or cannot be played
&quot;rdft&quot; is automatically selected.
</p>
<p>You can interactively cycle through the available show modes by
pressing the key <tt class="key">w</tt>.
</p>
</dd>
<dt><samp>-vf <var>filtergraph</var></samp></dt>
<dd><p>Create the filtergraph specified by <var>filtergraph</var> and use it to
filter the video stream.
</p>
<p><var>filtergraph</var> is a description of the filtergraph to apply to
the stream, and must have a single video input and a single video
output. In the filtergraph, the input is associated to the label
<code>in</code>, and the output to the label <code>out</code>. See the
ffmpeg-filters manual for more information about the filtergraph
syntax.
</p>
<p>You can specify this parameter multiple times and cycle through the specified
filtergraphs along with the show modes by pressing the key <tt class="key">w</tt>.
</p>
</dd>
<dt><samp>-af <var>filtergraph</var></samp></dt>
<dd><p><var>filtergraph</var> is a description of the filtergraph to apply to
the input audio.
Use the option &quot;-filters&quot; to show all the available filters (including
sources and sinks).
</p>
</dd>
<dt><samp>-i <var>input_file</var></samp></dt>
<dd><p>Read <var>input_file</var>.
</p></dd>
</dl>
<a name="Advanced-options"></a>
<h3 class="section">3.5 Advanced options<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-options" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-options" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><samp>-pix_fmt <var>format</var></samp></dt>
<dd><p>Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
</p>
</dd>
<dt><samp>-stats</samp></dt>
<dd><p>Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is on by default, to
explicitly disable it you need to specify <code>-nostats</code>.
</p>
</dd>
<dt><samp>-fast</samp></dt>
<dd><p>Non-spec-compliant optimizations.
</p></dd>
<dt><samp>-genpts</samp></dt>
<dd><p>Generate pts.
</p></dd>
<dt><samp>-sync <var>type</var></samp></dt>
<dd><p>Set the master clock to audio (<code>type=audio</code>), video
(<code>type=video</code>) or external (<code>type=ext</code>). Default is audio. The
master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
</p></dd>
<dt><samp>-ast <var>audio_stream_specifier</var></samp></dt>
<dd><p>Select the desired audio stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; audio stream is selected in the program of the
already selected video stream.
</p></dd>
<dt><samp>-vst <var>video_stream_specifier</var></samp></dt>
<dd><p>Select the desired video stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; video stream is selected.
</p></dd>
<dt><samp>-sst <var>subtitle_stream_specifier</var></samp></dt>
<dd><p>Select the desired subtitle stream using the given stream specifier. The stream
specifiers are described in the <a href="#Stream-specifiers">Stream specifiers</a> chapter. If this option
is not specified, the &quot;best&quot; subtitle stream is selected in the program of the
already selected video or audio stream.
</p></dd>
<dt><samp>-autoexit</samp></dt>
<dd><p>Exit when video is done playing.
</p></dd>
<dt><samp>-exitonkeydown</samp></dt>
<dd><p>Exit if any key is pressed.
</p></dd>
<dt><samp>-exitonmousedown</samp></dt>
<dd><p>Exit if any mouse button is pressed.
</p>
</dd>
<dt><samp>-codec:<var>media_specifier</var> <var>codec_name</var></samp></dt>
<dd><p>Force a specific decoder implementation for the stream identified by
<var>media_specifier</var>, which can assume the values <code>a</code> (audio),
<code>v</code> (video), and <code>s</code> subtitle.
</p>
</dd>
<dt><samp>-acodec <var>codec_name</var></samp></dt>
<dd><p>Force a specific audio decoder.
</p>
</dd>
<dt><samp>-vcodec <var>codec_name</var></samp></dt>
<dd><p>Force a specific video decoder.
</p>
</dd>
<dt><samp>-scodec <var>codec_name</var></samp></dt>
<dd><p>Force a specific subtitle decoder.
</p>
</dd>
<dt><samp>-autorotate</samp></dt>
<dd><p>Automatically rotate the video according to presentation metadata. Enabled by
default, use <samp>-noautorotate</samp> to disable it.
</p>
</dd>
<dt><samp>-framedrop</samp></dt>
<dd><p>Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use <samp>-noframedrop</samp> to disable it.
</p>
</dd>
<dt><samp>-infbuf</samp></dt>
<dd><p>Do not limit the input buffer size, read as much data as possible from the
input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use <samp>-noinfbuf</samp> to disable it.
</p>
</dd>
</dl>
<a name="While-playing"></a>
<h3 class="section">3.6 While playing<span class="pull-right"><a class="anchor hidden-xs" href="#While-playing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-While-playing" aria-hidden="true">TOC</a></span></h3>
<dl compact="compact">
<dt><tt class="key">q, ESC</tt></dt>
<dd><p>Quit.
</p>
</dd>
<dt><tt class="key">f</tt></dt>
<dd><p>Toggle full screen.
</p>
</dd>
<dt><tt class="key">p, SPC</tt></dt>
<dd><p>Pause.
</p>
</dd>
<dt><tt class="key">a</tt></dt>
<dd><p>Cycle audio channel in the current program.
</p>
</dd>
<dt><tt class="key">v</tt></dt>
<dd><p>Cycle video channel.
</p>
</dd>
<dt><tt class="key">t</tt></dt>
<dd><p>Cycle subtitle channel in the current program.
</p>
</dd>
<dt><tt class="key">c</tt></dt>
<dd><p>Cycle program.
</p>
</dd>
<dt><tt class="key">w</tt></dt>
<dd><p>Cycle video filters or show modes.
</p>
</dd>
<dt><tt class="key">s</tt></dt>
<dd><p>Step to the next frame.
</p>
<p>Pause if the stream is not already paused, step to the next video
frame, and pause.
</p>
</dd>
<dt><tt class="key">left/right</tt></dt>
<dd><p>Seek backward/forward 10 seconds.
</p>
</dd>
<dt><tt class="key">down/up</tt></dt>
<dd><p>Seek backward/forward 1 minute.
</p>
</dd>
<dt><tt class="key">page down/page up</tt></dt>
<dd><p>Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
</p>
</dd>
<dt><tt class="key">mouse click</tt></dt>
<dd><p>Seek to percentage in file corresponding to fraction of width.
</p>
</dd>
</dl>
<a name="See-Also"></a>
<h2 class="chapter">4 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffplay-all.html">ffmpeg-all</a>,
<a href="ffmpeg.html">ffmpeg</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-utils.html">ffmpeg-utils</a>,
<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>,
<a href="ffmpeg-bitstream-filters.html">ffmpeg-bitstream-filters</a>,
<a href="ffmpeg-formats.html">ffmpeg-formats</a>,
<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
<a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
<a href="ffmpeg-filters.html">ffmpeg-filters</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">5 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
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General Documentation
</title>
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<h1>
General Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-External-libraries" href="#External-libraries">1 External libraries</a>
<ul class="no-bullet">
<li><a name="toc-OpenJPEG" href="#OpenJPEG">1.1 OpenJPEG</a></li>
<li><a name="toc-OpenCORE_002c-VisualOn_002c-and-Fraunhofer-libraries" href="#OpenCORE_002c-VisualOn_002c-and-Fraunhofer-libraries">1.2 OpenCORE, VisualOn, and Fraunhofer libraries</a>
<ul class="no-bullet">
<li><a name="toc-OpenCORE-AMR" href="#OpenCORE-AMR">1.2.1 OpenCORE AMR</a></li>
<li><a name="toc-VisualOn-AAC-encoder-library" href="#VisualOn-AAC-encoder-library">1.2.2 VisualOn AAC encoder library</a></li>
<li><a name="toc-VisualOn-AMR_002dWB-encoder-library" href="#VisualOn-AMR_002dWB-encoder-library">1.2.3 VisualOn AMR-WB encoder library</a></li>
<li><a name="toc-Fraunhofer-AAC-library" href="#Fraunhofer-AAC-library">1.2.4 Fraunhofer AAC library</a></li>
</ul></li>
<li><a name="toc-LAME" href="#LAME">1.3 LAME</a></li>
<li><a name="toc-TwoLAME" href="#TwoLAME">1.4 TwoLAME</a></li>
<li><a name="toc-libvpx" href="#libvpx">1.5 libvpx</a></li>
<li><a name="toc-libwavpack" href="#libwavpack">1.6 libwavpack</a></li>
<li><a name="toc-OpenH264" href="#OpenH264">1.7 OpenH264</a></li>
<li><a name="toc-x264" href="#x264">1.8 x264</a></li>
<li><a name="toc-x265" href="#x265">1.9 x265</a></li>
<li><a name="toc-libilbc" href="#libilbc">1.10 libilbc</a></li>
<li><a name="toc-libzvbi" href="#libzvbi">1.11 libzvbi</a></li>
<li><a name="toc-AviSynth" href="#AviSynth">1.12 AviSynth</a></li>
</ul></li>
<li><a name="toc-Supported-File-Formats_002c-Codecs-or-Features" href="#Supported-File-Formats_002c-Codecs-or-Features">2 Supported File Formats, Codecs or Features</a>
<ul class="no-bullet">
<li><a name="toc-File-Formats" href="#File-Formats">2.1 File Formats</a></li>
<li><a name="toc-Image-Formats" href="#Image-Formats">2.2 Image Formats</a></li>
<li><a name="toc-Video-Codecs" href="#Video-Codecs">2.3 Video Codecs</a></li>
<li><a name="toc-Audio-Codecs" href="#Audio-Codecs">2.4 Audio Codecs</a></li>
<li><a name="toc-Subtitle-Formats" href="#Subtitle-Formats">2.5 Subtitle Formats</a></li>
<li><a name="toc-Network-Protocols" href="#Network-Protocols">2.6 Network Protocols</a></li>
<li><a name="toc-Input_002fOutput-Devices" href="#Input_002fOutput-Devices">2.7 Input/Output Devices</a></li>
<li><a name="toc-Timecode" href="#Timecode">2.8 Timecode</a></li>
</ul></li>
</ul>
</div>
<a name="External-libraries"></a>
<h2 class="chapter">1 External libraries<span class="pull-right"><a class="anchor hidden-xs" href="#External-libraries" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-External-libraries" aria-hidden="true">TOC</a></span></h2>
<p>FFmpeg can be hooked up with a number of external libraries to add support
for more formats. None of them are used by default, their use has to be
explicitly requested by passing the appropriate flags to
<code>./configure</code>.
</p>
<a name="OpenJPEG"></a>
<h3 class="section">1.1 OpenJPEG<span class="pull-right"><a class="anchor hidden-xs" href="#OpenJPEG" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OpenJPEG" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to
<a href="http://www.openjpeg.org/">http://www.openjpeg.org/</a> to get the libraries and follow the installation
instructions. To enable using OpenJPEG in FFmpeg, pass <code>--enable-libopenjpeg</code> to
<samp>./configure</samp>.
</p>
<a name="OpenCORE_002c-VisualOn_002c-and-Fraunhofer-libraries"></a>
<h3 class="section">1.2 OpenCORE, VisualOn, and Fraunhofer libraries<span class="pull-right"><a class="anchor hidden-xs" href="#OpenCORE_002c-VisualOn_002c-and-Fraunhofer-libraries" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OpenCORE_002c-VisualOn_002c-and-Fraunhofer-libraries" aria-hidden="true">TOC</a></span></h3>
<p>Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
libraries provide encoders for a number of audio codecs.
</p>
<div class="info">
<p>OpenCORE and VisualOn libraries are under the Apache License 2.0
(see <a href="http://www.apache.org/licenses/LICENSE-2.0">http://www.apache.org/licenses/LICENSE-2.0</a> for details), which is
incompatible to the LGPL version 2.1 and GPL version 2. You have to
upgrade FFmpeg&rsquo;s license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) by passing <code>--enable-version3</code> to configure in
order to use it.
</p>
<p>The Fraunhofer AAC library is licensed under a license incompatible to the GPL
and is not known to be compatible to the LGPL. Therefore, you have to pass
<code>--enable-nonfree</code> to configure to use it.
</p></div>
<a name="OpenCORE-AMR"></a>
<h4 class="subsection">1.2.1 OpenCORE AMR<span class="pull-right"><a class="anchor hidden-xs" href="#OpenCORE-AMR" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OpenCORE-AMR" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg can make use of the OpenCORE libraries for AMR-NB
decoding/encoding and AMR-WB decoding.
</p>
<p>Go to <a href="http://sourceforge.net/projects/opencore-amr/">http://sourceforge.net/projects/opencore-amr/</a> and follow the
instructions for installing the libraries.
Then pass <code>--enable-libopencore-amrnb</code> and/or
<code>--enable-libopencore-amrwb</code> to configure to enable them.
</p>
<a name="VisualOn-AAC-encoder-library"></a>
<h4 class="subsection">1.2.2 VisualOn AAC encoder library<span class="pull-right"><a class="anchor hidden-xs" href="#VisualOn-AAC-encoder-library" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-VisualOn-AAC-encoder-library" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg can make use of the VisualOn AACenc library for AAC encoding.
</p>
<p>Go to <a href="http://sourceforge.net/projects/opencore-amr/">http://sourceforge.net/projects/opencore-amr/</a> and follow the
instructions for installing the library.
Then pass <code>--enable-libvo-aacenc</code> to configure to enable it.
</p>
<a name="VisualOn-AMR_002dWB-encoder-library"></a>
<h4 class="subsection">1.2.3 VisualOn AMR-WB encoder library<span class="pull-right"><a class="anchor hidden-xs" href="#VisualOn-AMR_002dWB-encoder-library" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-VisualOn-AMR_002dWB-encoder-library" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.
</p>
<p>Go to <a href="http://sourceforge.net/projects/opencore-amr/">http://sourceforge.net/projects/opencore-amr/</a> and follow the
instructions for installing the library.
Then pass <code>--enable-libvo-amrwbenc</code> to configure to enable it.
</p>
<a name="Fraunhofer-AAC-library"></a>
<h4 class="subsection">1.2.4 Fraunhofer AAC library<span class="pull-right"><a class="anchor hidden-xs" href="#Fraunhofer-AAC-library" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Fraunhofer-AAC-library" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
</p>
<p>Go to <a href="http://sourceforge.net/projects/opencore-amr/">http://sourceforge.net/projects/opencore-amr/</a> and follow the
instructions for installing the library.
Then pass <code>--enable-libfdk-aac</code> to configure to enable it.
</p>
<a name="LAME"></a>
<h3 class="section">1.3 LAME<span class="pull-right"><a class="anchor hidden-xs" href="#LAME" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-LAME" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the LAME library for MP3 encoding.
</p>
<p>Go to <a href="http://lame.sourceforge.net/">http://lame.sourceforge.net/</a> and follow the
instructions for installing the library.
Then pass <code>--enable-libmp3lame</code> to configure to enable it.
</p>
<a name="TwoLAME"></a>
<h3 class="section">1.4 TwoLAME<span class="pull-right"><a class="anchor hidden-xs" href="#TwoLAME" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-TwoLAME" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the TwoLAME library for MP2 encoding.
</p>
<p>Go to <a href="http://www.twolame.org/">http://www.twolame.org/</a> and follow the
instructions for installing the library.
Then pass <code>--enable-libtwolame</code> to configure to enable it.
</p>
<a name="libvpx"></a>
<h3 class="section">1.5 libvpx<span class="pull-right"><a class="anchor hidden-xs" href="#libvpx" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libvpx" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the libvpx library for VP8/VP9 encoding.
</p>
<p>Go to <a href="http://www.webmproject.org/">http://www.webmproject.org/</a> and follow the instructions for
installing the library. Then pass <code>--enable-libvpx</code> to configure to
enable it.
</p>
<a name="libwavpack"></a>
<h3 class="section">1.6 libwavpack<span class="pull-right"><a class="anchor hidden-xs" href="#libwavpack" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libwavpack" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the libwavpack library for WavPack encoding.
</p>
<p>Go to <a href="http://www.wavpack.com/">http://www.wavpack.com/</a> and follow the instructions for
installing the library. Then pass <code>--enable-libwavpack</code> to configure to
enable it.
</p>
<a name="OpenH264"></a>
<h3 class="section">1.7 OpenH264<span class="pull-right"><a class="anchor hidden-xs" href="#OpenH264" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OpenH264" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the OpenH264 library for H.264 encoding.
</p>
<p>Go to <a href="http://www.openh264.org/">http://www.openh264.org/</a> and follow the instructions for
installing the library. Then pass <code>--enable-libopenh264</code> to configure to
enable it.
</p>
<a name="x264"></a>
<h3 class="section">1.8 x264<span class="pull-right"><a class="anchor hidden-xs" href="#x264" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-x264" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the x264 library for H.264 encoding.
</p>
<p>Go to <a href="http://www.videolan.org/developers/x264.html">http://www.videolan.org/developers/x264.html</a> and follow the
instructions for installing the library. Then pass <code>--enable-libx264</code> to
configure to enable it.
</p>
<div class="info">
<p>x264 is under the GNU Public License Version 2 or later
(see <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.html">http://www.gnu.org/licenses/old-licenses/gpl-2.0.html</a> for
details), you must upgrade FFmpeg&rsquo;s license to GPL in order to use it.
</p></div>
<a name="x265"></a>
<h3 class="section">1.9 x265<span class="pull-right"><a class="anchor hidden-xs" href="#x265" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-x265" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can make use of the x265 library for HEVC encoding.
</p>
<p>Go to <a href="http://x265.org/developers.html">http://x265.org/developers.html</a> and follow the instructions
for installing the library. Then pass <code>--enable-libx265</code> to configure
to enable it.
</p>
<div class="info">
<p>x265 is under the GNU Public License Version 2 or later
(see <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.html">http://www.gnu.org/licenses/old-licenses/gpl-2.0.html</a> for
details), you must upgrade FFmpeg&rsquo;s license to GPL in order to use it.
</p></div>
<a name="libilbc"></a>
<h3 class="section">1.10 libilbc<span class="pull-right"><a class="anchor hidden-xs" href="#libilbc" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libilbc" aria-hidden="true">TOC</a></span></h3>
<p>iLBC is a narrowband speech codec that has been made freely available
by Google as part of the WebRTC project. libilbc is a packaging friendly
copy of the iLBC codec. FFmpeg can make use of the libilbc library for
iLBC encoding and decoding.
</p>
<p>Go to <a href="https://github.com/TimothyGu/libilbc">https://github.com/TimothyGu/libilbc</a> and follow the instructions for
installing the library. Then pass <code>--enable-libilbc</code> to configure to
enable it.
</p>
<a name="libzvbi"></a>
<h3 class="section">1.11 libzvbi<span class="pull-right"><a class="anchor hidden-xs" href="#libzvbi" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-libzvbi" aria-hidden="true">TOC</a></span></h3>
<p>libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB
teletext pages and DVB teletext subtitles.
</p>
<p>Go to <a href="http://sourceforge.net/projects/zapping/">http://sourceforge.net/projects/zapping/</a> and follow the instructions for
installing the library. Then pass <code>--enable-libzvbi</code> to configure to
enable it.
</p>
<div class="info">
<p>libzvbi is licensed under the GNU General Public License Version 2 or later
(see <a href="http://www.gnu.org/licenses/old-licenses/gpl-2.0.html">http://www.gnu.org/licenses/old-licenses/gpl-2.0.html</a> for details),
you must upgrade FFmpeg&rsquo;s license to GPL in order to use it.
</p></div>
<a name="AviSynth"></a>
<h3 class="section">1.12 AviSynth<span class="pull-right"><a class="anchor hidden-xs" href="#AviSynth" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-AviSynth" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can read AviSynth scripts as input. To enable support, pass
<code>--enable-avisynth</code> to configure. The correct headers are
included in compat/avisynth/, which allows the user to enable support
without needing to search for these headers themselves.
</p>
<p>For Windows, supported AviSynth variants are
<a href="http://avisynth.nl">AviSynth 2.5 or 2.6</a> for 32-bit builds and
<a href="http://avs-plus.net">AviSynth+ 0.1</a> for 32-bit and 64-bit builds.
</p>
<p>For Linux and OS X, the supported AviSynth variant is
<a href="https://github.com/avxsynth/avxsynth">AvxSynth</a>.
</p>
<div class="info">
<p>AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg
with <code>--enable-avisynth</code>, and the binaries will work regardless of the
end user having AviSynth or AvxSynth installed - they&rsquo;ll only need to be
installed to use AviSynth scripts (obviously).
</p></div>
<a name="Supported-File-Formats_002c-Codecs-or-Features"></a>
<h2 class="chapter">2 Supported File Formats, Codecs or Features<span class="pull-right"><a class="anchor hidden-xs" href="#Supported-File-Formats_002c-Codecs-or-Features" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Supported-File-Formats_002c-Codecs-or-Features" aria-hidden="true">TOC</a></span></h2>
<p>You can use the <code>-formats</code> and <code>-codecs</code> options to have an exhaustive list.
</p>
<a name="File-Formats"></a>
<h3 class="section">2.1 File Formats<span class="pull-right"><a class="anchor hidden-xs" href="#File-Formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-File-Formats" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg supports the following file formats through the <code>libavformat</code>
library:
</p>
<table>
<tr><td width="40%">Name</td><td width="10%">Encoding</td><td width="10%">Decoding</td><td width="40%">Comments</td></tr>
<tr><td width="40%">4xm</td><td width="10%"></td><td width="10%">X</td><td width="40%">4X Technologies format, used in some games.</td></tr>
<tr><td width="40%">8088flex TMV</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ACT Voice</td><td width="10%"></td><td width="10%">X</td><td width="40%">contains G.729 audio</td></tr>
<tr><td width="40%">Adobe Filmstrip</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Audio IFF (AIFF)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">American Laser Games MM</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in games like Mad Dog McCree.</td></tr>
<tr><td width="40%">3GPP AMR</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Amazing Studio Packed Animation File</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in game Heart Of Darkness.</td></tr>
<tr><td width="40%">Apple HTTP Live Streaming</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Artworx Data Format</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADP</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used on the Nintendo Gamecube.</td></tr>
<tr><td width="40%">AFC</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used on the Nintendo Gamecube.</td></tr>
<tr><td width="40%">ASF</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">AST</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Audio format used on the Nintendo Wii.</td></tr>
<tr><td width="40%">AVI</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">AviSynth</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">AVR</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used on Mac.</td></tr>
<tr><td width="40%">AVS</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used by the Creature Shock game.</td></tr>
<tr><td width="40%">Beam Software SIFF</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio and video format used in some games by Beam Software.</td></tr>
<tr><td width="40%">Bethesda Softworks VID</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in some games from Bethesda Softworks.</td></tr>
<tr><td width="40%">Binary text</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Bink</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used by many games.</td></tr>
<tr><td width="40%">Bitmap Brothers JV</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Z and Z95 games.</td></tr>
<tr><td width="40%">Brute Force &amp; Ignorance</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in the game Flash Traffic: City of Angels.</td></tr>
<tr><td width="40%">BRSTM</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used on the Nintendo Wii.</td></tr>
<tr><td width="40%">BWF</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">CRI ADX</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Audio-only format used in console video games.</td></tr>
<tr><td width="40%">Discworld II BMV</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Interplay C93</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in the game Cyberia from Interplay.</td></tr>
<tr><td width="40%">Delphine Software International CIN</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used by Delphine Software games.</td></tr>
<tr><td width="40%">CD+G</td><td width="10%"></td><td width="10%">X</td><td width="40%">Video format used by CD+G karaoke disks</td></tr>
<tr><td width="40%">Phantom Cine</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Commodore CDXL</td><td width="10%"></td><td width="10%">X</td><td width="40%">Amiga CD video format</td></tr>
<tr><td width="40%">Core Audio Format</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Apple Core Audio Format</td></tr>
<tr><td width="40%">CRC testing format</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">Creative Voice</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Created for the Sound Blaster Pro.</td></tr>
<tr><td width="40%">CRYO APC</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used in some games by CRYO Interactive Entertainment.</td></tr>
<tr><td width="40%">D-Cinema audio</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Deluxe Paint Animation</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DFA</td><td width="10%"></td><td width="10%">X</td><td width="40%">This format is used in Chronomaster game</td></tr>
<tr><td width="40%">DSD Stream File (DSF)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DV video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">DXA</td><td width="10%"></td><td width="10%">X</td><td width="40%">This format is used in the non-Windows version of the Feeble Files
game and different game cutscenes repacked for use with ScummVM.</td></tr>
<tr><td width="40%">Electronic Arts cdata</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Electronic Arts Multimedia</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in various EA games; files have extensions like WVE and UV2.</td></tr>
<tr><td width="40%">Ensoniq Paris Audio File</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">FFM (FFserver live feed)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Flash (SWF)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Flash 9 (AVM2)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Only embedded audio is decoded.</td></tr>
<tr><td width="40%">FLI/FLC/FLX animation</td><td width="10%"></td><td width="10%">X</td><td width="40%">.fli/.flc files</td></tr>
<tr><td width="40%">Flash Video (FLV)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Macromedia Flash video files</td></tr>
<tr><td width="40%">framecrc testing format</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">FunCom ISS</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio format used in various games from FunCom like The Longest Journey.</td></tr>
<tr><td width="40%">G.723.1</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">G.729 BIT</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">G.729 raw</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">GIF Animation</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">GXF</td><td width="10%">X</td><td width="10%">X</td><td width="40%">General eXchange Format SMPTE 360M, used by Thomson Grass Valley
playout servers.</td></tr>
<tr><td width="40%">HNM</td><td width="10%"></td><td width="10%">X</td><td width="40%">Only version 4 supported, used in some games from Cryo Interactive</td></tr>
<tr><td width="40%">iCEDraw File</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ICO</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Microsoft Windows ICO</td></tr>
<tr><td width="40%">id Quake II CIN video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">id RoQ</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in Quake III, Jedi Knight 2 and other computer games.</td></tr>
<tr><td width="40%">IEC61937 encapsulation</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">IFF</td><td width="10%"></td><td width="10%">X</td><td width="40%">Interchange File Format</td></tr>
<tr><td width="40%">iLBC</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Interplay MVE</td><td width="10%"></td><td width="10%">X</td><td width="40%">Format used in various Interplay computer games.</td></tr>
<tr><td width="40%">IV8</td><td width="10%"></td><td width="10%">X</td><td width="40%">A format generated by IndigoVision 8000 video server.</td></tr>
<tr><td width="40%">IVF (On2)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">A format used by libvpx</td></tr>
<tr><td width="40%">IRCAM</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">LATM</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">LMLM4</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used by Linux Media Labs MPEG-4 PCI boards</td></tr>
<tr><td width="40%">LOAS</td><td width="10%"></td><td width="10%">X</td><td width="40%">contains LATM multiplexed AAC audio</td></tr>
<tr><td width="40%">LRC</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">LVF</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">LXF</td><td width="10%"></td><td width="10%">X</td><td width="40%">VR native stream format, used by Leitch/Harris&rsquo; video servers.</td></tr>
<tr><td width="40%">Magic Lantern Video (MLV)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Matroska</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Matroska audio</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">FFmpeg metadata</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Metadata in text format.</td></tr>
<tr><td width="40%">MAXIS XA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sim City 3000; file extension .xa.</td></tr>
<tr><td width="40%">MD Studio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Metal Gear Solid: The Twin Snakes</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Megalux Frame</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used by Megalux Ultimate Paint</td></tr>
<tr><td width="40%">Mobotix .mxg</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Monkey&rsquo;s Audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Motion Pixels MVI</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MOV/QuickTime/MP4</td><td width="10%">X</td><td width="10%">X</td><td width="40%">3GP, 3GP2, PSP, iPod variants supported</td></tr>
<tr><td width="40%">MP2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MP3</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-1 System</td><td width="10%">X</td><td width="10%">X</td><td width="40%">muxed audio and video, VCD format supported</td></tr>
<tr><td width="40%">MPEG-PS (program stream)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">also known as <code>VOB</code> file, SVCD and DVD format supported</td></tr>
<tr><td width="40%">MPEG-TS (transport stream)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">also known as DVB Transport Stream</td></tr>
<tr><td width="40%">MPEG-4</td><td width="10%">X</td><td width="10%">X</td><td width="40%">MPEG-4 is a variant of QuickTime.</td></tr>
<tr><td width="40%">Mirillis FIC video</td><td width="10%"></td><td width="10%">X</td><td width="40%">No cursor rendering.</td></tr>
<tr><td width="40%">MIME multipart JPEG</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">MSN TCP webcam</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used by MSN Messenger webcam streams.</td></tr>
<tr><td width="40%">MTV</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Musepack</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Musepack SV8</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Material eXchange Format (MXF)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">SMPTE 377M, used by D-Cinema, broadcast industry.</td></tr>
<tr><td width="40%">Material eXchange Format (MXF), D-10 Mapping</td><td width="10%">X</td><td width="10%">X</td><td width="40%">SMPTE 386M, D-10/IMX Mapping.</td></tr>
<tr><td width="40%">NC camera feed</td><td width="10%"></td><td width="10%">X</td><td width="40%">NC (AVIP NC4600) camera streams</td></tr>
<tr><td width="40%">NIST SPeech HEader REsources</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">NTT TwinVQ (VQF)</td><td width="10%"></td><td width="10%">X</td><td width="40%">Nippon Telegraph and Telephone Corporation TwinVQ.</td></tr>
<tr><td width="40%">Nullsoft Streaming Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">NuppelVideo</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">NUT</td><td width="10%">X</td><td width="10%">X</td><td width="40%">NUT Open Container Format</td></tr>
<tr><td width="40%">Ogg</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Playstation Portable PMP</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Portable Voice Format</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">TechnoTrend PVA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used by TechnoTrend DVB PCI boards.</td></tr>
<tr><td width="40%">QCP</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw ADTS (AAC)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw AC-3</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw Chinese AVS video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw CRI ADX</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw Dirac</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw DNxHD</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw DTS</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw DTS-HD</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw E-AC-3</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw FLAC</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw GSM</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw H.261</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw H.263</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw H.264</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw HEVC</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw Ingenient MJPEG</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw MJPEG</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw MLP</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw MPEG</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw MPEG-1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw MPEG-2</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw MPEG-4</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw NULL</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">raw video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw id RoQ</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">raw Shorten</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw TAK</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">raw TrueHD</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw VC-1</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM A-law</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM mu-law</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 8 bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 16 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 16 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 24 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 24 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 32 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM signed 32 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 8 bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 16 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 16 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 24 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 24 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 32 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM unsigned 32 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM floating-point 32 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM floating-point 32 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM floating-point 64 bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">raw PCM floating-point 64 bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RDT</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">REDCODE R3D</td><td width="10%"></td><td width="10%">X</td><td width="40%">File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.</td></tr>
<tr><td width="40%">RealMedia</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Redirector</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">RedSpark</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Renderware TeXture Dictionary</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">RL2</td><td width="10%"></td><td width="10%">X</td><td width="40%">Audio and video format used in some games by Entertainment Software Partners.</td></tr>
<tr><td width="40%">RPL/ARMovie</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Lego Mindstorms RSO</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RSD</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">RTMP</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Output is performed by publishing stream to RTMP server</td></tr>
<tr><td width="40%">RTP</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RTSP</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SAP</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SBG</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">SDP</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Sega FILM/CPK</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in many Sega Saturn console games.</td></tr>
<tr><td width="40%">Silicon Graphics Movie</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Sierra SOL</td><td width="10%"></td><td width="10%">X</td><td width="40%">.sol files used in Sierra Online games.</td></tr>
<tr><td width="40%">Sierra VMD</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sierra CD-ROM games.</td></tr>
<tr><td width="40%">Smacker</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used by many games.</td></tr>
<tr><td width="40%">SMJPEG</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in certain Loki game ports.</td></tr>
<tr><td width="40%">Smush</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in some LucasArts games.</td></tr>
<tr><td width="40%">Sony OpenMG (OMA)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Audio format used in Sony Sonic Stage and Sony Vegas.</td></tr>
<tr><td width="40%">Sony PlayStation STR</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Sony Wave64 (W64)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SoX native format</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SUN AU format</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SUP raw PGS subtitles</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Text files</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">THP</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used on the Nintendo GameCube.</td></tr>
<tr><td width="40%">Tiertex Limited SEQ</td><td width="10%"></td><td width="10%">X</td><td width="40%">Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.</td></tr>
<tr><td width="40%">True Audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">VC-1 test bitstream</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Vivo</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">WAV</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">WavPack</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">WebM</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Televison (WTV)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Wing Commander III movie</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in Origin&rsquo;s Wing Commander III computer game.</td></tr>
<tr><td width="40%">Westwood Studios audio</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in Westwood Studios games.</td></tr>
<tr><td width="40%">Westwood Studios VQA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Multimedia format used in Westwood Studios games.</td></tr>
<tr><td width="40%">XMV</td><td width="10%"></td><td width="10%">X</td><td width="40%">Microsoft video container used in Xbox games.</td></tr>
<tr><td width="40%">xWMA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Microsoft audio container used by XAudio 2.</td></tr>
<tr><td width="40%">eXtended BINary text (XBIN)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">YUV4MPEG pipe</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Psygnosis YOP</td><td width="10%"></td><td width="10%">X</td></tr>
</table>
<p><code>X</code> means that encoding (resp. decoding) is supported.
</p>
<a name="Image-Formats"></a>
<h3 class="section">2.2 Image Formats<span class="pull-right"><a class="anchor hidden-xs" href="#Image-Formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Image-Formats" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can read and write images for each frame of a video sequence. The
following image formats are supported:
</p>
<table>
<tr><td width="40%">Name</td><td width="10%">Encoding</td><td width="10%">Decoding</td><td width="40%">Comments</td></tr>
<tr><td width="40%">.Y.U.V</td><td width="10%">X</td><td width="10%">X</td><td width="40%">one raw file per component</td></tr>
<tr><td width="40%">Alias PIX</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Alias/Wavefront PIX image format</td></tr>
<tr><td width="40%">animated GIF</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">BMP</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Microsoft BMP image</td></tr>
<tr><td width="40%">BRender PIX</td><td width="10%"></td><td width="10%">X</td><td width="40%">Argonaut BRender 3D engine image format.</td></tr>
<tr><td width="40%">DPX</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Digital Picture Exchange</td></tr>
<tr><td width="40%">EXR</td><td width="10%"></td><td width="10%">X</td><td width="40%">OpenEXR</td></tr>
<tr><td width="40%">JPEG</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Progressive JPEG is not supported.</td></tr>
<tr><td width="40%">JPEG 2000</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">JPEG-LS</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">LJPEG</td><td width="10%">X</td><td width="10%"></td><td width="40%">Lossless JPEG</td></tr>
<tr><td width="40%">PAM</td><td width="10%">X</td><td width="10%">X</td><td width="40%">PAM is a PNM extension with alpha support.</td></tr>
<tr><td width="40%">PBM</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Portable BitMap image</td></tr>
<tr><td width="40%">PCX</td><td width="10%">X</td><td width="10%">X</td><td width="40%">PC Paintbrush</td></tr>
<tr><td width="40%">PGM</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Portable GrayMap image</td></tr>
<tr><td width="40%">PGMYUV</td><td width="10%">X</td><td width="10%">X</td><td width="40%">PGM with U and V components in YUV 4:2:0</td></tr>
<tr><td width="40%">PIC</td><td width="10%"></td><td width="10%">X</td><td width="40%">Pictor/PC Paint</td></tr>
<tr><td width="40%">PNG</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PPM</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Portable PixelMap image</td></tr>
<tr><td width="40%">PTX</td><td width="10%"></td><td width="10%">X</td><td width="40%">V.Flash PTX format</td></tr>
<tr><td width="40%">SGI</td><td width="10%">X</td><td width="10%">X</td><td width="40%">SGI RGB image format</td></tr>
<tr><td width="40%">Sun Rasterfile</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Sun RAS image format</td></tr>
<tr><td width="40%">TIFF</td><td width="10%">X</td><td width="10%">X</td><td width="40%">YUV, JPEG and some extension is not supported yet.</td></tr>
<tr><td width="40%">Truevision Targa</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Targa (.TGA) image format</td></tr>
<tr><td width="40%">WebP</td><td width="10%">E</td><td width="10%">X</td><td width="40%">WebP image format, encoding supported through external library libwebp</td></tr>
<tr><td width="40%">XBM</td><td width="10%">X</td><td width="10%">X</td><td width="40%">X BitMap image format</td></tr>
<tr><td width="40%">XFace</td><td width="10%">X</td><td width="10%">X</td><td width="40%">X-Face image format</td></tr>
<tr><td width="40%">XWD</td><td width="10%">X</td><td width="10%">X</td><td width="40%">X Window Dump image format</td></tr>
</table>
<p><code>X</code> means that encoding (resp. decoding) is supported.
</p>
<p><code>E</code> means that support is provided through an external library.
</p>
<a name="Video-Codecs"></a>
<h3 class="section">2.3 Video Codecs<span class="pull-right"><a class="anchor hidden-xs" href="#Video-Codecs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Video-Codecs" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Name</td><td width="10%">Encoding</td><td width="10%">Decoding</td><td width="40%">Comments</td></tr>
<tr><td width="40%">4X Movie</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in certain computer games.</td></tr>
<tr><td width="40%">8088flex TMV</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">A64 multicolor</td><td width="10%">X</td><td width="10%"></td><td width="40%">Creates video suitable to be played on a commodore 64 (multicolor mode).</td></tr>
<tr><td width="40%">Amazing Studio PAF Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">American Laser Games MM</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in games like Mad Dog McCree.</td></tr>
<tr><td width="40%">AMV Video</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in Chinese MP3 players.</td></tr>
<tr><td width="40%">ANSI/ASCII art</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Apple Intermediate Codec</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Apple MJPEG-B</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Apple ProRes</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Apple QuickDraw</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: qdrw</td></tr>
<tr><td width="40%">Asus v1</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: ASV1</td></tr>
<tr><td width="40%">Asus v2</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: ASV2</td></tr>
<tr><td width="40%">ATI VCR1</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VCR1</td></tr>
<tr><td width="40%">ATI VCR2</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VCR2</td></tr>
<tr><td width="40%">Auravision Aura</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Auravision Aura 2</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Autodesk Animator Flic video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Autodesk RLE</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: AASC</td></tr>
<tr><td width="40%">Avid 1:1 10-bit RGB Packer</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: AVrp</td></tr>
<tr><td width="40%">AVS (Audio Video Standard) video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Video encoding used by the Creature Shock game.</td></tr>
<tr><td width="40%">AYUV</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Microsoft uncompressed packed 4:4:4:4</td></tr>
<tr><td width="40%">Beam Software VB</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Bethesda VID video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in some games from Bethesda Softworks.</td></tr>
<tr><td width="40%">Bink Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Bitmap Brothers JV video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">y41p Brooktree uncompressed 4:1:1 12-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Brute Force &amp; Ignorance</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in the game Flash Traffic: City of Angels.</td></tr>
<tr><td width="40%">C93 video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in Cyberia game.</td></tr>
<tr><td width="40%">CamStudio</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: CSCD</td></tr>
<tr><td width="40%">CD+G</td><td width="10%"></td><td width="10%">X</td><td width="40%">Video codec for CD+G karaoke disks</td></tr>
<tr><td width="40%">CDXL</td><td width="10%"></td><td width="10%">X</td><td width="40%">Amiga CD video codec</td></tr>
<tr><td width="40%">Chinese AVS video</td><td width="10%">E</td><td width="10%">X</td><td width="40%">AVS1-P2, JiZhun profile, encoding through external library libxavs</td></tr>
<tr><td width="40%">Delphine Software International CIN video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in Delphine Software International games.</td></tr>
<tr><td width="40%">Discworld II BMV Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Canopus Lossless Codec</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Cinepak</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Cirrus Logic AccuPak</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: CLJR</td></tr>
<tr><td width="40%">CPiA Video Format</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Creative YUV (CYUV)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DFA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in Chronomaster game.</td></tr>
<tr><td width="40%">Dirac</td><td width="10%">E</td><td width="10%">X</td><td width="40%">supported through external library libschroedinger</td></tr>
<tr><td width="40%">Deluxe Paint Animation</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DNxHD</td><td width="10%">X</td><td width="10%">X</td><td width="40%">aka SMPTE VC3</td></tr>
<tr><td width="40%">Duck TrueMotion 1.0</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: DUCK</td></tr>
<tr><td width="40%">Duck TrueMotion 2.0</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: TM20</td></tr>
<tr><td width="40%">DV (Digital Video)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Dxtory capture format</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Feeble Files/ScummVM DXA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec originally used in Feeble Files game.</td></tr>
<tr><td width="40%">Electronic Arts CMV video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in NHL 95 game.</td></tr>
<tr><td width="40%">Electronic Arts Madcow video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Electronic Arts TGV video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Electronic Arts TGQ video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Electronic Arts TQI video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Escape 124</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Escape 130</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">FFmpeg video codec #1</td><td width="10%">X</td><td width="10%">X</td><td width="40%">lossless codec (fourcc: FFV1)</td></tr>
<tr><td width="40%">Flash Screen Video v1</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: FSV1</td></tr>
<tr><td width="40%">Flash Screen Video v2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Flash Video (FLV)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Sorenson H.263 used in Flash</td></tr>
<tr><td width="40%">Forward Uncompressed</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Fraps</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Go2Webinar</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: G2M4</td></tr>
<tr><td width="40%">H.261</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">H.263 / H.263-1996</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">H.263+ / H.263-1998 / H.263 version 2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libx264 and OpenH264</td></tr>
<tr><td width="40%">HEVC</td><td width="10%">X</td><td width="10%">X</td><td width="40%">encoding supported through the external library libx265</td></tr>
<tr><td width="40%">HNM version 4</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">HuffYUV</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">HuffYUV FFmpeg variant</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">IBM Ultimotion</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: ULTI</td></tr>
<tr><td width="40%">id Cinematic video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Quake II.</td></tr>
<tr><td width="40%">id RoQ video</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in Quake III, Jedi Knight 2, other computer games.</td></tr>
<tr><td width="40%">IFF ILBM</td><td width="10%"></td><td width="10%">X</td><td width="40%">IFF interleaved bitmap</td></tr>
<tr><td width="40%">IFF ByteRun1</td><td width="10%"></td><td width="10%">X</td><td width="40%">IFF run length encoded bitmap</td></tr>
<tr><td width="40%">Intel H.263</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Intel Indeo 2</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Intel Indeo 3</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Intel Indeo 4</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Intel Indeo 5</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Interplay C93</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in the game Cyberia from Interplay.</td></tr>
<tr><td width="40%">Interplay MVE video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Interplay .MVE files.</td></tr>
<tr><td width="40%">J2K</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Karl Morton&rsquo;s video codec</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in Worms games.</td></tr>
<tr><td width="40%">Kega Game Video (KGV1)</td><td width="10%"></td><td width="10%">X</td><td width="40%">Kega emulator screen capture codec.</td></tr>
<tr><td width="40%">Lagarith</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">LCL (LossLess Codec Library) MSZH</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">LCL (LossLess Codec Library) ZLIB</td><td width="10%">E</td><td width="10%">E</td></tr>
<tr><td width="40%">LOCO</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">LucasArts SANM/Smush</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in LucasArts games / SMUSH animations.</td></tr>
<tr><td width="40%">lossless MJPEG</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Microsoft ATC Screen</td><td width="10%"></td><td width="10%">X</td><td width="40%">Also known as Microsoft Screen 3.</td></tr>
<tr><td width="40%">Microsoft Expression Encoder Screen</td><td width="10%"></td><td width="10%">X</td><td width="40%">Also known as Microsoft Titanium Screen 2.</td></tr>
<tr><td width="40%">Microsoft RLE</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Microsoft Screen 1</td><td width="10%"></td><td width="10%">X</td><td width="40%">Also known as Windows Media Video V7 Screen.</td></tr>
<tr><td width="40%">Microsoft Screen 2</td><td width="10%"></td><td width="10%">X</td><td width="40%">Also known as Windows Media Video V9 Screen.</td></tr>
<tr><td width="40%">Microsoft Video 1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Mimic</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in MSN Messenger Webcam streams.</td></tr>
<tr><td width="40%">Miro VideoXL</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VIXL</td></tr>
<tr><td width="40%">MJPEG (Motion JPEG)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Mobotix MxPEG video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Motion Pixels video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-1 video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-2 video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-4 part 2</td><td width="10%">X</td><td width="10%">X</td><td width="40%">libxvidcore can be used alternatively for encoding.</td></tr>
<tr><td width="40%">MPEG-4 part 2 Microsoft variant version 1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-4 part 2 Microsoft variant version 2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG-4 part 2 Microsoft variant version 3</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Nintendo Gamecube THP video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">NuppelVideo/RTjpeg</td><td width="10%"></td><td width="10%">X</td><td width="40%">Video encoding used in NuppelVideo files.</td></tr>
<tr><td width="40%">On2 VP3</td><td width="10%"></td><td width="10%">X</td><td width="40%">still experimental</td></tr>
<tr><td width="40%">On2 VP5</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VP50</td></tr>
<tr><td width="40%">On2 VP6</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VP60,VP61,VP62</td></tr>
<tr><td width="40%">On2 VP7</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: VP70,VP71</td></tr>
<tr><td width="40%">VP8</td><td width="10%">E</td><td width="10%">X</td><td width="40%">fourcc: VP80, encoding supported through external library libvpx</td></tr>
<tr><td width="40%">VP9</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libvpx</td></tr>
<tr><td width="40%">Pinnacle TARGA CineWave YUV16</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: Y216</td></tr>
<tr><td width="40%">Prores</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: apch,apcn,apcs,apco</td></tr>
<tr><td width="40%">Q-team QPEG</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourccs: QPEG, Q1.0, Q1.1</td></tr>
<tr><td width="40%">QuickTime 8BPS video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">QuickTime Animation (RLE) video</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: &rsquo;rle &rsquo;</td></tr>
<tr><td width="40%">QuickTime Graphics (SMC)</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: &rsquo;smc &rsquo;</td></tr>
<tr><td width="40%">QuickTime video (RPZA)</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: rpza</td></tr>
<tr><td width="40%">R10K AJA Kona 10-bit RGB Codec</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">R210 Quicktime Uncompressed RGB 10-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Raw Video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RealVideo 1.0</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RealVideo 2.0</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">RealVideo 3.0</td><td width="10%"></td><td width="10%">X</td><td width="40%">still far from ideal</td></tr>
<tr><td width="40%">RealVideo 4.0</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Renderware TXD (TeXture Dictionary)</td><td width="10%"></td><td width="10%">X</td><td width="40%">Texture dictionaries used by the Renderware Engine.</td></tr>
<tr><td width="40%">RL2 video</td><td width="10%"></td><td width="10%">X</td><td width="40%">used in some games by Entertainment Software Partners</td></tr>
<tr><td width="40%">Sierra VMD video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sierra VMD files.</td></tr>
<tr><td width="40%">Silicon Graphics Motion Video Compressor 1 (MVC1)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Silicon Graphics Motion Video Compressor 2 (MVC2)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Silicon Graphics RLE 8-bit video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Smacker video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Video encoding used in Smacker.</td></tr>
<tr><td width="40%">SMPTE VC-1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Snow</td><td width="10%">X</td><td width="10%">X</td><td width="40%">experimental wavelet codec (fourcc: SNOW)</td></tr>
<tr><td width="40%">Sony PlayStation MDEC (Motion DECoder)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Sorenson Vector Quantizer 1</td><td width="10%">X</td><td width="10%">X</td><td width="40%">fourcc: SVQ1</td></tr>
<tr><td width="40%">Sorenson Vector Quantizer 3</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: SVQ3</td></tr>
<tr><td width="40%">Sunplus JPEG (SP5X)</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: SP5X</td></tr>
<tr><td width="40%">TechSmith Screen Capture Codec</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: TSCC</td></tr>
<tr><td width="40%">TechSmith Screen Capture Codec 2</td><td width="10%"></td><td width="10%">X</td><td width="40%">fourcc: TSC2</td></tr>
<tr><td width="40%">Theora</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libtheora</td></tr>
<tr><td width="40%">Tiertex Limited SEQ video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in DOS CD-ROM FlashBack game.</td></tr>
<tr><td width="40%">Ut Video</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">v210 QuickTime uncompressed 4:2:2 10-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">v308 QuickTime uncompressed 4:4:4</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">v408 QuickTime uncompressed 4:4:4:4</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">v410 QuickTime uncompressed 4:4:4 10-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">VBLE Lossless Codec</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">VMware Screen Codec / VMware Video</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in videos captured by VMware.</td></tr>
<tr><td width="40%">Westwood Studios VQA (Vector Quantized Animation) video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Image</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Video 7</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Video 8</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Video 9</td><td width="10%"></td><td width="10%">X</td><td width="40%">not completely working</td></tr>
<tr><td width="40%">Wing Commander III / Xan</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Wing Commander III .MVE files.</td></tr>
<tr><td width="40%">Wing Commander IV / Xan</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Wing Commander IV.</td></tr>
<tr><td width="40%">Winnov WNV1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">WMV7</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">YAMAHA SMAF</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Psygnosis YOP Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">yuv4</td><td width="10%">X</td><td width="10%">X</td><td width="40%">libquicktime uncompressed packed 4:2:0</td></tr>
<tr><td width="40%">ZeroCodec Lossless Video</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ZLIB</td><td width="10%">X</td><td width="10%">X</td><td width="40%">part of LCL, encoder experimental</td></tr>
<tr><td width="40%">Zip Motion Blocks Video</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Encoder works only in PAL8.</td></tr>
</table>
<p><code>X</code> means that encoding (resp. decoding) is supported.
</p>
<p><code>E</code> means that support is provided through an external library.
</p>
<a name="Audio-Codecs"></a>
<h3 class="section">2.4 Audio Codecs<span class="pull-right"><a class="anchor hidden-xs" href="#Audio-Codecs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Audio-Codecs" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Name</td><td width="10%">Encoding</td><td width="10%">Decoding</td><td width="40%">Comments</td></tr>
<tr><td width="40%">8SVX exponential</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">8SVX fibonacci</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">AAC+</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libaacplus</td></tr>
<tr><td width="40%">AAC</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libfaac and libvo-aacenc</td></tr>
<tr><td width="40%">AC-3</td><td width="10%">IX</td><td width="10%">IX</td></tr>
<tr><td width="40%">ADPCM 4X Movie</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM CDROM XA</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Creative Technology</td><td width="10%"></td><td width="10%">X</td><td width="40%">16 -&gt; 4, 8 -&gt; 4, 8 -&gt; 3, 8 -&gt; 2</td></tr>
<tr><td width="40%">ADPCM Electronic Arts</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in various EA titles.</td></tr>
<tr><td width="40%">ADPCM Electronic Arts Maxis CDROM XS</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sim City 3000.</td></tr>
<tr><td width="40%">ADPCM Electronic Arts R1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Electronic Arts R2</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Electronic Arts R3</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Electronic Arts XAS</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM G.722</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM G.726</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA AMV</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in AMV files</td></tr>
<tr><td width="40%">ADPCM IMA Electronic Arts EACS</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA Electronic Arts SEAD</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA Funcom</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA QuickTime</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA Loki SDL MJPEG</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA WAV</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA Westwood</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM ISS IMA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in FunCom games.</td></tr>
<tr><td width="40%">ADPCM IMA Dialogic</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM IMA Duck DK3</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in some Sega Saturn console games.</td></tr>
<tr><td width="40%">ADPCM IMA Duck DK4</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in some Sega Saturn console games.</td></tr>
<tr><td width="40%">ADPCM IMA Radical</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Microsoft</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM MS IMA</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Nintendo Gamecube AFC</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Nintendo Gamecube DTK</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Nintendo Gamecube THP</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM QT IMA</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM SEGA CRI ADX</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in Sega Dreamcast games.</td></tr>
<tr><td width="40%">ADPCM Shockwave Flash</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Sound Blaster Pro 2-bit</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Sound Blaster Pro 2.6-bit</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM Sound Blaster Pro 4-bit</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ADPCM VIMA</td><td width="10%">Used in LucasArts SMUSH animations.</td></tr>
<tr><td width="40%">ADPCM Westwood Studios IMA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Westwood Studios games like Command and Conquer.</td></tr>
<tr><td width="40%">ADPCM Yamaha</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">AMR-NB</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libopencore-amrnb</td></tr>
<tr><td width="40%">AMR-WB</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libvo-amrwbenc</td></tr>
<tr><td width="40%">Amazing Studio PAF Audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Apple lossless audio</td><td width="10%">X</td><td width="10%">X</td><td width="40%">QuickTime fourcc &rsquo;alac&rsquo;</td></tr>
<tr><td width="40%">ATRAC1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ATRAC3</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">ATRAC3+</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Bink Audio</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Bink and Smacker files in many games.</td></tr>
<tr><td width="40%">CELT</td><td width="10%"></td><td width="10%">E</td><td width="40%">decoding supported through external library libcelt</td></tr>
<tr><td width="40%">Delphine Software International CIN audio</td><td width="10%"></td><td width="10%">X</td><td width="40%">Codec used in Delphine Software International games.</td></tr>
<tr><td width="40%">Discworld II BMV Audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">COOK</td><td width="10%"></td><td width="10%">X</td><td width="40%">All versions except 5.1 are supported.</td></tr>
<tr><td width="40%">DCA (DTS Coherent Acoustics)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">DPCM id RoQ</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Used in Quake III, Jedi Knight 2 and other computer games.</td></tr>
<tr><td width="40%">DPCM Interplay</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in various Interplay computer games.</td></tr>
<tr><td width="40%">DPCM Sierra Online</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sierra Online game audio files.</td></tr>
<tr><td width="40%">DPCM Sol</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DPCM Xan</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Origin&rsquo;s Wing Commander IV AVI files.</td></tr>
<tr><td width="40%">DSD (Direct Stream Digitial), least significant bit first</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DSD (Direct Stream Digitial), most significant bit first</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DSD (Direct Stream Digitial), least significant bit first, planar</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DSD (Direct Stream Digitial), most significant bit first, planar</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DSP Group TrueSpeech</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DV audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Enhanced AC-3</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">EVRC (Enhanced Variable Rate Codec)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">FLAC (Free Lossless Audio Codec)</td><td width="10%">X</td><td width="10%">IX</td></tr>
<tr><td width="40%">G.723.1</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">G.729</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">GSM</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libgsm</td></tr>
<tr><td width="40%">GSM Microsoft variant</td><td width="10%">E</td><td width="10%">X</td><td width="40%">encoding supported through external library libgsm</td></tr>
<tr><td width="40%">IAC (Indeo Audio Coder)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">iLBC (Internet Low Bitrate Codec)</td><td width="10%">E</td><td width="10%">E</td><td width="40%">encoding and decoding supported through external library libilbc</td></tr>
<tr><td width="40%">IMC (Intel Music Coder)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MACE (Macintosh Audio Compression/Expansion) 3:1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MACE (Macintosh Audio Compression/Expansion) 6:1</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MLP (Meridian Lossless Packing)</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in DVD-Audio discs.</td></tr>
<tr><td width="40%">Monkey&rsquo;s Audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MP1 (MPEG audio layer 1)</td><td width="10%"></td><td width="10%">IX</td></tr>
<tr><td width="40%">MP2 (MPEG audio layer 2)</td><td width="10%">IX</td><td width="10%">IX</td><td width="40%">encoding supported also through external library TwoLAME</td></tr>
<tr><td width="40%">MP3 (MPEG audio layer 3)</td><td width="10%">E</td><td width="10%">IX</td><td width="40%">encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported</td></tr>
<tr><td width="40%">MPEG-4 Audio Lossless Coding (ALS)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Musepack SV7</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Musepack SV8</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Nellymoser Asao</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">On2 AVC (Audio for Video Codec)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Opus</td><td width="10%">E</td><td width="10%">E</td><td width="40%">supported through external library libopus</td></tr>
<tr><td width="40%">PCM A-law</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM mu-law</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 8-bit planar</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 16-bit big-endian planar</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 16-bit little-endian planar</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 24-bit little-endian planar</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 32-bit little-endian planar</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM 32-bit floating point big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM 32-bit floating point little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM 64-bit floating point big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM 64-bit floating point little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM D-Cinema audio signed 24-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 8-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 16-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 16-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 24-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 24-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 32-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 32-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM signed 16/20/24-bit big-endian in MPEG-TS</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 8-bit</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 16-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 16-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 24-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 24-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 32-bit big-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM unsigned 32-bit little-endian</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PCM Zork</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">QCELP / PureVoice</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">QDesign Music Codec 2</td><td width="10%"></td><td width="10%">X</td><td width="40%">There are still some distortions.</td></tr>
<tr><td width="40%">RealAudio 1.0 (14.4K)</td><td width="10%">X</td><td width="10%">X</td><td width="40%">Real 14400 bit/s codec</td></tr>
<tr><td width="40%">RealAudio 2.0 (28.8K)</td><td width="10%"></td><td width="10%">X</td><td width="40%">Real 28800 bit/s codec</td></tr>
<tr><td width="40%">RealAudio 3.0 (dnet)</td><td width="10%">IX</td><td width="10%">X</td><td width="40%">Real low bitrate AC-3 codec</td></tr>
<tr><td width="40%">RealAudio Lossless</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">RealAudio SIPR / ACELP.NET</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Shorten</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Sierra VMD audio</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in Sierra VMD files.</td></tr>
<tr><td width="40%">Smacker audio</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">SMPTE 302M AES3 audio</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Sonic</td><td width="10%">X</td><td width="10%">X</td><td width="40%">experimental codec</td></tr>
<tr><td width="40%">Sonic lossless</td><td width="10%">X</td><td width="10%">X</td><td width="40%">experimental codec</td></tr>
<tr><td width="40%">Speex</td><td width="10%">E</td><td width="10%">E</td><td width="40%">supported through external library libspeex</td></tr>
<tr><td width="40%">TAK (Tom&rsquo;s lossless Audio Kompressor)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">True Audio (TTA)</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">TrueHD</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in HD-DVD and Blu-Ray discs.</td></tr>
<tr><td width="40%">TwinVQ (VQF flavor)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">VIMA</td><td width="10%"></td><td width="10%">X</td><td width="40%">Used in LucasArts SMUSH animations.</td></tr>
<tr><td width="40%">Vorbis</td><td width="10%">E</td><td width="10%">X</td><td width="40%">A native but very primitive encoder exists.</td></tr>
<tr><td width="40%">Voxware MetaSound</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">WavPack</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Westwood Audio (SND1)</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Audio 1</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Audio 2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Audio Lossless</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Audio Pro</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Windows Media Audio Voice</td><td width="10%"></td><td width="10%">X</td></tr>
</table>
<p><code>X</code> means that encoding (resp. decoding) is supported.
</p>
<p><code>E</code> means that support is provided through an external library.
</p>
<p><code>I</code> means that an integer-only version is available, too (ensures high
performance on systems without hardware floating point support).
</p>
<a name="Subtitle-Formats"></a>
<h3 class="section">2.5 Subtitle Formats<span class="pull-right"><a class="anchor hidden-xs" href="#Subtitle-Formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Subtitle-Formats" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Name</td><td width="10%">Muxing</td><td width="10%">Demuxing</td><td width="10%">Encoding</td><td width="10%">Decoding</td></tr>
<tr><td width="40%">3GPP Timed Text</td><td width="10%"></td><td width="10%"></td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">AQTitle</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DVB</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">DVB teletext</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">E</td></tr>
<tr><td width="40%">DVD</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">JACOsub</td><td width="10%">X</td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MicroDVD</td><td width="10%">X</td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MPL2</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">MPsub (MPlayer)</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">PGS</td><td width="10%"></td><td width="10%"></td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">PJS (Phoenix)</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">RealText</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">SAMI</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Spruce format (STL)</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">SSA/ASS</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SubRip (SRT)</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SubViewer v1</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">SubViewer</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">TED Talks captions</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">VobSub (IDX+SUB)</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">VPlayer</td><td width="10%"></td><td width="10%">X</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">WebVTT</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">XSUB</td><td width="10%"></td><td width="10%"></td><td width="10%">X</td><td width="10%">X</td></tr>
</table>
<p><code>X</code> means that the feature is supported.
</p>
<p><code>E</code> means that support is provided through an external library.
</p>
<a name="Network-Protocols"></a>
<h3 class="section">2.6 Network Protocols<span class="pull-right"><a class="anchor hidden-xs" href="#Network-Protocols" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Network-Protocols" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Name</td><td width="10%">Support</td></tr>
<tr><td width="40%">file</td><td width="10%">X</td></tr>
<tr><td width="40%">FTP</td><td width="10%">X</td></tr>
<tr><td width="40%">Gopher</td><td width="10%">X</td></tr>
<tr><td width="40%">HLS</td><td width="10%">X</td></tr>
<tr><td width="40%">HTTP</td><td width="10%">X</td></tr>
<tr><td width="40%">HTTPS</td><td width="10%">X</td></tr>
<tr><td width="40%">Icecast</td><td width="10%">X</td></tr>
<tr><td width="40%">MMSH</td><td width="10%">X</td></tr>
<tr><td width="40%">MMST</td><td width="10%">X</td></tr>
<tr><td width="40%">pipe</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMP</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMPE</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMPS</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMPT</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMPTE</td><td width="10%">X</td></tr>
<tr><td width="40%">RTMPTS</td><td width="10%">X</td></tr>
<tr><td width="40%">RTP</td><td width="10%">X</td></tr>
<tr><td width="40%">SAMBA</td><td width="10%">E</td></tr>
<tr><td width="40%">SCTP</td><td width="10%">X</td></tr>
<tr><td width="40%">SFTP</td><td width="10%">E</td></tr>
<tr><td width="40%">TCP</td><td width="10%">X</td></tr>
<tr><td width="40%">TLS</td><td width="10%">X</td></tr>
<tr><td width="40%">UDP</td><td width="10%">X</td></tr>
</table>
<p><code>X</code> means that the protocol is supported.
</p>
<p><code>E</code> means that support is provided through an external library.
</p>
<a name="Input_002fOutput-Devices"></a>
<h3 class="section">2.7 Input/Output Devices<span class="pull-right"><a class="anchor hidden-xs" href="#Input_002fOutput-Devices" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Input_002fOutput-Devices" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Name</td><td width="10%">Input</td><td width="10%">Output</td></tr>
<tr><td width="40%">ALSA</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">BKTR</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">caca</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">DV1394</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">Lavfi virtual device</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">Linux framebuffer</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">JACK</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">LIBCDIO</td><td width="10%">X</td></tr>
<tr><td width="40%">LIBDC1394</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">OpenAL</td><td width="10%">X</td></tr>
<tr><td width="40%">OpenGL</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">OSS</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">PulseAudio</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">SDL</td><td width="10%"></td><td width="10%">X</td></tr>
<tr><td width="40%">Video4Linux2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">VfW capture</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">X11 grabbing</td><td width="10%">X</td><td width="10%"></td></tr>
<tr><td width="40%">Win32 grabbing</td><td width="10%">X</td><td width="10%"></td></tr>
</table>
<p><code>X</code> means that input/output is supported.
</p>
<a name="Timecode"></a>
<h3 class="section">2.8 Timecode<span class="pull-right"><a class="anchor hidden-xs" href="#Timecode" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Timecode" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">Codec/format</td><td width="10%">Read</td><td width="10%">Write</td></tr>
<tr><td width="40%">AVI</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">DV</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">GXF</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MOV</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MPEG1/2</td><td width="10%">X</td><td width="10%">X</td></tr>
<tr><td width="40%">MXF</td><td width="10%">X</td><td width="10%">X</td></tr>
</table>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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Using git to develop FFmpeg
</title>
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<h1>
Using git to develop FFmpeg
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Introduction" href="#Introduction">1 Introduction</a></li>
<li><a name="toc-Basics-Usage" href="#Basics-Usage">2 Basics Usage</a>
<ul class="no-bullet">
<li><a name="toc-Get-GIT" href="#Get-GIT">2.1 Get GIT</a></li>
<li><a name="toc-Cloning-the-source-tree" href="#Cloning-the-source-tree">2.2 Cloning the source tree</a></li>
<li><a name="toc-Updating-the-source-tree-to-the-latest-revision" href="#Updating-the-source-tree-to-the-latest-revision">2.3 Updating the source tree to the latest revision</a></li>
<li><a name="toc-Rebasing-your-local-branches" href="#Rebasing-your-local-branches">2.4 Rebasing your local branches</a></li>
<li><a name="toc-Adding_002fremoving-files_002fdirectories" href="#Adding_002fremoving-files_002fdirectories">2.5 Adding/removing files/directories</a></li>
<li><a name="toc-Showing-modifications" href="#Showing-modifications">2.6 Showing modifications</a></li>
<li><a name="toc-Inspecting-the-changelog" href="#Inspecting-the-changelog">2.7 Inspecting the changelog</a></li>
<li><a name="toc-Checking-source-tree-status" href="#Checking-source-tree-status">2.8 Checking source tree status</a></li>
<li><a name="toc-Committing" href="#Committing">2.9 Committing</a></li>
<li><a name="toc-Preparing-a-patchset" href="#Preparing-a-patchset">2.10 Preparing a patchset</a></li>
<li><a name="toc-Sending-patches-for-review" href="#Sending-patches-for-review">2.11 Sending patches for review</a></li>
<li><a name="toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files">2.12 Renaming/moving/copying files or contents of files</a></li>
</ul></li>
<li><a name="toc-Git-configuration" href="#Git-configuration">3 Git configuration</a>
<ul class="no-bullet">
<li><a name="toc-Personal-Git-installation" href="#Personal-Git-installation">3.1 Personal Git installation</a></li>
<li><a name="toc-Repository-configuration" href="#Repository-configuration">3.2 Repository configuration</a></li>
</ul></li>
<li><a name="toc-FFmpeg-specific" href="#FFmpeg-specific">4 FFmpeg specific</a>
<ul class="no-bullet">
<li><a name="toc-Reverting-broken-commits" href="#Reverting-broken-commits">4.1 Reverting broken commits</a></li>
<li><a name="toc-Pushing-changes-to-remote-trees" href="#Pushing-changes-to-remote-trees">4.2 Pushing changes to remote trees</a></li>
<li><a name="toc-Finding-a-specific-svn-revision" href="#Finding-a-specific-svn-revision">4.3 Finding a specific svn revision</a></li>
</ul></li>
<li><a name="toc-pre_002dpush-checklist" href="#pre_002dpush-checklist">5 pre-push checklist</a></li>
<li><a name="toc-Server-Issues" href="#Server-Issues">6 Server Issues</a></li>
</ul>
</div>
<a name="Introduction"></a>
<h2 class="chapter">1 Introduction<span class="pull-right"><a class="anchor hidden-xs" href="#Introduction" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Introduction" aria-hidden="true">TOC</a></span></h2>
<p>This document aims in giving some quick references on a set of useful git
commands. You should always use the extensive and detailed documentation
provided directly by git:
</p>
<div class="example">
<pre class="example">git --help
man git
</pre></div>
<p>shows you the available subcommands,
</p>
<div class="example">
<pre class="example">git &lt;command&gt; --help
man git-&lt;command&gt;
</pre></div>
<p>shows information about the subcommand &lt;command&gt;.
</p>
<p>Additional information could be found on the
<a href="http://gitref.org">Git Reference</a> website
</p>
<p>For more information about the Git project, visit the
</p>
<p><a href="http://git-scm.com/">Git website</a>
</p>
<p>Consult these resources whenever you have problems, they are quite exhaustive.
</p>
<p>What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project
</p>
<a name="Basics-Usage"></a>
<h2 class="chapter">2 Basics Usage<span class="pull-right"><a class="anchor hidden-xs" href="#Basics-Usage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Basics-Usage" aria-hidden="true">TOC</a></span></h2>
<a name="Get-GIT"></a>
<h3 class="section">2.1 Get GIT<span class="pull-right"><a class="anchor hidden-xs" href="#Get-GIT" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Get-GIT" aria-hidden="true">TOC</a></span></h3>
<p>You can get git from <a href="http://git-scm.com/">http://git-scm.com/</a>
Most distribution and operating system provide a package for it.
</p>
<a name="Cloning-the-source-tree"></a>
<h3 class="section">2.2 Cloning the source tree<span class="pull-right"><a class="anchor hidden-xs" href="#Cloning-the-source-tree" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cloning-the-source-tree" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git clone git://source.ffmpeg.org/ffmpeg &lt;target&gt;
</pre></div>
<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var>.
</p>
<div class="example">
<pre class="example">git clone git@source.ffmpeg.org:ffmpeg &lt;target&gt;
</pre></div>
<p>This will put the FFmpeg sources into the directory <var>&lt;target&gt;</var> and let
you push back your changes to the remote repository.
</p>
<p>Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
</p>
<div class="example">
<pre class="example">git config --global core.autocrlf false
</pre></div>
<a name="Updating-the-source-tree-to-the-latest-revision"></a>
<h3 class="section">2.3 Updating the source tree to the latest revision<span class="pull-right"><a class="anchor hidden-xs" href="#Updating-the-source-tree-to-the-latest-revision" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Updating-the-source-tree-to-the-latest-revision" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git pull (--rebase)
</pre></div>
<p>pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
</p>
<div class="warning">
<p><code>--rebase</code> (see below) is recommended.
</p></div>
<a name="Rebasing-your-local-branches"></a>
<h3 class="section">2.4 Rebasing your local branches<span class="pull-right"><a class="anchor hidden-xs" href="#Rebasing-your-local-branches" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Rebasing-your-local-branches" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git pull --rebase
</pre></div>
<p>fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg&rsquo;s master tree. The master tree will reject pushes with merge commits.
</p>
<a name="Adding_002fremoving-files_002fdirectories"></a>
<h3 class="section">2.5 Adding/removing files/directories<span class="pull-right"><a class="anchor hidden-xs" href="#Adding_002fremoving-files_002fdirectories" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding_002fremoving-files_002fdirectories" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git add [-A] &lt;filename/dirname&gt;
git rm [-r] &lt;filename/dirname&gt;
</pre></div>
<p>GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
</p>
<a name="Showing-modifications"></a>
<h3 class="section">2.6 Showing modifications<span class="pull-right"><a class="anchor hidden-xs" href="#Showing-modifications" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Showing-modifications" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git diff &lt;filename(s)&gt;
</pre></div>
<p>will show all local modifications in your working directory as unified diff.
</p>
<a name="Inspecting-the-changelog"></a>
<h3 class="section">2.7 Inspecting the changelog<span class="pull-right"><a class="anchor hidden-xs" href="#Inspecting-the-changelog" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Inspecting-the-changelog" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git log &lt;filename(s)&gt;
</pre></div>
<p>You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org/
</p>
<a name="Checking-source-tree-status"></a>
<h3 class="section">2.8 Checking source tree status<span class="pull-right"><a class="anchor hidden-xs" href="#Checking-source-tree-status" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Checking-source-tree-status" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git status
</pre></div>
<p>detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
</p>
<a name="Committing"></a>
<h3 class="section">2.9 Committing<span class="pull-right"><a class="anchor hidden-xs" href="#Committing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Committing" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git diff --check
</pre></div>
<p>to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It&rsquo;s very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
</p>
<p>For cosmetics-only commits you should get (almost) empty output from
</p>
<div class="example">
<pre class="example">git diff -w -b &lt;filename(s)&gt;
</pre></div>
<p>Also check the output of
</p>
<div class="example">
<pre class="example">git status
</pre></div>
<p>to make sure you don&rsquo;t have untracked files or deletions.
</p>
<div class="example">
<pre class="example">git add [-i|-p|-A] &lt;filenames/dirnames&gt;
</pre></div>
<p>Make sure you have told git your name and email address
</p>
<div class="example">
<pre class="example">git config --global user.name &quot;My Name&quot;
git config --global user.email my@email.invalid
</pre></div>
<p>Use <var>&ndash;global</var> to set the global configuration for all your git checkouts.
</p>
<p>Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
</p>
<div class="example">
<pre class="example">git commit
</pre></div>
<p>Git will commit the selected changes to your current local branch.
</p>
<p>You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
</p>
<div class="example">
<pre class="example">git config --global core.editor
</pre></div>
<p>or set by one of the following environment variables:
<var>GIT_EDITOR</var>, <var>VISUAL</var> or <var>EDITOR</var>.
</p>
<p>Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just &quot;bug fix&quot; or &quot;10l&quot; is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don&rsquo;t
include filenames in log messages, Git provides that information.
</p>
<p>Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
</p>
<a name="Preparing-a-patchset"></a>
<h3 class="section">2.10 Preparing a patchset<span class="pull-right"><a class="anchor hidden-xs" href="#Preparing-a-patchset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Preparing-a-patchset" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git format-patch &lt;commit&gt; [-o directory]
</pre></div>
<p>will generate a set of patches for each commit between <var>&lt;commit&gt;</var> and
current <var>HEAD</var>. E.g.
</p>
<div class="example">
<pre class="example">git format-patch origin/master
</pre></div>
<p>will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
</p>
<div class="example">
<pre class="example">git format-patch -n
</pre></div>
<p>which will generate patches from last <var>n</var> commits.
By default the patches are created in the current directory.
</p>
<a name="Sending-patches-for-review"></a>
<h3 class="section">2.11 Sending patches for review<span class="pull-right"><a class="anchor hidden-xs" href="#Sending-patches-for-review" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Sending-patches-for-review" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git send-email &lt;commit list|directory&gt;
</pre></div>
<p>will send the patches created by <code>git format-patch</code> or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. <var>git-email</var>
package on Debian-based distros).
</p>
<a name="Renaming_002fmoving_002fcopying-files-or-contents-of-files"></a>
<h3 class="section">2.12 Renaming/moving/copying files or contents of files<span class="pull-right"><a class="anchor hidden-xs" href="#Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Renaming_002fmoving_002fcopying-files-or-contents-of-files" aria-hidden="true">TOC</a></span></h3>
<p>Git automatically tracks such changes, making those normal commits.
</p>
<div class="example">
<pre class="example">mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
</pre></div>
<a name="Git-configuration"></a>
<h2 class="chapter">3 Git configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Git-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Git-configuration" aria-hidden="true">TOC</a></span></h2>
<p>In order to simplify a few workflows, it is advisable to configure both
your personal Git installation and your local FFmpeg repository.
</p>
<a name="Personal-Git-installation"></a>
<h3 class="section">3.1 Personal Git installation<span class="pull-right"><a class="anchor hidden-xs" href="#Personal-Git-installation" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Personal-Git-installation" aria-hidden="true">TOC</a></span></h3>
<p>Add the following to your <samp>~/.gitconfig</samp> to help <code>git send-email</code>
and <code>git format-patch</code> detect renames:
</p>
<div class="example">
<pre class="example">[diff]
renames = copy
</pre></div>
<a name="Repository-configuration"></a>
<h3 class="section">3.2 Repository configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Repository-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Repository-configuration" aria-hidden="true">TOC</a></span></h3>
<p>In order to have <code>git send-email</code> automatically send patches
to the ffmpeg-devel mailing list, add the following stanza
to <samp>/path/to/ffmpeg/repository/.git/config</samp>:
</p>
<div class="example">
<pre class="example">[sendemail]
to = ffmpeg-devel@ffmpeg.org
</pre></div>
<a name="FFmpeg-specific"></a>
<h2 class="chapter">4 FFmpeg specific<span class="pull-right"><a class="anchor hidden-xs" href="#FFmpeg-specific" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-FFmpeg-specific" aria-hidden="true">TOC</a></span></h2>
<a name="Reverting-broken-commits"></a>
<h3 class="section">4.1 Reverting broken commits<span class="pull-right"><a class="anchor hidden-xs" href="#Reverting-broken-commits" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Reverting-broken-commits" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git reset &lt;commit&gt;
</pre></div>
<p><code>git reset</code> will uncommit the changes till <var>&lt;commit&gt;</var> rewriting
the current branch history.
</p>
<div class="example">
<pre class="example">git commit --amend
</pre></div>
<p>allows one to amend the last commit details quickly.
</p>
<div class="example">
<pre class="example">git rebase -i origin/master
</pre></div>
<p>will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
</p>
<div class="info">
<p><code>git reset</code>, <code>git commit --amend</code> and <code>git rebase</code>
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
</p></div>
<div class="example">
<pre class="example">git revert &lt;commit&gt;
</pre></div>
<p><code>git revert</code> will generate a revert commit. This will not make the
faulty commit disappear from the history.
</p>
<a name="Pushing-changes-to-remote-trees"></a>
<h3 class="section">4.2 Pushing changes to remote trees<span class="pull-right"><a class="anchor hidden-xs" href="#Pushing-changes-to-remote-trees" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Pushing-changes-to-remote-trees" aria-hidden="true">TOC</a></span></h3>
<div class="example">
<pre class="example">git push
</pre></div>
<p>Will push the changes to the default remote (<var>origin</var>).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to and to sync the local tree.
</p>
<div class="example">
<pre class="example">git remote add &lt;name&gt; &lt;url&gt;
</pre></div>
<p>Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
</p>
<div class="example">
<pre class="example">git push &lt;remote&gt; &lt;refspec&gt;
</pre></div>
<p>Will push the changes to the <var>&lt;remote&gt;</var> repository.
Omitting <var>&lt;refspec&gt;</var> makes <code>git push</code> update all the remote
branches matching the local ones.
</p>
<a name="Finding-a-specific-svn-revision"></a>
<h3 class="section">4.3 Finding a specific svn revision<span class="pull-right"><a class="anchor hidden-xs" href="#Finding-a-specific-svn-revision" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Finding-a-specific-svn-revision" aria-hidden="true">TOC</a></span></h3>
<p>Since version 1.7.1 git supports <var>:/foo</var> syntax for specifying commits
based on a regular expression. see man gitrevisions
</p>
<div class="example">
<pre class="example">git show :/'as revision 23456'
</pre></div>
<p>will show the svn changeset <var>r23456</var>. With older git versions searching in
the <code>git log</code> output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
</p>
<div class="example">
<pre class="example">git checkout -b svn_23456 :/'as revision 23456'
</pre></div>
<p>or for git &lt; 1.7.1 with
</p>
<div class="example">
<pre class="example">git checkout -b svn_23456 $SHA1
</pre></div>
<p>where <var>$SHA1</var> is the commit hash from the <code>git log</code> output.
</p>
<a name="pre_002dpush-checklist"></a>
<h2 class="chapter">5 pre-push checklist<span class="pull-right"><a class="anchor hidden-xs" href="#pre_002dpush-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-pre_002dpush-checklist" aria-hidden="true">TOC</a></span></h2>
<p>Once you have a set of commits that you feel are ready for pushing,
work through the following checklist to doublecheck everything is in
proper order. This list tries to be exhaustive. In case you are just
pushing a typo in a comment, some of the steps may be unnecessary.
Apply your common sense, but if in doubt, err on the side of caution.
</p>
<p>First, make sure that the commits and branches you are going to push
match what you want pushed and that nothing is missing, extraneous or
wrong. You can see what will be pushed by running the git push command
with &ndash;dry-run first. And then inspecting the commits listed with
<code>git log -p 1234567..987654</code>. The <code>git status</code> command
may help in finding local changes that have been forgotten to be added.
</p>
<p>Next let the code pass through a full run of our testsuite.
</p>
<ul>
<li> <code>make distclean</code>
</li><li> <code>/path/to/ffmpeg/configure</code>
</li><li> <code>make check</code>
</li><li> if fate fails due to missing samples run <code>make fate-rsync</code> and retry
</li></ul>
<p>Make sure all your changes have been checked before pushing them, the
testsuite only checks against regressions and that only to some extend. It does
obviously not check newly added features/code to be working unless you have
added a test for that (which is recommended).
</p>
<p>Also note that every single commit should pass the test suite, not just
the result of a series of patches.
</p>
<p>Once everything passed, push the changes to your public ffmpeg clone and post a
merge request to ffmpeg-devel. You can also push them directly but this is not
recommended.
</p>
<a name="Server-Issues"></a>
<h2 class="chapter">6 Server Issues<span class="pull-right"><a class="anchor hidden-xs" href="#Server-Issues" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Server-Issues" aria-hidden="true">TOC</a></span></h2>
<p>Contact the project admins <a href="mailto:root@ffmpeg.org">root@ffmpeg.org</a> if you have technical
problems with the GIT server.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavcodec Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libavcodec Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libavcodec library provides a generic encoding/decoding framework
and contains multiple decoders and encoders for audio, video and
subtitle streams, and several bitstream filters.
</p>
<p>The shared architecture provides various services ranging from bit
stream I/O to DSP optimizations, and makes it suitable for
implementing robust and fast codecs as well as for experimentation.
</p>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-codecs.html">ffmpeg-codecs</a>, <a href="ffmpeg-bitstream-filters.html">bitstream-filters</a>,
<a href="libavutil.html">libavutil</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavdevice Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libavdevice Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libavdevice library provides a generic framework for grabbing from
and rendering to many common multimedia input/output devices, and
supports several input and output devices, including Video4Linux2,
VfW, DShow, and ALSA.
</p>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-devices.html">ffmpeg-devices</a>,
<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavfilter Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libavfilter Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libavfilter library provides a generic audio/video filtering
framework containing several filters, sources and sinks.
</p>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-filters.html">ffmpeg-filters</a>,
<a href="libavutil.html">libavutil</a>, <a href="libswscale.html">libswscale</a>, <a href="libswresample.html">libswresample</a>,
<a href="libavcodec.html">libavcodec</a>, <a href="libavformat.html">libavformat</a>, <a href="libavdevice.html">libavdevice</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavformat Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libavformat Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libavformat library provides a generic framework for multiplexing
and demultiplexing (muxing and demuxing) audio, video and subtitle
streams. It encompasses multiple muxers and demuxers for multimedia
container formats.
</p>
<p>It also supports several input and output protocols to access a media
resource.
</p>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-formats.html">ffmpeg-formats</a>, <a href="ffmpeg-protocols.html">ffmpeg-protocols</a>,
<a href="libavutil.html">libavutil</a>, <a href="libavcodec.html">libavcodec</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libavutil Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libavutil Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats). It is not a library for
code needed by both libavcodec and libavformat.
</p>
<p>The goals for this library is to be:
</p>
<dl compact="compact">
<dt><strong>Modular</strong></dt>
<dd><p>It should have few interdependencies and the possibility of disabling individual
parts during <code>./configure</code>.
</p>
</dd>
<dt><strong>Small</strong></dt>
<dd><p>Both sources and objects should be small.
</p>
</dd>
<dt><strong>Efficient</strong></dt>
<dd><p>It should have low CPU and memory usage.
</p>
</dd>
<dt><strong>Useful</strong></dt>
<dd><p>It should avoid useless features that almost no one needs.
</p></dd>
</dl>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-utils.html">ffmpeg-utils</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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Externals/ffmpeg/dev/doc/libswresample.html vendored
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libswresample Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libswresample Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libswresample library performs highly optimized audio resampling,
rematrixing and sample format conversion operations.
</p>
<p>Specifically, this library performs the following conversions:
</p>
<ul>
<li> <em>Resampling</em>: is the process of changing the audio rate, for
example from a high sample rate of 44100Hz to 8000Hz. Audio
conversion from high to low sample rate is a lossy process. Several
resampling options and algorithms are available.
</li><li> <em>Format conversion</em>: is the process of converting the type of
samples, for example from 16-bit signed samples to unsigned 8-bit or
float samples. It also handles packing conversion, when passing from
packed layout (all samples belonging to distinct channels interleaved
in the same buffer), to planar layout (all samples belonging to the
same channel stored in a dedicated buffer or &quot;plane&quot;).
</li><li> <em>Rematrixing</em>: is the process of changing the channel layout, for
example from stereo to mono. When the input channels cannot be mapped
to the output streams, the process is lossy, since it involves
different gain factors and mixing.
</li></ul>
<p>Various other audio conversions (e.g. stretching and padding) are
enabled through dedicated options.
</p>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-resampler.html">ffmpeg-resampler</a>,
<a href="libavutil.html">libavutil</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

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Externals/ffmpeg/dev/doc/libswscale.html vendored
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Libswscale Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Libswscale Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-See-Also" href="#See-Also">2 See Also</a></li>
<li><a name="toc-Authors" href="#Authors">3 Authors</a></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>The libswscale library performs highly optimized image scaling and
colorspace and pixel format conversion operations.
</p>
<p>Specifically, this library performs the following conversions:
</p>
<ul>
<li> <em>Rescaling</em>: is the process of changing the video size. Several
rescaling options and algorithms are available. This is usually a
lossy process.
</li><li> <em>Pixel format conversion</em>: is the process of converting the image
format and colorspace of the image, for example from planar YUV420P to
RGB24 packed. It also handles packing conversion, that is converts
from packed layout (all pixels belonging to distinct planes
interleaved in the same buffer), to planar layout (all samples
belonging to the same plane stored in a dedicated buffer or &quot;plane&quot;).
<p>This is usually a lossy process in case the source and destination
colorspaces differ.
</p></li></ul>
<a name="See-Also"></a>
<h2 class="chapter">2 See Also<span class="pull-right"><a class="anchor hidden-xs" href="#See-Also" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-See-Also" aria-hidden="true">TOC</a></span></h2>
<p><a href="ffmpeg.html">ffmpeg</a>, <a href="ffplay.html">ffplay</a>, <a href="ffprobe.html">ffprobe</a>, <a href="ffserver.html">ffserver</a>,
<a href="ffmpeg-scaler.html">ffmpeg-scaler</a>,
<a href="libavutil.html">libavutil</a>
</p>
<a name="Authors"></a>
<h2 class="chapter">3 Authors<span class="pull-right"><a class="anchor hidden-xs" href="#Authors" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Authors" aria-hidden="true">TOC</a></span></h2>
<p>The FFmpeg developers.
</p>
<p>For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
<code>git log</code> in the FFmpeg source directory, or browsing the
online repository at <a href="http://source.ffmpeg.org">http://source.ffmpeg.org</a>.
</p>
<p>Maintainers for the specific components are listed in the file
<samp>MAINTAINERS</samp> in the source code tree.
</p>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

211
Externals/ffmpeg/dev/doc/nut.html vendored
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
NUT
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
NUT
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Description" href="#Description">1 Description</a></li>
<li><a name="toc-Modes" href="#Modes">2 Modes</a>
<ul class="no-bullet">
<li><a name="toc-BROADCAST" href="#BROADCAST">2.1 BROADCAST</a></li>
<li><a name="toc-PIPE" href="#PIPE">2.2 PIPE</a></li>
</ul></li>
<li><a name="toc-Container_002dspecific-codec-tags" href="#Container_002dspecific-codec-tags">3 Container-specific codec tags</a>
<ul class="no-bullet">
<li><a name="toc-Generic-raw-YUVA-formats" href="#Generic-raw-YUVA-formats">3.1 Generic raw YUVA formats</a></li>
<li><a name="toc-Raw-Audio" href="#Raw-Audio">3.2 Raw Audio</a></li>
<li><a name="toc-Subtitles" href="#Subtitles">3.3 Subtitles</a></li>
<li><a name="toc-Raw-Data" href="#Raw-Data">3.4 Raw Data</a></li>
<li><a name="toc-Codecs" href="#Codecs">3.5 Codecs</a></li>
</ul></li>
</ul>
</div>
<a name="Description"></a>
<h2 class="chapter">1 Description<span class="pull-right"><a class="anchor hidden-xs" href="#Description" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Description" aria-hidden="true">TOC</a></span></h2>
<p>NUT is a low overhead generic container format. It stores audio, video,
subtitle and user-defined streams in a simple, yet efficient, way.
</p>
<p>It was created by a group of FFmpeg and MPlayer developers in 2003
and was finalized in 2008.
</p>
<p>The official nut specification is at svn://svn.mplayerhq.hu/nut
In case of any differences between this text and the official specification,
the official specification shall prevail.
</p>
<a name="Modes"></a>
<h2 class="chapter">2 Modes<span class="pull-right"><a class="anchor hidden-xs" href="#Modes" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Modes" aria-hidden="true">TOC</a></span></h2>
<p>NUT has some variants signaled by using the flags field in its main header.
</p>
<table>
<tr><td width="40%">BROADCAST</td><td width="40%">Extend the syncpoint to report the sender wallclock</td></tr>
<tr><td width="40%">PIPE</td><td width="40%">Omit completely the syncpoint</td></tr>
</table>
<a name="BROADCAST"></a>
<h3 class="section">2.1 BROADCAST<span class="pull-right"><a class="anchor hidden-xs" href="#BROADCAST" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BROADCAST" aria-hidden="true">TOC</a></span></h3>
<p>The BROADCAST variant provides a secondary time reference to facilitate
detecting endpoint latency and network delays.
It assumes all the endpoint clocks are syncronized.
To be used in real-time scenarios.
</p>
<a name="PIPE"></a>
<h3 class="section">2.2 PIPE<span class="pull-right"><a class="anchor hidden-xs" href="#PIPE" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-PIPE" aria-hidden="true">TOC</a></span></h3>
<p>The PIPE variant assumes NUT is used as non-seekable intermediate container,
by not using syncpoint removes unneeded overhead and reduces the overall
memory usage.
</p>
<a name="Container_002dspecific-codec-tags"></a>
<h2 class="chapter">3 Container-specific codec tags<span class="pull-right"><a class="anchor hidden-xs" href="#Container_002dspecific-codec-tags" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Container_002dspecific-codec-tags" aria-hidden="true">TOC</a></span></h2>
<a name="Generic-raw-YUVA-formats"></a>
<h3 class="section">3.1 Generic raw YUVA formats<span class="pull-right"><a class="anchor hidden-xs" href="#Generic-raw-YUVA-formats" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Generic-raw-YUVA-formats" aria-hidden="true">TOC</a></span></h3>
<p>Since many exotic planar YUVA pixel formats are not considered by
the AVI/QuickTime FourCC lists, the following scheme is adopted for
representing them.
</p>
<p>The first two bytes can contain the values:
Y1 = only Y
Y2 = Y+A
Y3 = YUV
Y4 = YUVA
</p>
<p>The third byte represents the width and height chroma subsampling
values for the UV planes, that is the amount to shift the luma
width/height right to find the chroma width/height.
</p>
<p>The fourth byte is the number of bits used (8, 16, ...).
</p>
<p>If the order of bytes is inverted, that means that each component has
to be read big-endian.
</p>
<a name="Raw-Audio"></a>
<h3 class="section">3.2 Raw Audio<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Audio" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Audio" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">ALAW</td><td width="40%">A-LAW</td></tr>
<tr><td width="40%">ULAW</td><td width="40%">MU-LAW</td></tr>
<tr><td width="40%">P&lt;type&gt;&lt;interleaving&gt;&lt;bits&gt;</td><td width="40%">little-endian PCM</td></tr>
<tr><td width="40%">&lt;bits&gt;&lt;interleaving&gt;&lt;type&gt;P</td><td width="40%">big-endian PCM</td></tr>
</table>
<p>&lt;type&gt; is S for signed integer, U for unsigned integer, F for IEEE float
&lt;interleaving&gt; is D for default, P is for planar.
&lt;bits&gt; is 8/16/24/32
</p>
<div class="example">
<pre class="example">PFD[32] would for example be signed 32 bit little-endian IEEE float
</pre></div>
<a name="Subtitles"></a>
<h3 class="section">3.3 Subtitles<span class="pull-right"><a class="anchor hidden-xs" href="#Subtitles" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Subtitles" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
<tr><td width="40%">SSA[0]</td><td width="40%">SubStation Alpha</td></tr>
<tr><td width="40%">DVDS</td><td width="40%">DVD subtitles</td></tr>
<tr><td width="40%">DVBS</td><td width="40%">DVB subtitles</td></tr>
</table>
<a name="Raw-Data"></a>
<h3 class="section">3.4 Raw Data<span class="pull-right"><a class="anchor hidden-xs" href="#Raw-Data" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Raw-Data" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">UTF8</td><td width="40%">Raw UTF-8</td></tr>
</table>
<a name="Codecs"></a>
<h3 class="section">3.5 Codecs<span class="pull-right"><a class="anchor hidden-xs" href="#Codecs" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Codecs" aria-hidden="true">TOC</a></span></h3>
<table>
<tr><td width="40%">3IV1</td><td width="40%">non-compliant MPEG-4 generated by old 3ivx</td></tr>
<tr><td width="40%">ASV1</td><td width="40%">Asus Video</td></tr>
<tr><td width="40%">ASV2</td><td width="40%">Asus Video 2</td></tr>
<tr><td width="40%">CVID</td><td width="40%">Cinepak</td></tr>
<tr><td width="40%">CYUV</td><td width="40%">Creative YUV</td></tr>
<tr><td width="40%">DIVX</td><td width="40%">non-compliant MPEG-4 generated by old DivX</td></tr>
<tr><td width="40%">DUCK</td><td width="40%">Truemotion 1</td></tr>
<tr><td width="40%">FFV1</td><td width="40%">FFmpeg video 1</td></tr>
<tr><td width="40%">FFVH</td><td width="40%">FFmpeg Huffyuv</td></tr>
<tr><td width="40%">H261</td><td width="40%">ITU H.261</td></tr>
<tr><td width="40%">H262</td><td width="40%">ITU H.262</td></tr>
<tr><td width="40%">H263</td><td width="40%">ITU H.263</td></tr>
<tr><td width="40%">H264</td><td width="40%">ITU H.264</td></tr>
<tr><td width="40%">HFYU</td><td width="40%">Huffyuv</td></tr>
<tr><td width="40%">I263</td><td width="40%">Intel H.263</td></tr>
<tr><td width="40%">IV31</td><td width="40%">Indeo 3.1</td></tr>
<tr><td width="40%">IV32</td><td width="40%">Indeo 3.2</td></tr>
<tr><td width="40%">IV50</td><td width="40%">Indeo 5.0</td></tr>
<tr><td width="40%">LJPG</td><td width="40%">ITU JPEG (lossless)</td></tr>
<tr><td width="40%">MJLS</td><td width="40%">ITU JPEG-LS</td></tr>
<tr><td width="40%">MJPG</td><td width="40%">ITU JPEG</td></tr>
<tr><td width="40%">MPG4</td><td width="40%">MS MPEG-4v1 (not ISO MPEG-4)</td></tr>
<tr><td width="40%">MP42</td><td width="40%">MS MPEG-4v2</td></tr>
<tr><td width="40%">MP43</td><td width="40%">MS MPEG-4v3</td></tr>
<tr><td width="40%">MP4V</td><td width="40%">ISO MPEG-4 Part 2 Video (from old encoders)</td></tr>
<tr><td width="40%">mpg1</td><td width="40%">ISO MPEG-1 Video</td></tr>
<tr><td width="40%">mpg2</td><td width="40%">ISO MPEG-2 Video</td></tr>
<tr><td width="40%">MRLE</td><td width="40%">MS RLE</td></tr>
<tr><td width="40%">MSVC</td><td width="40%">MS Video 1</td></tr>
<tr><td width="40%">RT21</td><td width="40%">Indeo 2.1</td></tr>
<tr><td width="40%">RV10</td><td width="40%">RealVideo 1.0</td></tr>
<tr><td width="40%">RV20</td><td width="40%">RealVideo 2.0</td></tr>
<tr><td width="40%">RV30</td><td width="40%">RealVideo 3.0</td></tr>
<tr><td width="40%">RV40</td><td width="40%">RealVideo 4.0</td></tr>
<tr><td width="40%">SNOW</td><td width="40%">FFmpeg Snow</td></tr>
<tr><td width="40%">SVQ1</td><td width="40%">Sorenson Video 1</td></tr>
<tr><td width="40%">SVQ3</td><td width="40%">Sorenson Video 3</td></tr>
<tr><td width="40%">theo</td><td width="40%">Xiph Theora</td></tr>
<tr><td width="40%">TM20</td><td width="40%">Truemotion 2.0</td></tr>
<tr><td width="40%">UMP4</td><td width="40%">non-compliant MPEG-4 generated by UB Video MPEG-4</td></tr>
<tr><td width="40%">VCR1</td><td width="40%">ATI VCR1</td></tr>
<tr><td width="40%">VP30</td><td width="40%">VP 3.0</td></tr>
<tr><td width="40%">VP31</td><td width="40%">VP 3.1</td></tr>
<tr><td width="40%">VP50</td><td width="40%">VP 5.0</td></tr>
<tr><td width="40%">VP60</td><td width="40%">VP 6.0</td></tr>
<tr><td width="40%">VP61</td><td width="40%">VP 6.1</td></tr>
<tr><td width="40%">VP62</td><td width="40%">VP 6.2</td></tr>
<tr><td width="40%">VP70</td><td width="40%">VP 7.0</td></tr>
<tr><td width="40%">WMV1</td><td width="40%">MS WMV7</td></tr>
<tr><td width="40%">WMV2</td><td width="40%">MS WMV8</td></tr>
<tr><td width="40%">WMV3</td><td width="40%">MS WMV9</td></tr>
<tr><td width="40%">WV1F</td><td width="40%">non-compliant MPEG-4 generated by ?</td></tr>
<tr><td width="40%">WVC1</td><td width="40%">VC-1</td></tr>
<tr><td width="40%">XVID</td><td width="40%">non-compliant MPEG-4 generated by old Xvid</td></tr>
<tr><td width="40%">XVIX</td><td width="40%">non-compliant MPEG-4 generated by old Xvid with interlacing bug</td></tr>
</table>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
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<title>
Platform Specific Information
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
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<h1>
Platform Specific Information
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Unix_002dlike" href="#Unix_002dlike">1 Unix-like</a>
<ul class="no-bullet">
<li><a name="toc-Advanced-linking-configuration" href="#Advanced-linking-configuration">1.1 Advanced linking configuration</a></li>
<li><a name="toc-BSD" href="#BSD">1.2 BSD</a></li>
<li><a name="toc-_0028Open_0029Solaris" href="#g_t_0028Open_0029Solaris">1.3 (Open)Solaris</a></li>
<li><a name="toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029">1.4 Darwin (Mac OS X, iPhone)</a></li>
</ul></li>
<li><a name="toc-DOS" href="#DOS">2 DOS</a></li>
<li><a name="toc-OS_002f2" href="#OS_002f2">3 OS/2</a></li>
<li><a name="toc-Windows" href="#Windows">4 Windows</a>
<ul class="no-bullet">
<li><a name="toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64">4.1 Native Windows compilation using MinGW or MinGW-w64</a></li>
<li><a name="toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows</a>
<ul class="no-bullet">
<li><a name="toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b">4.2.1 Linking to FFmpeg with Microsoft Visual C++</a></li>
</ul></li>
<li><a name="toc-Cross-compilation-for-Windows-with-Linux-1" href="#Cross-compilation-for-Windows-with-Linux-1">4.3 Cross compilation for Windows with Linux</a></li>
<li><a name="toc-Compilation-under-Cygwin" href="#Compilation-under-Cygwin">4.4 Compilation under Cygwin</a></li>
<li><a name="toc-Crosscompilation-for-Windows-under-Cygwin" href="#Crosscompilation-for-Windows-under-Cygwin">4.5 Crosscompilation for Windows under Cygwin</a></li>
</ul></li>
<li><a name="toc-Plan-9" href="#Plan-9">5 Plan 9</a></li>
</ul>
</div>
<a name="Unix_002dlike"></a>
<h2 class="chapter">1 Unix-like<span class="pull-right"><a class="anchor hidden-xs" href="#Unix_002dlike" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Unix_002dlike" aria-hidden="true">TOC</a></span></h2>
<p>Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
</p>
<div class="example">
<pre class="example">$(gcc -print-prog-name=as) --version
</pre></div>
<p>If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass <code>--disable-asm</code>
to configure.
</p>
<a name="Advanced-linking-configuration"></a>
<h3 class="section">1.1 Advanced linking configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Advanced-linking-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Advanced-linking-configuration" aria-hidden="true">TOC</a></span></h3>
<p>If you compiled FFmpeg libraries statically and you want to use them to
build your own shared library, you may need to force PIC support (with
<code>--enable-pic</code> during FFmpeg configure) and add the following option
to your project LDFLAGS:
</p>
<div class="example">
<pre class="example">-Wl,-Bsymbolic
</pre></div>
<p>If your target platform requires position independent binaries, you should
pass the correct linking flag (e.g. <code>-pie</code>) to <code>--extra-ldexeflags</code>.
</p>
<a name="BSD"></a>
<h3 class="section">1.2 BSD<span class="pull-right"><a class="anchor hidden-xs" href="#BSD" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-BSD" aria-hidden="true">TOC</a></span></h3>
<p>BSD make will not build FFmpeg, you need to install and use GNU Make
(<code>gmake</code>).
</p>
<a name="g_t_0028Open_0029Solaris"></a>
<h3 class="section">1.3 (Open)Solaris<span class="pull-right"><a class="anchor hidden-xs" href="#_0028Open_0029Solaris" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-_0028Open_0029Solaris" aria-hidden="true">TOC</a></span></h3>
<p>GNU Make is required to build FFmpeg, so you have to invoke (<code>gmake</code>),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either <code>--extra-libs=/usr/lib/values-xpg6.o</code>
or <code>--extra-libs=/usr/lib/64/values-xpg6.o</code> to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
</p>
<div class="example">
<pre class="example">bash ./configure
</pre></div>
<a name="Darwin"></a><a name="Darwin-_0028Mac-OS-X_002c-iPhone_0029"></a>
<h3 class="section">1.4 Darwin (Mac OS X, iPhone)<span class="pull-right"><a class="anchor hidden-xs" href="#Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Darwin-_0028Mac-OS-X_002c-iPhone_0029" aria-hidden="true">TOC</a></span></h3>
<p>The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
</p>
<p>Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
<a href="https://github.com/FFmpeg/gas-preprocessor">https://github.com/FFmpeg/gas-preprocessor</a> or
<a href="https://github.com/yuvi/gas-preprocessor">https://github.com/yuvi/gas-preprocessor</a>(currently outdated) to build the optimized
assembly functions. Put the Perl script somewhere
in your PATH, FFmpeg&rsquo;s configure will pick it up automatically.
</p>
<p>Mac OS X on amd64 and x86 requires <code>yasm</code> to build most of the
optimized assembly functions. <a href="http://www.finkproject.org/">Fink</a>,
<a href="http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml">Gentoo Prefix</a>,
<a href="https://mxcl.github.com/homebrew/">Homebrew</a>
or <a href="http://www.macports.org">MacPorts</a> can easily provide it.
</p>
<a name="DOS"></a>
<h2 class="chapter">2 DOS<span class="pull-right"><a class="anchor hidden-xs" href="#DOS" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-DOS" aria-hidden="true">TOC</a></span></h2>
<p>Using a cross-compiler is preferred for various reasons.
<a href="http://www.delorie.com/howto/djgpp/linux-x-djgpp.html">http://www.delorie.com/howto/djgpp/linux-x-djgpp.html</a>
</p>
<a name="OS_002f2"></a>
<h2 class="chapter">3 OS/2<span class="pull-right"><a class="anchor hidden-xs" href="#OS_002f2" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-OS_002f2" aria-hidden="true">TOC</a></span></h2>
<p>For information about compiling FFmpeg on OS/2 see
<a href="http://www.edm2.com/index.php/FFmpeg">http://www.edm2.com/index.php/FFmpeg</a>.
</p>
<a name="Windows"></a>
<h2 class="chapter">4 Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Windows" aria-hidden="true">TOC</a></span></h2>
<p>To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at <a href="http://ffmpeg.zeranoe.com/forum/">http://ffmpeg.zeranoe.com/forum/</a>.
</p>
<a name="Native-Windows-compilation-using-MinGW-or-MinGW_002dw64"></a>
<h3 class="section">4.1 Native Windows compilation using MinGW or MinGW-w64<span class="pull-right"><a class="anchor hidden-xs" href="#Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Native-Windows-compilation-using-MinGW-or-MinGW_002dw64" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
<a href="http://www.mingw.org/">http://www.mingw.org/</a> or <a href="http://mingw-w64.sourceforge.net/">http://mingw-w64.sourceforge.net/</a>.
You can find detailed installation instructions in the download section and
the FAQ.
</p>
<p>Notes:
</p>
<ul>
<li> Building natively using MSYS can be sped up by disabling implicit rules
in the Makefile by calling <code>make -r</code> instead of plain <code>make</code>. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example during
<code>make install</code>).
</li><li> In order to compile FFplay, you must have the MinGW development library
of <a href="http://www.libsdl.org/">SDL</a> and <code>pkg-config</code> installed.
</li><li> By using <code>./configure --enable-shared</code> when configuring FFmpeg,
you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
libavformat) as DLLs.
</li></ul>
<a name="Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows"></a>
<h3 class="section">4.2 Microsoft Visual C++ or Intel C++ Compiler for Windows<span class="pull-right"><a class="anchor hidden-xs" href="#Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Microsoft-Visual-C_002b_002b-or-Intel-C_002b_002b-Compiler-for-Windows" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility
and wrapper, or with MSVC 2013 and ICL natively.
</p>
<p>You will need the following prerequisites:
</p>
<ul>
<li> <a href="https://github.com/libav/c99-to-c89/">C99-to-C89 Converter &amp; Wrapper</a>
(if using MSVC 2012 or earlier)
</li><li> <a href="http://code.google.com/p/msinttypes/">msinttypes</a>
(if using MSVC 2012 or earlier)
</li><li> <a href="http://www.mingw.org/">MSYS</a>
</li><li> <a href="http://yasm.tortall.net/">YASM</a>
</li><li> <a href="http://gnuwin32.sourceforge.net/packages/bc.htm">bc for Windows</a> if
you want to run <a href="fate.html">FATE</a>.
</li></ul>
<p>To set up a proper environment in MSYS, you need to run <code>msys.bat</code> from
the Visual Studio or Intel Compiler command prompt.
</p>
<p>Place <code>yasm.exe</code> somewhere in your <code>PATH</code>. If using MSVC 2012 or
earlier, place <code>c99wrap.exe</code> and <code>c99conv.exe</code> somewhere in your
<code>PATH</code> as well.
</p>
<p>Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the <code>LIB</code>
and <code>INCLUDE</code> environment variables to include the <strong>Windows-style</strong>
paths to these directories. Alternatively, you can try and use the
<code>--extra-cflags</code>/<code>--extra-ldflags</code> configure options. If using MSVC
2012 or earlier, place <code>inttypes.h</code> somewhere the compiler can see too.
</p>
<p>Finally, run:
</p>
<div class="example">
<pre class="example">For MSVC:
./configure --toolchain=msvc
For ICL:
./configure --toolchain=icl
make
make install
</pre></div>
<p>If you wish to compile shared libraries, add <code>--enable-shared</code> to your
configure options. Note that due to the way MSVC and ICL handle DLL imports and
exports, you cannot compile static and shared libraries at the same time, and
enabling shared libraries will automatically disable the static ones.
</p>
<p>Notes:
</p>
<ul>
<li> It is possible that coreutils&rsquo; <code>link.exe</code> conflicts with MSVC&rsquo;s linker.
You can find out by running <code>which link</code> to see which <code>link.exe</code> you
are using. If it is located at <code>/bin/link.exe</code>, then you have the wrong one
in your <code>PATH</code>. Either move or remove that copy, or make sure MSVC&rsquo;s
<code>link.exe</code> takes precedence in your <code>PATH</code> over coreutils&rsquo;.
</li><li> If you wish to build with zlib support, you will have to grab a compatible
zlib binary from somewhere, with an MSVC import lib, or if you wish to link
statically, you can follow the instructions below to build a compatible
<code>zlib.lib</code> with MSVC. Regardless of which method you use, you must still
follow step 3, or compilation will fail.
<ol>
<li> Grab the <a href="http://zlib.net/">zlib sources</a>.
</li><li> Edit <code>win32/Makefile.msc</code> so that it uses -MT instead of -MD, since
this is how FFmpeg is built as well.
</li><li> Edit <code>zconf.h</code> and remove its inclusion of <code>unistd.h</code>. This gets
erroneously included when building FFmpeg.
</li><li> Run <code>nmake -f win32/Makefile.msc</code>.
</li><li> Move <code>zlib.lib</code>, <code>zconf.h</code>, and <code>zlib.h</code> to somewhere MSVC
can see.
</li></ol>
</li><li> FFmpeg has been tested with the following on i686 and x86_64:
<ul>
<li> Visual Studio 2010 Pro and Express
</li><li> Visual Studio 2012 Pro and Express
</li><li> Visual Studio 2013 Pro and Express
</li><li> Intel Composer XE 2013
</li><li> Intel Composer XE 2013 SP1
</li></ul>
<p>Anything else is not officially supported.
</p>
</li></ul>
<a name="Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b"></a>
<h4 class="subsection">4.2.1 Linking to FFmpeg with Microsoft Visual C++<span class="pull-right"><a class="anchor hidden-xs" href="#Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Linking-to-FFmpeg-with-Microsoft-Visual-C_002b_002b" aria-hidden="true">TOC</a></span></h4>
<p>If you plan to link with MSVC-built static libraries, you will need
to make sure you have <code>Runtime Library</code> set to
<code>Multi-threaded (/MT)</code> in your project&rsquo;s settings.
</p>
<p>You will need to define <code>inline</code> to something MSVC understands:
</p><div class="example">
<pre class="example">#define inline __inline
</pre></div>
<p>Also note, that as stated in <strong>Microsoft Visual C++</strong>, you will need
an MSVC-compatible <a href="http://code.google.com/p/msinttypes/">inttypes.h</a>.
</p>
<p>If you plan on using import libraries created by dlltool, you must
set <code>References</code> to <code>No (/OPT:NOREF)</code> under the linker optimization
settings, otherwise the resulting binaries will fail during runtime.
This is not required when using import libraries generated by <code>lib.exe</code>.
This issue is reported upstream at
<a href="http://sourceware.org/bugzilla/show_bug.cgi?id=12633">http://sourceware.org/bugzilla/show_bug.cgi?id=12633</a>.
</p>
<p>To create import libraries that work with the <code>/OPT:REF</code> option
(which is enabled by default in Release mode), follow these steps:
</p>
<ol>
<li> Open the <em>Visual Studio Command Prompt</em>.
<p>Alternatively, in a normal command line prompt, call <samp>vcvars32.bat</samp>
which sets up the environment variables for the Visual C++ tools
(the standard location for this file is something like
<samp>C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat</samp>).
</p>
</li><li> Enter the <samp>bin</samp> directory where the created LIB and DLL files
are stored.
</li><li> Generate new import libraries with <code>lib.exe</code>:
<div class="example">
<pre class="example">lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib
</pre></div>
<p>Replace <code>foo-version</code> and <code>foo</code> with the respective library names.
</p>
</li></ol>
<a name="Cross-compilation-for-Windows-with-Linux"></a><a name="Cross-compilation-for-Windows-with-Linux-1"></a>
<h3 class="section">4.3 Cross compilation for Windows with Linux<span class="pull-right"><a class="anchor hidden-xs" href="#Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Cross-compilation-for-Windows-with-Linux-1" aria-hidden="true">TOC</a></span></h3>
<p>You must use the MinGW cross compilation tools available at
<a href="http://www.mingw.org/">http://www.mingw.org/</a>.
</p>
<p>Then configure FFmpeg with the following options:
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
</pre></div>
<p>(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
</p>
<p>Then you can easily test FFmpeg with <a href="http://www.winehq.com/">Wine</a>.
</p>
<a name="Compilation-under-Cygwin"></a>
<h3 class="section">4.4 Compilation under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Compilation-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Compilation-under-Cygwin" aria-hidden="true">TOC</a></span></h3>
<p>Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
</p>
<p>Install your Cygwin with all the &quot;Base&quot; packages, plus the
following &quot;Devel&quot; ones:
</p><div class="example">
<pre class="example">binutils, gcc4-core, make, git, mingw-runtime, texinfo
</pre></div>
<p>In order to run FATE you will also need the following &quot;Utils&quot; packages:
</p><div class="example">
<pre class="example">bc, diffutils
</pre></div>
<p>If you want to build FFmpeg with additional libraries, download Cygwin
&quot;Devel&quot; packages for Ogg and Vorbis from any Cygwin packages repository:
</p><div class="example">
<pre class="example">libogg-devel, libvorbis-devel
</pre></div>
<p>These library packages are only available from
<a href="http://sourceware.org/cygwinports/">Cygwin Ports</a>:
</p>
<div class="example">
<pre class="example">yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
</pre></div>
<p>The recommendation for x264 is to build it from source, as it evolves too
quickly for Cygwin Ports to be up to date.
</p>
<a name="Crosscompilation-for-Windows-under-Cygwin"></a>
<h3 class="section">4.5 Crosscompilation for Windows under Cygwin<span class="pull-right"><a class="anchor hidden-xs" href="#Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Crosscompilation-for-Windows-under-Cygwin" aria-hidden="true">TOC</a></span></h3>
<p>With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
</p>
<p>Just install your Cygwin as explained before, plus these additional
&quot;Devel&quot; packages:
</p><div class="example">
<pre class="example">gcc-mingw-core, mingw-runtime, mingw-zlib
</pre></div>
<p>and add some special flags to your configure invocation.
</p>
<p>For a static build run
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
</pre></div>
<p>and for a build with shared libraries
</p><div class="example">
<pre class="example">./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
</pre></div>
<a name="Plan-9"></a>
<h2 class="chapter">5 Plan 9<span class="pull-right"><a class="anchor hidden-xs" href="#Plan-9" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Plan-9" aria-hidden="true">TOC</a></span></h2>
<p>The native <a href="http://plan9.bell-labs.com/plan9/">Plan 9</a> compiler
does not implement all the C99 features needed by FFmpeg so the gcc
port must be used. Furthermore, a few items missing from the C
library and shell environment need to be fixed.
</p>
<ul>
<li> GNU awk, grep, make, and sed
<p>Working packages of these tools can be found at
<a href="http://code.google.com/p/ports2plan9/downloads/list">ports2plan9</a>.
They can be installed with <a href="http://9front.org/">9front&rsquo;s</a> <code>pkg</code>
utility by setting <code>pkgpath</code> to
<code>http://ports2plan9.googlecode.com/files/</code>.
</p>
</li><li> Missing/broken <code>head</code> and <code>printf</code> commands
<p>Replacements adequate for building FFmpeg can be found in the
<code>compat/plan9</code> directory. Place these somewhere they will be
found by the shell. These are not full implementations of the
commands and are <em>not</em> suitable for general use.
</p>
</li><li> Missing C99 <code>stdint.h</code> and <code>inttypes.h</code>
<p>Replacement headers are available from
<a href="http://code.google.com/p/plan9front/issues/detail?id=152">http://code.google.com/p/plan9front/issues/detail?id=152</a>.
</p>
</li><li> Missing or non-standard library functions
<p>Some functions in the C library are missing or incomplete. The
<code><a href="http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz">gcc-apelibs-1207</a></code> package from
<a href="http://code.google.com/p/ports2plan9/downloads/list">ports2plan9</a>
includes an updated C library, but installing the full package gives
unusable executables. Instead, keep the files from <code>gccbin.tgz</code>
under <code>/386/lib/gnu</code>. From the <code>libc.a</code> archive in the
<code>gcc-apelibs-1207</code> package, extract the following object files and
turn them into a library:
</p>
<ul>
<li> <code>strerror.o</code>
</li><li> <code>strtoll.o</code>
</li><li> <code>snprintf.o</code>
</li><li> <code>vsnprintf.o</code>
</li><li> <code>vfprintf.o</code>
</li><li> <code>_IO_getc.o</code>
</li><li> <code>_IO_putc.o</code>
</li></ul>
<p>Use the <code>--extra-libs</code> option of <code>configure</code> to inform the
build system of this library.
</p>
</li><li> FPU exceptions enabled by default
<p>Unlike most other systems, Plan 9 enables FPU exceptions by default.
These must be disabled before calling any FFmpeg functions. While the
included tools will do this automatically, other users of the
libraries must do it themselves.
</p>
</li></ul>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

509
Externals/ffmpeg/dev/include/libavdevice/avdevice.h vendored
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@ -1,509 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVDEVICE_AVDEVICE_H
#define AVDEVICE_AVDEVICE_H
#include "version.h"
/**
* @file
* @ingroup lavd
* Main libavdevice API header
*/
/**
* @defgroup lavd Special devices muxing/demuxing library
* @{
* Libavdevice is a complementary library to @ref libavf "libavformat". It
* provides various "special" platform-specific muxers and demuxers, e.g. for
* grabbing devices, audio capture and playback etc. As a consequence, the
* (de)muxers in libavdevice are of the AVFMT_NOFILE type (they use their own
* I/O functions). The filename passed to avformat_open_input() often does not
* refer to an actually existing file, but has some special device-specific
* meaning - e.g. for x11grab it is the display name.
*
* To use libavdevice, simply call avdevice_register_all() to register all
* compiled muxers and demuxers. They all use standard libavformat API.
* @}
*/
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/dict.h"
#include "libavformat/avformat.h"
/**
* Return the LIBAVDEVICE_VERSION_INT constant.
*/
unsigned avdevice_version(void);
/**
* Return the libavdevice build-time configuration.
*/
const char *avdevice_configuration(void);
/**
* Return the libavdevice license.
*/
const char *avdevice_license(void);
/**
* Initialize libavdevice and register all the input and output devices.
* @warning This function is not thread safe.
*/
void avdevice_register_all(void);
/**
* Audio input devices iterator.
*
* If d is NULL, returns the first registered input audio/video device,
* if d is non-NULL, returns the next registered input audio/video device after d
* or NULL if d is the last one.
*/
AVInputFormat *av_input_audio_device_next(AVInputFormat *d);
/**
* Video input devices iterator.
*
* If d is NULL, returns the first registered input audio/video device,
* if d is non-NULL, returns the next registered input audio/video device after d
* or NULL if d is the last one.
*/
AVInputFormat *av_input_video_device_next(AVInputFormat *d);
/**
* Audio output devices iterator.
*
* If d is NULL, returns the first registered output audio/video device,
* if d is non-NULL, returns the next registered output audio/video device after d
* or NULL if d is the last one.
*/
AVOutputFormat *av_output_audio_device_next(AVOutputFormat *d);
/**
* Video output devices iterator.
*
* If d is NULL, returns the first registered output audio/video device,
* if d is non-NULL, returns the next registered output audio/video device after d
* or NULL if d is the last one.
*/
AVOutputFormat *av_output_video_device_next(AVOutputFormat *d);
typedef struct AVDeviceRect {
int x; /**< x coordinate of top left corner */
int y; /**< y coordinate of top left corner */
int width; /**< width */
int height; /**< height */
} AVDeviceRect;
/**
* Message types used by avdevice_app_to_dev_control_message().
*/
enum AVAppToDevMessageType {
/**
* Dummy message.
*/
AV_APP_TO_DEV_NONE = MKBETAG('N','O','N','E'),
/**
* Window size change message.
*
* Message is sent to the device every time the application changes the size
* of the window device renders to.
* Message should also be sent right after window is created.
*
* data: AVDeviceRect: new window size.
*/
AV_APP_TO_DEV_WINDOW_SIZE = MKBETAG('G','E','O','M'),
/**
* Repaint request message.
*
* Message is sent to the device when window has to be repainted.
*
* data: AVDeviceRect: area required to be repainted.
* NULL: whole area is required to be repainted.
*/
AV_APP_TO_DEV_WINDOW_REPAINT = MKBETAG('R','E','P','A'),
/**
* Request pause/play.
*
* Application requests pause/unpause playback.
* Mostly usable with devices that have internal buffer.
* By default devices are not paused.
*
* data: NULL
*/
AV_APP_TO_DEV_PAUSE = MKBETAG('P', 'A', 'U', ' '),
AV_APP_TO_DEV_PLAY = MKBETAG('P', 'L', 'A', 'Y'),
AV_APP_TO_DEV_TOGGLE_PAUSE = MKBETAG('P', 'A', 'U', 'T'),
/**
* Volume control message.
*
* Set volume level. It may be device-dependent if volume
* is changed per stream or system wide. Per stream volume
* change is expected when possible.
*
* data: double: new volume with range of 0.0 - 1.0.
*/
AV_APP_TO_DEV_SET_VOLUME = MKBETAG('S', 'V', 'O', 'L'),
/**
* Mute control messages.
*
* Change mute state. It may be device-dependent if mute status
* is changed per stream or system wide. Per stream mute status
* change is expected when possible.
*
* data: NULL.
*/
AV_APP_TO_DEV_MUTE = MKBETAG(' ', 'M', 'U', 'T'),
AV_APP_TO_DEV_UNMUTE = MKBETAG('U', 'M', 'U', 'T'),
AV_APP_TO_DEV_TOGGLE_MUTE = MKBETAG('T', 'M', 'U', 'T'),
/**
* Get volume/mute messages.
*
* Force the device to send AV_DEV_TO_APP_VOLUME_LEVEL_CHANGED or
* AV_DEV_TO_APP_MUTE_STATE_CHANGED command respectively.
*
* data: NULL.
*/
AV_APP_TO_DEV_GET_VOLUME = MKBETAG('G', 'V', 'O', 'L'),
AV_APP_TO_DEV_GET_MUTE = MKBETAG('G', 'M', 'U', 'T'),
};
/**
* Message types used by avdevice_dev_to_app_control_message().
*/
enum AVDevToAppMessageType {
/**
* Dummy message.
*/
AV_DEV_TO_APP_NONE = MKBETAG('N','O','N','E'),
/**
* Create window buffer message.
*
* Device requests to create a window buffer. Exact meaning is device-
* and application-dependent. Message is sent before rendering first
* frame and all one-shot initializations should be done here.
* Application is allowed to ignore preferred window buffer size.
*
* @note: Application is obligated to inform about window buffer size
* with AV_APP_TO_DEV_WINDOW_SIZE message.
*
* data: AVDeviceRect: preferred size of the window buffer.
* NULL: no preferred size of the window buffer.
*/
AV_DEV_TO_APP_CREATE_WINDOW_BUFFER = MKBETAG('B','C','R','E'),
/**
* Prepare window buffer message.
*
* Device requests to prepare a window buffer for rendering.
* Exact meaning is device- and application-dependent.
* Message is sent before rendering of each frame.
*
* data: NULL.
*/
AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER = MKBETAG('B','P','R','E'),
/**
* Display window buffer message.
*
* Device requests to display a window buffer.
* Message is sent when new frame is ready to be displayed.
* Usually buffers need to be swapped in handler of this message.
*
* data: NULL.
*/
AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER = MKBETAG('B','D','I','S'),
/**
* Destroy window buffer message.
*
* Device requests to destroy a window buffer.
* Message is sent when device is about to be destroyed and window
* buffer is not required anymore.
*
* data: NULL.
*/
AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER = MKBETAG('B','D','E','S'),
/**
* Buffer fullness status messages.
*
* Device signals buffer overflow/underflow.
*
* data: NULL.
*/
AV_DEV_TO_APP_BUFFER_OVERFLOW = MKBETAG('B','O','F','L'),
AV_DEV_TO_APP_BUFFER_UNDERFLOW = MKBETAG('B','U','F','L'),
/**
* Buffer readable/writable.
*
* Device informs that buffer is readable/writable.
* When possible, device informs how many bytes can be read/write.
*
* @warning Device may not inform when number of bytes than can be read/write changes.
*
* data: int64_t: amount of bytes available to read/write.
* NULL: amount of bytes available to read/write is not known.
*/
AV_DEV_TO_APP_BUFFER_READABLE = MKBETAG('B','R','D',' '),
AV_DEV_TO_APP_BUFFER_WRITABLE = MKBETAG('B','W','R',' '),
/**
* Mute state change message.
*
* Device informs that mute state has changed.
*
* data: int: 0 for not muted state, non-zero for muted state.
*/
AV_DEV_TO_APP_MUTE_STATE_CHANGED = MKBETAG('C','M','U','T'),
/**
* Volume level change message.
*
* Device informs that volume level has changed.
*
* data: double: new volume with range of 0.0 - 1.0.
*/
AV_DEV_TO_APP_VOLUME_LEVEL_CHANGED = MKBETAG('C','V','O','L'),
};
/**
* Send control message from application to device.
*
* @param s device context.
* @param type message type.
* @param data message data. Exact type depends on message type.
* @param data_size size of message data.
* @return >= 0 on success, negative on error.
* AVERROR(ENOSYS) when device doesn't implement handler of the message.
*/
int avdevice_app_to_dev_control_message(struct AVFormatContext *s,
enum AVAppToDevMessageType type,
void *data, size_t data_size);
/**
* Send control message from device to application.
*
* @param s device context.
* @param type message type.
* @param data message data. Can be NULL.
* @param data_size size of message data.
* @return >= 0 on success, negative on error.
* AVERROR(ENOSYS) when application doesn't implement handler of the message.
*/
int avdevice_dev_to_app_control_message(struct AVFormatContext *s,
enum AVDevToAppMessageType type,
void *data, size_t data_size);
/**
* Following API allows user to probe device capabilities (supported codecs,
* pixel formats, sample formats, resolutions, channel counts, etc).
* It is build on top op AVOption API.
* Queried capabilities allows to set up converters of video or audio
* parameters that fit to the device.
*
* List of capabilities that can be queried:
* - Capabilities valid for both audio and video devices:
* - codec: supported audio/video codecs.
* type: AV_OPT_TYPE_INT (AVCodecID value)
* - Capabilities valid for audio devices:
* - sample_format: supported sample formats.
* type: AV_OPT_TYPE_INT (AVSampleFormat value)
* - sample_rate: supported sample rates.
* type: AV_OPT_TYPE_INT
* - channels: supported number of channels.
* type: AV_OPT_TYPE_INT
* - channel_layout: supported channel layouts.
* type: AV_OPT_TYPE_INT64
* - Capabilities valid for video devices:
* - pixel_format: supported pixel formats.
* type: AV_OPT_TYPE_INT (AVPixelFormat value)
* - window_size: supported window sizes (describes size of the window size presented to the user).
* type: AV_OPT_TYPE_IMAGE_SIZE
* - frame_size: supported frame sizes (describes size of provided video frames).
* type: AV_OPT_TYPE_IMAGE_SIZE
* - fps: supported fps values
* type: AV_OPT_TYPE_RATIONAL
*
* Value of the capability may be set by user using av_opt_set() function
* and AVDeviceCapabilitiesQuery object. Following queries will
* limit results to the values matching already set capabilities.
* For example, setting a codec may impact number of formats or fps values
* returned during next query. Setting invalid value may limit results to zero.
*
* Example of the usage basing on opengl output device:
*
* @code
* AVFormatContext *oc = NULL;
* AVDeviceCapabilitiesQuery *caps = NULL;
* AVOptionRanges *ranges;
* int ret;
*
* if ((ret = avformat_alloc_output_context2(&oc, NULL, "opengl", NULL)) < 0)
* goto fail;
* if (avdevice_capabilities_create(&caps, oc, NULL) < 0)
* goto fail;
*
* //query codecs
* if (av_opt_query_ranges(&ranges, caps, "codec", AV_OPT_MULTI_COMPONENT_RANGE)) < 0)
* goto fail;
* //pick codec here and set it
* av_opt_set(caps, "codec", AV_CODEC_ID_RAWVIDEO, 0);
*
* //query format
* if (av_opt_query_ranges(&ranges, caps, "pixel_format", AV_OPT_MULTI_COMPONENT_RANGE)) < 0)
* goto fail;
* //pick format here and set it
* av_opt_set(caps, "pixel_format", AV_PIX_FMT_YUV420P, 0);
*
* //query and set more capabilities
*
* fail:
* //clean up code
* avdevice_capabilities_free(&query, oc);
* avformat_free_context(oc);
* @endcode
*/
/**
* Structure describes device capabilities.
*
* It is used by devices in conjunction with av_device_capabilities AVOption table
* to implement capabilities probing API based on AVOption API. Should not be used directly.
*/
typedef struct AVDeviceCapabilitiesQuery {
const AVClass *av_class;
AVFormatContext *device_context;
enum AVCodecID codec;
enum AVSampleFormat sample_format;
enum AVPixelFormat pixel_format;
int sample_rate;
int channels;
int64_t channel_layout;
int window_width;
int window_height;
int frame_width;
int frame_height;
AVRational fps;
} AVDeviceCapabilitiesQuery;
/**
* AVOption table used by devices to implement device capabilities API. Should not be used by a user.
*/
extern const AVOption av_device_capabilities[];
/**
* Initialize capabilities probing API based on AVOption API.
*
* avdevice_capabilities_free() must be called when query capabilities API is
* not used anymore.
*
* @param[out] caps Device capabilities data. Pointer to a NULL pointer must be passed.
* @param s Context of the device.
* @param device_options An AVDictionary filled with device-private options.
* On return this parameter will be destroyed and replaced with a dict
* containing options that were not found. May be NULL.
* The same options must be passed later to avformat_write_header() for output
* devices or avformat_open_input() for input devices, or at any other place
* that affects device-private options.
*
* @return >= 0 on success, negative otherwise.
*/
int avdevice_capabilities_create(AVDeviceCapabilitiesQuery **caps, AVFormatContext *s,
AVDictionary **device_options);
/**
* Free resources created by avdevice_capabilities_create()
*
* @param caps Device capabilities data to be freed.
* @param s Context of the device.
*/
void avdevice_capabilities_free(AVDeviceCapabilitiesQuery **caps, AVFormatContext *s);
/**
* Structure describes basic parameters of the device.
*/
typedef struct AVDeviceInfo {
char *device_name; /**< device name, format depends on device */
char *device_description; /**< human friendly name */
} AVDeviceInfo;
/**
* List of devices.
*/
typedef struct AVDeviceInfoList {
AVDeviceInfo **devices; /**< list of autodetected devices */
int nb_devices; /**< number of autodetected devices */
int default_device; /**< index of default device or -1 if no default */
} AVDeviceInfoList;
/**
* List devices.
*
* Returns available device names and their parameters.
*
* @note: Some devices may accept system-dependent device names that cannot be
* autodetected. The list returned by this function cannot be assumed to
* be always completed.
*
* @param s device context.
* @param[out] device_list list of autodetected devices.
* @return count of autodetected devices, negative on error.
*/
int avdevice_list_devices(struct AVFormatContext *s, AVDeviceInfoList **device_list);
/**
* Convenient function to free result of avdevice_list_devices().
*
* @param devices device list to be freed.
*/
void avdevice_free_list_devices(AVDeviceInfoList **device_list);
/**
* List devices.
*
* Returns available device names and their parameters.
* These are convinient wrappers for avdevice_list_devices().
* Device context is allocated and deallocated internally.
*
* @param device device format. May be NULL if device name is set.
* @param device_name device name. May be NULL if device format is set.
* @param device_options An AVDictionary filled with device-private options. May be NULL.
* The same options must be passed later to avformat_write_header() for output
* devices or avformat_open_input() for input devices, or at any other place
* that affects device-private options.
* @param[out] device_list list of autodetected devices
* @return count of autodetected devices, negative on error.
* @note device argument takes precedence over device_name when both are set.
*/
int avdevice_list_input_sources(struct AVInputFormat *device, const char *device_name,
AVDictionary *device_options, AVDeviceInfoList **device_list);
int avdevice_list_output_sinks(struct AVOutputFormat *device, const char *device_name,
AVDictionary *device_options, AVDeviceInfoList **device_list);
#endif /* AVDEVICE_AVDEVICE_H */

50
Externals/ffmpeg/dev/include/libavdevice/version.h vendored
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@ -1,50 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVDEVICE_VERSION_H
#define AVDEVICE_VERSION_H
/**
* @file
* @ingroup lavd
* Libavdevice version macros
*/
#include "libavutil/version.h"
#define LIBAVDEVICE_VERSION_MAJOR 56
#define LIBAVDEVICE_VERSION_MINOR 4
#define LIBAVDEVICE_VERSION_MICRO 100
#define LIBAVDEVICE_VERSION_INT AV_VERSION_INT(LIBAVDEVICE_VERSION_MAJOR, \
LIBAVDEVICE_VERSION_MINOR, \
LIBAVDEVICE_VERSION_MICRO)
#define LIBAVDEVICE_VERSION AV_VERSION(LIBAVDEVICE_VERSION_MAJOR, \
LIBAVDEVICE_VERSION_MINOR, \
LIBAVDEVICE_VERSION_MICRO)
#define LIBAVDEVICE_BUILD LIBAVDEVICE_VERSION_INT
#define LIBAVDEVICE_IDENT "Lavd" AV_STRINGIFY(LIBAVDEVICE_VERSION)
/**
* FF_API_* defines may be placed below to indicate public API that will be
* dropped at a future version bump. The defines themselves are not part of
* the public API and may change, break or disappear at any time.
*/
#endif /* AVDEVICE_VERSION_H */

91
Externals/ffmpeg/dev/include/libavfilter/asrc_abuffer.h vendored
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@ -1,91 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_ASRC_ABUFFER_H
#define AVFILTER_ASRC_ABUFFER_H
#include "avfilter.h"
/**
* @file
* memory buffer source for audio
*
* @deprecated use buffersrc.h instead.
*/
/**
* Queue an audio buffer to the audio buffer source.
*
* @param abuffersrc audio source buffer context
* @param data pointers to the samples planes
* @param linesize linesizes of each audio buffer plane
* @param nb_samples number of samples per channel
* @param sample_fmt sample format of the audio data
* @param ch_layout channel layout of the audio data
* @param planar flag to indicate if audio data is planar or packed
* @param pts presentation timestamp of the audio buffer
* @param flags unused
*
* @deprecated use av_buffersrc_add_ref() instead.
*/
attribute_deprecated
int av_asrc_buffer_add_samples(AVFilterContext *abuffersrc,
uint8_t *data[8], int linesize[8],
int nb_samples, int sample_rate,
int sample_fmt, int64_t ch_layout, int planar,
int64_t pts, int av_unused flags);
/**
* Queue an audio buffer to the audio buffer source.
*
* This is similar to av_asrc_buffer_add_samples(), but the samples
* are stored in a buffer with known size.
*
* @param abuffersrc audio source buffer context
* @param buf pointer to the samples data, packed is assumed
* @param size the size in bytes of the buffer, it must contain an
* integer number of samples
* @param sample_fmt sample format of the audio data
* @param ch_layout channel layout of the audio data
* @param pts presentation timestamp of the audio buffer
* @param flags unused
*
* @deprecated use av_buffersrc_add_ref() instead.
*/
attribute_deprecated
int av_asrc_buffer_add_buffer(AVFilterContext *abuffersrc,
uint8_t *buf, int buf_size,
int sample_rate,
int sample_fmt, int64_t ch_layout, int planar,
int64_t pts, int av_unused flags);
/**
* Queue an audio buffer to the audio buffer source.
*
* @param abuffersrc audio source buffer context
* @param samplesref buffer ref to queue
* @param flags unused
*
* @deprecated use av_buffersrc_add_ref() instead.
*/
attribute_deprecated
int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *abuffersrc,
AVFilterBufferRef *samplesref,
int av_unused flags);
#endif /* AVFILTER_ASRC_ABUFFER_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_AVCODEC_H
#define AVFILTER_AVCODEC_H
/**
* @file
* libavcodec/libavfilter gluing utilities
*
* This should be included in an application ONLY if the installed
* libavfilter has been compiled with libavcodec support, otherwise
* symbols defined below will not be available.
*/
#include "avfilter.h"
#if FF_API_AVFILTERBUFFER
/**
* Create and return a picref reference from the data and properties
* contained in frame.
*
* @param perms permissions to assign to the new buffer reference
* @deprecated avfilter APIs work natively with AVFrame instead.
*/
attribute_deprecated
AVFilterBufferRef *avfilter_get_video_buffer_ref_from_frame(const AVFrame *frame, int perms);
/**
* Create and return a picref reference from the data and properties
* contained in frame.
*
* @param perms permissions to assign to the new buffer reference
* @deprecated avfilter APIs work natively with AVFrame instead.
*/
attribute_deprecated
AVFilterBufferRef *avfilter_get_audio_buffer_ref_from_frame(const AVFrame *frame,
int perms);
/**
* Create and return a buffer reference from the data and properties
* contained in frame.
*
* @param perms permissions to assign to the new buffer reference
* @deprecated avfilter APIs work natively with AVFrame instead.
*/
attribute_deprecated
AVFilterBufferRef *avfilter_get_buffer_ref_from_frame(enum AVMediaType type,
const AVFrame *frame,
int perms);
#endif
#endif /* AVFILTER_AVCODEC_H */

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/*
* Filter graphs
* copyright (c) 2007 Bobby Bingham
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_AVFILTERGRAPH_H
#define AVFILTER_AVFILTERGRAPH_H
#include "avfilter.h"
#include "libavutil/log.h"
#endif /* AVFILTER_AVFILTERGRAPH_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_BUFFERSINK_H
#define AVFILTER_BUFFERSINK_H
/**
* @file
* @ingroup lavfi_buffersink
* memory buffer sink API for audio and video
*/
#include "avfilter.h"
/**
* @defgroup lavfi_buffersink Buffer sink API
* @ingroup lavfi
* @{
*/
#if FF_API_AVFILTERBUFFER
/**
* Get an audio/video buffer data from buffer_sink and put it in bufref.
*
* This function works with both audio and video buffer sinks.
*
* @param buffer_sink pointer to a buffersink or abuffersink context
* @param flags a combination of AV_BUFFERSINK_FLAG_* flags
* @return >= 0 in case of success, a negative AVERROR code in case of
* failure
*/
attribute_deprecated
int av_buffersink_get_buffer_ref(AVFilterContext *buffer_sink,
AVFilterBufferRef **bufref, int flags);
/**
* Get the number of immediately available frames.
*/
attribute_deprecated
int av_buffersink_poll_frame(AVFilterContext *ctx);
/**
* Get a buffer with filtered data from sink and put it in buf.
*
* @param ctx pointer to a context of a buffersink or abuffersink AVFilter.
* @param buf pointer to the buffer will be written here if buf is non-NULL. buf
* must be freed by the caller using avfilter_unref_buffer().
* Buf may also be NULL to query whether a buffer is ready to be
* output.
*
* @return >= 0 in case of success, a negative AVERROR code in case of
* failure.
*/
attribute_deprecated
int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf);
/**
* Same as av_buffersink_read, but with the ability to specify the number of
* samples read. This function is less efficient than av_buffersink_read(),
* because it copies the data around.
*
* @param ctx pointer to a context of the abuffersink AVFilter.
* @param buf pointer to the buffer will be written here if buf is non-NULL. buf
* must be freed by the caller using avfilter_unref_buffer(). buf
* will contain exactly nb_samples audio samples, except at the end
* of stream, when it can contain less than nb_samples.
* Buf may also be NULL to query whether a buffer is ready to be
* output.
*
* @warning do not mix this function with av_buffersink_read(). Use only one or
* the other with a single sink, not both.
*/
attribute_deprecated
int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **buf,
int nb_samples);
#endif
/**
* Get a frame with filtered data from sink and put it in frame.
*
* @param ctx pointer to a buffersink or abuffersink filter context.
* @param frame pointer to an allocated frame that will be filled with data.
* The data must be freed using av_frame_unref() / av_frame_free()
* @param flags a combination of AV_BUFFERSINK_FLAG_* flags
*
* @return >= 0 in for success, a negative AVERROR code for failure.
*/
int av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flags);
/**
* Tell av_buffersink_get_buffer_ref() to read video/samples buffer
* reference, but not remove it from the buffer. This is useful if you
* need only to read a video/samples buffer, without to fetch it.
*/
#define AV_BUFFERSINK_FLAG_PEEK 1
/**
* Tell av_buffersink_get_buffer_ref() not to request a frame from its input.
* If a frame is already buffered, it is read (and removed from the buffer),
* but if no frame is present, return AVERROR(EAGAIN).
*/
#define AV_BUFFERSINK_FLAG_NO_REQUEST 2
/**
* Struct to use for initializing a buffersink context.
*/
typedef struct {
const enum AVPixelFormat *pixel_fmts; ///< list of allowed pixel formats, terminated by AV_PIX_FMT_NONE
} AVBufferSinkParams;
/**
* Create an AVBufferSinkParams structure.
*
* Must be freed with av_free().
*/
AVBufferSinkParams *av_buffersink_params_alloc(void);
/**
* Struct to use for initializing an abuffersink context.
*/
typedef struct {
const enum AVSampleFormat *sample_fmts; ///< list of allowed sample formats, terminated by AV_SAMPLE_FMT_NONE
const int64_t *channel_layouts; ///< list of allowed channel layouts, terminated by -1
const int *channel_counts; ///< list of allowed channel counts, terminated by -1
int all_channel_counts; ///< if not 0, accept any channel count or layout
int *sample_rates; ///< list of allowed sample rates, terminated by -1
} AVABufferSinkParams;
/**
* Create an AVABufferSinkParams structure.
*
* Must be freed with av_free().
*/
AVABufferSinkParams *av_abuffersink_params_alloc(void);
/**
* Set the frame size for an audio buffer sink.
*
* All calls to av_buffersink_get_buffer_ref will return a buffer with
* exactly the specified number of samples, or AVERROR(EAGAIN) if there is
* not enough. The last buffer at EOF will be padded with 0.
*/
void av_buffersink_set_frame_size(AVFilterContext *ctx, unsigned frame_size);
/**
* Get the frame rate of the input.
*/
AVRational av_buffersink_get_frame_rate(AVFilterContext *ctx);
/**
* Get a frame with filtered data from sink and put it in frame.
*
* @param ctx pointer to a context of a buffersink or abuffersink AVFilter.
* @param frame pointer to an allocated frame that will be filled with data.
* The data must be freed using av_frame_unref() / av_frame_free()
*
* @return
* - >= 0 if a frame was successfully returned.
* - AVERROR(EAGAIN) if no frames are available at this point; more
* input frames must be added to the filtergraph to get more output.
* - AVERROR_EOF if there will be no more output frames on this sink.
* - A different negative AVERROR code in other failure cases.
*/
int av_buffersink_get_frame(AVFilterContext *ctx, AVFrame *frame);
/**
* Same as av_buffersink_get_frame(), but with the ability to specify the number
* of samples read. This function is less efficient than
* av_buffersink_get_frame(), because it copies the data around.
*
* @param ctx pointer to a context of the abuffersink AVFilter.
* @param frame pointer to an allocated frame that will be filled with data.
* The data must be freed using av_frame_unref() / av_frame_free()
* frame will contain exactly nb_samples audio samples, except at
* the end of stream, when it can contain less than nb_samples.
*
* @return The return codes have the same meaning as for
* av_buffersink_get_samples().
*
* @warning do not mix this function with av_buffersink_get_frame(). Use only one or
* the other with a single sink, not both.
*/
int av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples);
/**
* @}
*/
#endif /* AVFILTER_BUFFERSINK_H */

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/*
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_BUFFERSRC_H
#define AVFILTER_BUFFERSRC_H
/**
* @file
* @ingroup lavfi_buffersrc
* Memory buffer source API.
*/
#include "libavcodec/avcodec.h"
#include "avfilter.h"
/**
* @defgroup lavfi_buffersrc Buffer source API
* @ingroup lavfi
* @{
*/
enum {
/**
* Do not check for format changes.
*/
AV_BUFFERSRC_FLAG_NO_CHECK_FORMAT = 1,
#if FF_API_AVFILTERBUFFER
/**
* Ignored
*/
AV_BUFFERSRC_FLAG_NO_COPY = 2,
#endif
/**
* Immediately push the frame to the output.
*/
AV_BUFFERSRC_FLAG_PUSH = 4,
/**
* Keep a reference to the frame.
* If the frame if reference-counted, create a new reference; otherwise
* copy the frame data.
*/
AV_BUFFERSRC_FLAG_KEEP_REF = 8,
};
/**
* Add buffer data in picref to buffer_src.
*
* @param buffer_src pointer to a buffer source context
* @param picref a buffer reference, or NULL to mark EOF
* @param flags a combination of AV_BUFFERSRC_FLAG_*
* @return >= 0 in case of success, a negative AVERROR code
* in case of failure
*/
int av_buffersrc_add_ref(AVFilterContext *buffer_src,
AVFilterBufferRef *picref, int flags);
/**
* Get the number of failed requests.
*
* A failed request is when the request_frame method is called while no
* frame is present in the buffer.
* The number is reset when a frame is added.
*/
unsigned av_buffersrc_get_nb_failed_requests(AVFilterContext *buffer_src);
#if FF_API_AVFILTERBUFFER
/**
* Add a buffer to a filtergraph.
*
* @param ctx an instance of the buffersrc filter
* @param buf buffer containing frame data to be passed down the filtergraph.
* This function will take ownership of buf, the user must not free it.
* A NULL buf signals EOF -- i.e. no more frames will be sent to this filter.
*
* @deprecated use av_buffersrc_write_frame() or av_buffersrc_add_frame()
*/
attribute_deprecated
int av_buffersrc_buffer(AVFilterContext *ctx, AVFilterBufferRef *buf);
#endif
/**
* Add a frame to the buffer source.
*
* @param ctx an instance of the buffersrc filter
* @param frame frame to be added. If the frame is reference counted, this
* function will make a new reference to it. Otherwise the frame data will be
* copied.
*
* @return 0 on success, a negative AVERROR on error
*
* This function is equivalent to av_buffersrc_add_frame_flags() with the
* AV_BUFFERSRC_FLAG_KEEP_REF flag.
*/
int av_buffersrc_write_frame(AVFilterContext *ctx, const AVFrame *frame);
/**
* Add a frame to the buffer source.
*
* @param ctx an instance of the buffersrc filter
* @param frame frame to be added. If the frame is reference counted, this
* function will take ownership of the reference(s) and reset the frame.
* Otherwise the frame data will be copied. If this function returns an error,
* the input frame is not touched.
*
* @return 0 on success, a negative AVERROR on error.
*
* @note the difference between this function and av_buffersrc_write_frame() is
* that av_buffersrc_write_frame() creates a new reference to the input frame,
* while this function takes ownership of the reference passed to it.
*
* This function is equivalent to av_buffersrc_add_frame_flags() without the
* AV_BUFFERSRC_FLAG_KEEP_REF flag.
*/
int av_buffersrc_add_frame(AVFilterContext *ctx, AVFrame *frame);
/**
* Add a frame to the buffer source.
*
* By default, if the frame is reference-counted, this function will take
* ownership of the reference(s) and reset the frame. This can be controlled
* using the flags.
*
* If this function returns an error, the input frame is not touched.
*
* @param buffer_src pointer to a buffer source context
* @param frame a frame, or NULL to mark EOF
* @param flags a combination of AV_BUFFERSRC_FLAG_*
* @return >= 0 in case of success, a negative AVERROR code
* in case of failure
*/
int av_buffersrc_add_frame_flags(AVFilterContext *buffer_src,
AVFrame *frame, int flags);
/**
* @}
*/
#endif /* AVFILTER_BUFFERSRC_H */

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/*
* Version macros.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_VERSION_H
#define AVFILTER_VERSION_H
/**
* @file
* @ingroup lavfi
* Libavfilter version macros
*/
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 5
#define LIBAVFILTER_VERSION_MINOR 7
#define LIBAVFILTER_VERSION_MICRO 101
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
LIBAVFILTER_VERSION_MICRO)
#define LIBAVFILTER_VERSION AV_VERSION(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
LIBAVFILTER_VERSION_MICRO)
#define LIBAVFILTER_BUILD LIBAVFILTER_VERSION_INT
#define LIBAVFILTER_IDENT "Lavfi" AV_STRINGIFY(LIBAVFILTER_VERSION)
/**
* FF_API_* defines may be placed below to indicate public API that will be
* dropped at a future version bump. The defines themselves are not part of
* the public API and may change, break or disappear at any time.
*/
#ifndef FF_API_AVFILTERPAD_PUBLIC
#define FF_API_AVFILTERPAD_PUBLIC (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_FOO_COUNT
#define FF_API_FOO_COUNT (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_AVFILTERBUFFER
#define FF_API_AVFILTERBUFFER (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_OLD_FILTER_OPTS
#define FF_API_OLD_FILTER_OPTS (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_AVFILTER_OPEN
#define FF_API_AVFILTER_OPEN (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_AVFILTER_INIT_FILTER
#define FF_API_AVFILTER_INIT_FILTER (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_OLD_FILTER_REGISTER
#define FF_API_OLD_FILTER_REGISTER (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#ifndef FF_API_OLD_GRAPH_PARSE
#define FF_API_OLD_GRAPH_PARSE (LIBAVFILTER_VERSION_MAJOR < 5)
#endif
#ifndef FF_API_NOCONST_GET_NAME
#define FF_API_NOCONST_GET_NAME (LIBAVFILTER_VERSION_MAJOR < 6)
#endif
#endif /* AVFILTER_VERSION_H */

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/*
* Copyright (C) 2001-2003 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef POSTPROC_POSTPROCESS_H
#define POSTPROC_POSTPROCESS_H
/**
* @file
* @ingroup lpp
* external API header
*/
/**
* @defgroup lpp Libpostproc
* @{
*/
#include "libpostproc/version.h"
/**
* Return the LIBPOSTPROC_VERSION_INT constant.
*/
unsigned postproc_version(void);
/**
* Return the libpostproc build-time configuration.
*/
const char *postproc_configuration(void);
/**
* Return the libpostproc license.
*/
const char *postproc_license(void);
#define PP_QUALITY_MAX 6
#define QP_STORE_T int8_t
#include <inttypes.h>
typedef void pp_context;
typedef void pp_mode;
#if LIBPOSTPROC_VERSION_INT < (52<<16)
typedef pp_context pp_context_t;
typedef pp_mode pp_mode_t;
extern const char *const pp_help; ///< a simple help text
#else
extern const char pp_help[]; ///< a simple help text
#endif
void pp_postprocess(const uint8_t * src[3], const int srcStride[3],
uint8_t * dst[3], const int dstStride[3],
int horizontalSize, int verticalSize,
const QP_STORE_T *QP_store, int QP_stride,
pp_mode *mode, pp_context *ppContext, int pict_type);
/**
* Return a pp_mode or NULL if an error occurred.
*
* @param name the string after "-pp" on the command line
* @param quality a number from 0 to PP_QUALITY_MAX
*/
pp_mode *pp_get_mode_by_name_and_quality(const char *name, int quality);
void pp_free_mode(pp_mode *mode);
pp_context *pp_get_context(int width, int height, int flags);
void pp_free_context(pp_context *ppContext);
#define PP_CPU_CAPS_MMX 0x80000000
#define PP_CPU_CAPS_MMX2 0x20000000
#define PP_CPU_CAPS_3DNOW 0x40000000
#define PP_CPU_CAPS_ALTIVEC 0x10000000
#define PP_CPU_CAPS_AUTO 0x00080000
#define PP_FORMAT 0x00000008
#define PP_FORMAT_420 (0x00000011|PP_FORMAT)
#define PP_FORMAT_422 (0x00000001|PP_FORMAT)
#define PP_FORMAT_411 (0x00000002|PP_FORMAT)
#define PP_FORMAT_444 (0x00000000|PP_FORMAT)
#define PP_FORMAT_440 (0x00000010|PP_FORMAT)
#define PP_PICT_TYPE_QP2 0x00000010 ///< MPEG2 style QScale
/**
* @}
*/
#endif /* POSTPROC_POSTPROCESS_H */

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/*
* Version macros.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef POSTPROC_POSTPROCESS_VERSION_H
#define POSTPROC_POSTPROCESS_VERSION_H
/**
* @file
* Libpostproc version macros
*/
#include "libavutil/avutil.h"
#define LIBPOSTPROC_VERSION_MAJOR 53
#define LIBPOSTPROC_VERSION_MINOR 3
#define LIBPOSTPROC_VERSION_MICRO 100
#define LIBPOSTPROC_VERSION_INT AV_VERSION_INT(LIBPOSTPROC_VERSION_MAJOR, \
LIBPOSTPROC_VERSION_MINOR, \
LIBPOSTPROC_VERSION_MICRO)
#define LIBPOSTPROC_VERSION AV_VERSION(LIBPOSTPROC_VERSION_MAJOR, \
LIBPOSTPROC_VERSION_MINOR, \
LIBPOSTPROC_VERSION_MICRO)
#define LIBPOSTPROC_BUILD LIBPOSTPROC_VERSION_INT
#define LIBPOSTPROC_IDENT "postproc" AV_STRINGIFY(LIBPOSTPROC_VERSION)
#endif /* POSTPROC_POSTPROCESS_VERSION_H */

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/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_SWRESAMPLE_H
#define SWRESAMPLE_SWRESAMPLE_H
/**
* @file
* @ingroup lswr
* libswresample public header
*/
/**
* @defgroup lswr Libswresample
* @{
*
* Libswresample (lswr) is a library that handles audio resampling, sample
* format conversion and mixing.
*
* Interaction with lswr is done through SwrContext, which is
* allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
* must be set with the @ref avoptions API.
*
* The first thing you will need to do in order to use lswr is to allocate
* SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
* are using the former, you must set options through the @ref avoptions API.
* The latter function provides the same feature, but it allows you to set some
* common options in the same statement.
*
* For example the following code will setup conversion from planar float sample
* format to interleaved signed 16-bit integer, downsampling from 48kHz to
* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
* matrix). This is using the swr_alloc() function.
* @code
* SwrContext *swr = swr_alloc();
* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
* av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
* @endcode
*
* The same job can be done using swr_alloc_set_opts() as well:
* @code
* SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
* AV_CH_LAYOUT_STEREO, // out_ch_layout
* AV_SAMPLE_FMT_S16, // out_sample_fmt
* 44100, // out_sample_rate
* AV_CH_LAYOUT_5POINT1, // in_ch_layout
* AV_SAMPLE_FMT_FLTP, // in_sample_fmt
* 48000, // in_sample_rate
* 0, // log_offset
* NULL); // log_ctx
* @endcode
*
* Once all values have been set, it must be initialized with swr_init(). If
* you need to change the conversion parameters, you can change the parameters
* using @ref AVOptions, as described above in the first example; or by using
* swr_alloc_set_opts(), but with the first argument the allocated context.
* You must then call swr_init() again.
*
* The conversion itself is done by repeatedly calling swr_convert().
* Note that the samples may get buffered in swr if you provide insufficient
* output space or if sample rate conversion is done, which requires "future"
* samples. Samples that do not require future input can be retrieved at any
* time by using swr_convert() (in_count can be set to 0).
* At the end of conversion the resampling buffer can be flushed by calling
* swr_convert() with NULL in and 0 in_count.
*
* The samples used in the conversion process can be managed with the libavutil
* @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
* function used in the following example.
*
* The delay between input and output, can at any time be found by using
* swr_get_delay().
*
* The following code demonstrates the conversion loop assuming the parameters
* from above and caller-defined functions get_input() and handle_output():
* @code
* uint8_t **input;
* int in_samples;
*
* while (get_input(&input, &in_samples)) {
* uint8_t *output;
* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
* in_samples, 44100, 48000, AV_ROUND_UP);
* av_samples_alloc(&output, NULL, 2, out_samples,
* AV_SAMPLE_FMT_S16, 0);
* out_samples = swr_convert(swr, &output, out_samples,
* input, in_samples);
* handle_output(output, out_samples);
* av_freep(&output);
* }
* @endcode
*
* When the conversion is finished, the conversion
* context and everything associated with it must be freed with swr_free().
* A swr_close() function is also available, but it exists mainly for
* compatibility with libavresample, and is not required to be called.
*
* There will be no memory leak if the data is not completely flushed before
* swr_free().
*/
#include <stdint.h>
#include "libavutil/frame.h"
#include "libavutil/samplefmt.h"
#include "libswresample/version.h"
#if LIBSWRESAMPLE_VERSION_MAJOR < 1
#define SWR_CH_MAX 32 ///< Maximum number of channels
#endif
/**
* @name Option constants
* These constants are used for the @ref avoptions interface for lswr.
* @{
*
*/
#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
/** Dithering algorithms */
enum SwrDitherType {
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_NS = 64, ///< not part of API/ABI
SWR_DITHER_NS_LIPSHITZ,
SWR_DITHER_NS_F_WEIGHTED,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
SWR_DITHER_NS_SHIBATA,
SWR_DITHER_NS_LOW_SHIBATA,
SWR_DITHER_NS_HIGH_SHIBATA,
SWR_DITHER_NB, ///< not part of API/ABI
};
/** Resampling Engines */
enum SwrEngine {
SWR_ENGINE_SWR, /**< SW Resampler */
SWR_ENGINE_SOXR, /**< SoX Resampler */
SWR_ENGINE_NB, ///< not part of API/ABI
};
/** Resampling Filter Types */
enum SwrFilterType {
SWR_FILTER_TYPE_CUBIC, /**< Cubic */
SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
};
/**
* @}
*/
/**
* The libswresample context. Unlike libavcodec and libavformat, this structure
* is opaque. This means that if you would like to set options, you must use
* the @ref avoptions API and cannot directly set values to members of the
* structure.
*/
typedef struct SwrContext SwrContext;
/**
* Get the AVClass for SwrContext. It can be used in combination with
* AV_OPT_SEARCH_FAKE_OBJ for examining options.
*
* @see av_opt_find().
* @return the AVClass of SwrContext
*/
const AVClass *swr_get_class(void);
/**
* @name SwrContext constructor functions
* @{
*/
/**
* Allocate SwrContext.
*
* If you use this function you will need to set the parameters (manually or
* with swr_alloc_set_opts()) before calling swr_init().
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc(void);
/**
* Initialize context after user parameters have been set.
* @note The context must be configured using the AVOption API.
*
* @see av_opt_set_int()
* @see av_opt_set_dict()
*
* @param[in,out] s Swr context to initialize
* @return AVERROR error code in case of failure.
*/
int swr_init(struct SwrContext *s);
/**
* Check whether an swr context has been initialized or not.
*
* @param[in] s Swr context to check
* @see swr_init()
* @return positive if it has been initialized, 0 if not initialized
*/
int swr_is_initialized(struct SwrContext *s);
/**
* Allocate SwrContext if needed and set/reset common parameters.
*
* This function does not require s to be allocated with swr_alloc(). On the
* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
* on the allocated context.
*
* @param s existing Swr context if available, or NULL if not
* @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
* @param out_sample_rate output sample rate (frequency in Hz)
* @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
* @param in_sample_rate input sample rate (frequency in Hz)
* @param log_offset logging level offset
* @param log_ctx parent logging context, can be NULL
*
* @see swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
/**
* @}
*
* @name SwrContext destructor functions
* @{
*/
/**
* Free the given SwrContext and set the pointer to NULL.
*
* @param[in] s a pointer to a pointer to Swr context
*/
void swr_free(struct SwrContext **s);
/**
* Closes the context so that swr_is_initialized() returns 0.
*
* The context can be brought back to life by running swr_init(),
* swr_init() can also be used without swr_close().
* This function is mainly provided for simplifying the usecase
* where one tries to support libavresample and libswresample.
*
* @param[in,out] s Swr context to be closed
*/
void swr_close(struct SwrContext *s);
/**
* @}
*
* @name Core conversion functions
* @{
*/
/** Convert audio.
*
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space then the input will be buffered.
* You can avoid this buffering by providing more output space than input.
* Conversion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel
* @param in input buffers, only the first one need to be set in case of packed audio
* @param in_count number of input samples available in one channel
*
* @return number of samples output per channel, negative value on error
*/
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
const uint8_t **in , int in_count);
/**
* Convert the next timestamp from input to output
* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers.
* See ffmpeg-resampler(1) for the two modes of compensation.
*
* @param s[in] initialized Swr context
* @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
* @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
* function used internally for timestamp compensation.
* @return the output timestamp for the next output sample
*/
int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
/**
* @}
*
* @name Low-level option setting functions
* These functons provide a means to set low-level options that is not possible
* with the AVOption API.
* @{
*/
/**
* Activate resampling compensation ("soft" compensation). This function is
* internally called when needed in swr_next_pts().
*
* @param[in,out] s allocated Swr context. If it is not initialized,
* or SWR_FLAG_RESAMPLE is not set, swr_init() is
* called with the flag set.
* @param[in] sample_delta delta in PTS per sample
* @param[in] compensation_distance number of samples to compensate for
* @return >= 0 on success, AVERROR error codes if:
* @li @c s is NULL,
* @li @c compensation_distance is less than 0,
* @li @c compensation_distance is 0 but sample_delta is not,
* @li compensation unsupported by resampler, or
* @li swr_init() fails when called.
*/
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
/**
* Set a customized input channel mapping.
*
* @param[in,out] s allocated Swr context, not yet initialized
* @param[in] channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
/**
* Set a customized remix matrix.
*
* @param s allocated Swr context, not yet initialized
* @param matrix remix coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o
* @param stride offset between lines of the matrix
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
/**
* @}
*
* @name Sample handling functions
* @{
*/
/**
* Drops the specified number of output samples.
*
* This function, along with swr_inject_silence(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_drop_output(struct SwrContext *s, int count);
/**
* Injects the specified number of silence samples.
*
* This function, along with swr_drop_output(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_inject_silence(struct SwrContext *s, int count);
/**
* Gets the delay the next input sample will experience relative to the next output sample.
*
* Swresample can buffer data if more input has been provided than available
* output space, also converting between sample rates needs a delay.
* This function returns the sum of all such delays.
* The exact delay is not necessarily an integer value in either input or
* output sample rate. Especially when downsampling by a large value, the
* output sample rate may be a poor choice to represent the delay, similarly
* for upsampling and the input sample rate.
*
* @param s swr context
* @param base timebase in which the returned delay will be:
* @li if it's set to 1 the returned delay is in seconds
* @li if it's set to 1000 the returned delay is in milliseconds
* @li if it's set to the input sample rate then the returned
* delay is in input samples
* @li if it's set to the output sample rate then the returned
* delay is in output samples
* @li if it's the least common multiple of in_sample_rate and
* out_sample_rate then an exact rounding-free delay will be
* returned
* @returns the delay in 1 / @c base units.
*/
int64_t swr_get_delay(struct SwrContext *s, int64_t base);
/**
* @}
*
* @name Configuration accessors
* @{
*/
/**
* Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
*
* This is useful to check if the build-time libswresample has the same version
* as the run-time one.
*
* @returns the unsigned int-typed version
*/
unsigned swresample_version(void);
/**
* Return the swr build-time configuration.
*
* @returns the build-time @c ./configure flags
*/
const char *swresample_configuration(void);
/**
* Return the swr license.
*
* @returns the license of libswresample, determined at build-time
*/
const char *swresample_license(void);
/**
* @}
*
* @name AVFrame based API
* @{
*/
/**
* Convert the samples in the input AVFrame and write them to the output AVFrame.
*
* Input and output AVFrames must have channel_layout, sample_rate and format set.
*
* If the output AVFrame does not have the data pointers allocated the nb_samples
* field will be set using av_frame_get_buffer()
* is called to allocate the frame.
*
* The output AVFrame can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* or to swr_convert().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. swr_get_delay() tells the number of
* remaining samples. To get this data as output, call this function or
* swr_convert() with NULL input.
*
* If the SwrContext configuration does not match the output and
* input AVFrame settings the conversion does not take place and depending on
* which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
* or the result of a bitwise-OR of them is returned.
*
* @see swr_delay()
* @see swr_convert()
* @see swr_get_delay()
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure or nonmatching
* configuration.
*/
int swr_convert_frame(SwrContext *swr,
AVFrame *output, const AVFrame *input);
/**
* Configure or reconfigure the SwrContext using the information
* provided by the AVFrames.
*
* The original resampling context is reset even on failure.
* The function calls swr_close() internally if the context is open.
*
* @see swr_close();
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure.
*/
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
/**
* @}
* @}
*/
#endif /* SWRESAMPLE_SWRESAMPLE_H */

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Externals/ffmpeg/dev/include/libswresample/version.h vendored
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@ -1,45 +0,0 @@
/*
* Version macros.
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWR_VERSION_H
#define SWR_VERSION_H
/**
* @file
* Libswresample version macros
*/
#include "libavutil/avutil.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 1
#define LIBSWRESAMPLE_VERSION_MINOR 1
#define LIBSWRESAMPLE_VERSION_MICRO 100
#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_VERSION AV_VERSION(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_BUILD LIBSWRESAMPLE_VERSION_INT
#define LIBSWRESAMPLE_IDENT "SwR" AV_STRINGIFY(LIBSWRESAMPLE_VERSION)
#endif /* SWR_VERSION_H */

219
Externals/ffmpeg/dev/lib/avcodec-56.def vendored
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@ -1,219 +0,0 @@
EXPORTS
audio_resample
audio_resample_close
av_audio_convert
av_audio_convert_alloc
av_audio_convert_free
av_audio_resample_init
av_bitstream_filter_close
av_bitstream_filter_filter
av_bitstream_filter_init
av_bitstream_filter_next
av_codec_ffversion DATA
av_codec_get_chroma_intra_matrix
av_codec_get_codec_descriptor
av_codec_get_lowres
av_codec_get_max_lowres
av_codec_get_pkt_timebase
av_codec_get_seek_preroll
av_codec_is_decoder
av_codec_is_encoder
av_codec_next
av_codec_set_chroma_intra_matrix
av_codec_set_codec_descriptor
av_codec_set_lowres
av_codec_set_pkt_timebase
av_codec_set_seek_preroll
av_copy_packet
av_copy_packet_side_data
av_dct_calc
av_dct_end
av_dct_init
av_destruct_packet
av_dup_packet
av_dv_codec_profile
av_dv_codec_profile2
av_dv_frame_profile
av_fast_padded_malloc
av_fast_padded_mallocz
av_fft_calc
av_fft_end
av_fft_init
av_fft_permute
av_free_packet
av_get_audio_frame_duration
av_get_bits_per_sample
av_get_codec_tag_string
av_get_exact_bits_per_sample
av_get_pcm_codec
av_get_profile_name
av_grow_packet
av_hwaccel_next
av_imdct_calc
av_imdct_half
av_init_packet
av_lockmgr_register
av_log_ask_for_sample
av_log_missing_feature
av_mdct_calc
av_mdct_end
av_mdct_init
av_new_packet
av_packet_copy_props
av_packet_free_side_data
av_packet_from_data
av_packet_get_side_data
av_packet_merge_side_data
av_packet_move_ref
av_packet_new_side_data
av_packet_pack_dictionary
av_packet_ref
av_packet_rescale_ts
av_packet_shrink_side_data
av_packet_split_side_data
av_packet_unpack_dictionary
av_packet_unref
av_parser_change
av_parser_close
av_parser_init
av_parser_next
av_parser_parse2
av_picture_copy
av_picture_crop
av_picture_pad
av_rdft_calc
av_rdft_end
av_rdft_init
av_register_bitstream_filter
av_register_codec_parser
av_register_hwaccel
av_resample
av_resample_close
av_resample_compensate
av_resample_init
av_shrink_packet
av_vorbis_parse_frame
av_vorbis_parse_frame_flags
av_vorbis_parse_free
av_vorbis_parse_init
av_vorbis_parse_reset
av_xiphlacing
available_bits
avcodec_align_dimensions
avcodec_align_dimensions2
avcodec_alloc_context3
avcodec_alloc_frame
avcodec_chroma_pos_to_enum
avcodec_close
avcodec_configuration
avcodec_copy_context
avcodec_dct_alloc
avcodec_dct_get_class
avcodec_dct_init
avcodec_decode_audio3
avcodec_decode_audio4
avcodec_decode_subtitle2
avcodec_decode_video2
avcodec_default_execute
avcodec_default_execute2
avcodec_default_get_buffer
avcodec_default_get_buffer2
avcodec_default_get_format
avcodec_default_reget_buffer
avcodec_default_release_buffer
avcodec_descriptor_get
avcodec_descriptor_get_by_name
avcodec_descriptor_next
avcodec_encode_audio
avcodec_encode_audio2
avcodec_encode_subtitle
avcodec_encode_video
avcodec_encode_video2
avcodec_enum_to_chroma_pos
avcodec_fill_audio_frame
avcodec_find_best_pix_fmt2
avcodec_find_best_pix_fmt_of_2
avcodec_find_best_pix_fmt_of_list
avcodec_find_decoder
avcodec_find_decoder_by_name
avcodec_find_encoder
avcodec_find_encoder_by_name
avcodec_flush_buffers
avcodec_free_context
avcodec_free_frame
avcodec_get_chroma_sub_sample
avcodec_get_class
avcodec_get_context_defaults3
avcodec_get_edge_width
avcodec_get_frame_class
avcodec_get_frame_defaults
avcodec_get_name
avcodec_get_pix_fmt_loss
avcodec_get_subtitle_rect_class
avcodec_get_type
avcodec_is_open
avcodec_license
avcodec_open2
avcodec_pix_fmt_to_codec_tag
avcodec_register
avcodec_register_all
avcodec_set_dimensions
avcodec_string
avcodec_version
aver_isf_history
avpicture_alloc
avpicture_deinterlace
avpicture_fill
avpicture_free
avpicture_get_size
avpicture_layout
avpriv_aac_parse_header
avpriv_ac3_channel_layout_tab DATA
avpriv_ac3_parse_header
avpriv_ac3_parse_header2
avpriv_align_put_bits
avpriv_bprint_to_extradata
avpriv_color_frame
avpriv_copy_bits
avpriv_copy_pce_data
avpriv_dca_convert_bitstream
avpriv_dca_sample_rates DATA
avpriv_dirac_parse_sequence_header
avpriv_dnxhd_get_frame_size
avpriv_do_elbg
avpriv_dv_frame_profile2
avpriv_exif_decode_ifd
avpriv_find_pix_fmt
avpriv_find_start_code
avpriv_flac_is_extradata_valid
avpriv_flac_parse_streaminfo
avpriv_get_raw_pix_fmt_tags
avpriv_h264_has_num_reorder_frames
avpriv_init_elbg
avpriv_lock_avformat
avpriv_mjpeg_bits_ac_chrominance DATA
avpriv_mjpeg_bits_ac_luminance DATA
avpriv_mjpeg_bits_dc_chrominance DATA
avpriv_mjpeg_bits_dc_luminance DATA
avpriv_mjpeg_val_ac_chrominance DATA
avpriv_mjpeg_val_ac_luminance DATA
avpriv_mjpeg_val_dc DATA
avpriv_mpa_bitrate_tab DATA
avpriv_mpa_decode_header
avpriv_mpa_decode_header2
avpriv_mpa_freq_tab DATA
avpriv_mpeg4audio_get_config
avpriv_mpeg4audio_sample_rates DATA
avpriv_mpegaudio_decode_header
avpriv_pix_fmt_bps_avi DATA
avpriv_pix_fmt_bps_mov DATA
avpriv_put_string
avpriv_split_xiph_headers
avpriv_tak_parse_streaminfo
avpriv_toupper4
avpriv_unlock_avformat
avpriv_vorbis_parse_extradata
avpriv_vorbis_parse_frame
avpriv_vorbis_parse_frame_flags
avpriv_vorbis_parse_reset
avsubtitle_free

19
Externals/ffmpeg/dev/lib/avdevice-56.def vendored
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@ -1,19 +0,0 @@
EXPORTS
av_device_capabilities DATA
av_device_ffversion DATA
av_input_audio_device_next
av_input_video_device_next
av_output_audio_device_next
av_output_video_device_next
avdevice_app_to_dev_control_message
avdevice_capabilities_create
avdevice_capabilities_free
avdevice_configuration
avdevice_dev_to_app_control_message
avdevice_free_list_devices
avdevice_license
avdevice_list_devices
avdevice_list_input_sources
avdevice_list_output_sinks
avdevice_register_all
avdevice_version

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EXPORTS
av_abuffersink_params_alloc
av_buffersink_get_buffer_ref
av_buffersink_get_frame
av_buffersink_get_frame_flags
av_buffersink_get_frame_rate
av_buffersink_get_samples
av_buffersink_params_alloc
av_buffersink_poll_frame
av_buffersink_read
av_buffersink_read_samples
av_buffersink_set_frame_size
av_buffersrc_add_frame
av_buffersrc_add_frame_flags
av_buffersrc_add_ref
av_buffersrc_buffer
av_buffersrc_get_nb_failed_requests
av_buffersrc_write_frame
av_filter_ffversion DATA
av_filter_next
avfilter_add_matrix
avfilter_all_channel_layouts DATA
avfilter_config_links
avfilter_configuration
avfilter_copy_buf_props
avfilter_copy_buffer_ref_props
avfilter_copy_frame_props
avfilter_free
avfilter_get_audio_buffer_ref_from_arrays
avfilter_get_audio_buffer_ref_from_arrays_channels
avfilter_get_audio_buffer_ref_from_frame
avfilter_get_buffer_ref_from_frame
avfilter_get_by_name
avfilter_get_class
avfilter_get_matrix
avfilter_get_video_buffer_ref_from_arrays
avfilter_get_video_buffer_ref_from_frame
avfilter_graph_add_filter
avfilter_graph_alloc
avfilter_graph_alloc_filter
avfilter_graph_config
avfilter_graph_create_filter
avfilter_graph_dump
avfilter_graph_free
avfilter_graph_get_filter
avfilter_graph_parse
avfilter_graph_parse2
avfilter_graph_parse_ptr
avfilter_graph_queue_command
avfilter_graph_request_oldest
avfilter_graph_send_command
avfilter_graph_set_auto_convert
avfilter_init_dict
avfilter_init_filter
avfilter_init_str
avfilter_inout_alloc
avfilter_inout_free
avfilter_insert_filter
avfilter_license
avfilter_link
avfilter_link_free
avfilter_link_get_channels
avfilter_link_set_closed
avfilter_make_format64_list
avfilter_mul_matrix
avfilter_next
avfilter_open
avfilter_pad_count
avfilter_pad_get_name
avfilter_pad_get_type
avfilter_process_command
avfilter_ref_buffer
avfilter_ref_get_channels
avfilter_register
avfilter_register_all
avfilter_sub_matrix
avfilter_transform
avfilter_uninit
avfilter_unref_buffer
avfilter_unref_bufferp
avfilter_version

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Externals/ffmpeg/dev/lib/avformat-56.def vendored
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EXPORTS
av_add_index_entry
av_append_packet
av_codec_get_id
av_codec_get_tag
av_codec_get_tag2
av_convert_lang_to
av_demuxer_open
av_dump_format
av_filename_number_test
av_find_best_stream
av_find_default_stream_index
av_find_input_format
av_find_program_from_stream
av_fmt_ctx_get_duration_estimation_method
av_format_ffversion DATA
av_format_get_audio_codec
av_format_get_control_message_cb
av_format_get_metadata_header_padding
av_format_get_opaque
av_format_get_probe_score
av_format_get_subtitle_codec
av_format_get_video_codec
av_format_inject_global_side_data
av_format_set_audio_codec
av_format_set_control_message_cb
av_format_set_metadata_header_padding
av_format_set_opaque
av_format_set_subtitle_codec
av_format_set_video_codec
av_get_frame_filename
av_get_output_timestamp
av_get_packet
av_guess_codec
av_guess_format
av_guess_frame_rate
av_guess_sample_aspect_ratio
av_hex_dump
av_hex_dump_log
av_iformat_next
av_index_search_timestamp
av_interleaved_write_frame
av_interleaved_write_uncoded_frame
av_match_ext
av_new_program
av_oformat_next
av_pkt_dump2
av_pkt_dump_log2
av_probe_input_buffer
av_probe_input_buffer2
av_probe_input_format
av_probe_input_format2
av_probe_input_format3
av_read_frame
av_read_pause
av_read_play
av_register_all
av_register_input_format
av_register_output_format
av_sdp_create
av_seek_frame
av_stream_get_end_pts
av_stream_get_parser
av_stream_get_r_frame_rate
av_stream_get_recommended_encoder_configuration
av_stream_get_side_data
av_stream_set_r_frame_rate
av_stream_set_recommended_encoder_configuration
av_url_split
av_write_frame
av_write_trailer
av_write_uncoded_frame
av_write_uncoded_frame_query
avformat_alloc_context
avformat_alloc_output_context2
avformat_close_input
avformat_configuration
avformat_find_stream_info
avformat_free_context
avformat_get_class
avformat_get_mov_audio_tags
avformat_get_mov_video_tags
avformat_get_riff_audio_tags
avformat_get_riff_video_tags
avformat_license
avformat_match_stream_specifier
avformat_network_deinit
avformat_network_init
avformat_new_stream
avformat_open_input
avformat_query_codec
avformat_queue_attached_pictures
avformat_seek_file
avformat_version
avformat_write_header
avio_alloc_context
avio_check
avio_close
avio_close_dyn_buf
avio_closep
avio_enum_protocols
avio_feof
avio_find_protocol_name
avio_flush
avio_get_str
avio_get_str16be
avio_get_str16le
avio_open
avio_open2
avio_open_dyn_buf
avio_pause
avio_printf
avio_put_str
avio_put_str16le
avio_r8
avio_rb16
avio_rb24
avio_rb32
avio_rb64
avio_read
avio_read_to_bprint
avio_rl16
avio_rl24
avio_rl32
avio_rl64
avio_seek
avio_seek_time
avio_size
avio_skip
avio_w8
avio_wb16
avio_wb24
avio_wb32
avio_wb64
avio_wl16
avio_wl24
avio_wl32
avio_wl64
avio_write
avpriv_dv_get_packet
avpriv_dv_init_demux
avpriv_dv_produce_packet
avpriv_mpegts_parse_close
avpriv_mpegts_parse_open
avpriv_mpegts_parse_packet
avpriv_new_chapter
avpriv_set_pts_info
ff_inet_aton
ff_rtp_get_local_rtcp_port
ff_rtp_get_local_rtp_port
ff_rtsp_parse_line
ff_socket_nonblock
ffio_open_dyn_packet_buf
ffio_set_buf_size
ffurl_close
ffurl_open
ffurl_read_complete
ffurl_seek
ffurl_size
ffurl_write
url_feof

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EXPORTS
av_add_q
av_add_stable
av_adler32_update
av_aes_alloc
av_aes_crypt
av_aes_init
av_aes_size DATA
av_asprintf
av_audio_fifo_alloc
av_audio_fifo_drain
av_audio_fifo_free
av_audio_fifo_read
av_audio_fifo_realloc
av_audio_fifo_reset
av_audio_fifo_size
av_audio_fifo_space
av_audio_fifo_write
av_base64_decode
av_base64_encode
av_basename
av_blowfish_crypt
av_blowfish_crypt_ecb
av_blowfish_init
av_bmg_get
av_bprint_append_data
av_bprint_channel_layout
av_bprint_chars
av_bprint_clear
av_bprint_escape
av_bprint_finalize
av_bprint_get_buffer
av_bprint_init
av_bprint_init_for_buffer
av_bprint_strftime
av_bprintf
av_buffer_alloc
av_buffer_allocz
av_buffer_create
av_buffer_default_free
av_buffer_get_opaque
av_buffer_get_ref_count
av_buffer_is_writable
av_buffer_make_writable
av_buffer_pool_get
av_buffer_pool_init
av_buffer_pool_uninit
av_buffer_realloc
av_buffer_ref
av_buffer_unref
av_calloc
av_camellia_alloc
av_camellia_crypt
av_camellia_init
av_camellia_size DATA
av_cast5_alloc
av_cast5_crypt
av_cast5_crypt2
av_cast5_init
av_cast5_size DATA
av_channel_layout_extract_channel
av_chroma_location_name
av_color_primaries_name
av_color_range_name
av_color_space_name
av_color_transfer_name
av_compare_mod
av_compare_ts
av_cpu_count
av_crc
av_crc_get_table
av_crc_init
av_ctz
av_d2q
av_d2str
av_default_get_category
av_default_item_name
av_des_crypt
av_des_init
av_des_mac
av_dict_copy
av_dict_count
av_dict_free
av_dict_get
av_dict_get_string
av_dict_parse_string
av_dict_set
av_dict_set_int
av_dirname
av_display_matrix_flip
av_display_rotation_get
av_display_rotation_set
av_div_q
av_downmix_info_update_side_data
av_dynarray2_add
av_dynarray_add
av_dynarray_add_nofree
av_escape
av_expr_eval
av_expr_free
av_expr_parse
av_expr_parse_and_eval
av_fast_malloc
av_fast_realloc
av_fifo_alloc
av_fifo_alloc_array
av_fifo_drain
av_fifo_free
av_fifo_freep
av_fifo_generic_read
av_fifo_generic_write
av_fifo_grow
av_fifo_realloc2
av_fifo_reset
av_fifo_size
av_fifo_space
av_file_map
av_file_unmap
av_find_best_pix_fmt_of_2
av_find_info_tag
av_find_nearest_q_idx
av_fopen_utf8
av_force_cpu_flags
av_frame_alloc
av_frame_clone
av_frame_copy
av_frame_copy_props
av_frame_free
av_frame_get_best_effort_timestamp
av_frame_get_buffer
av_frame_get_channel_layout
av_frame_get_channels
av_frame_get_color_range
av_frame_get_colorspace
av_frame_get_decode_error_flags
av_frame_get_metadata
av_frame_get_pkt_duration
av_frame_get_pkt_pos
av_frame_get_pkt_size
av_frame_get_plane_buffer
av_frame_get_qp_table
av_frame_get_sample_rate
av_frame_get_side_data
av_frame_is_writable
av_frame_make_writable
av_frame_move_ref
av_frame_new_side_data
av_frame_ref
av_frame_remove_side_data
av_frame_set_best_effort_timestamp
av_frame_set_channel_layout
av_frame_set_channels
av_frame_set_color_range
av_frame_set_colorspace
av_frame_set_decode_error_flags
av_frame_set_metadata
av_frame_set_pkt_duration
av_frame_set_pkt_pos
av_frame_set_pkt_size
av_frame_set_qp_table
av_frame_set_sample_rate
av_frame_side_data_name
av_frame_unref
av_free
av_freep
av_gcd
av_get_alt_sample_fmt
av_get_bits_per_pixel
av_get_bytes_per_sample
av_get_channel_description
av_get_channel_layout
av_get_channel_layout_channel_index
av_get_channel_layout_nb_channels
av_get_channel_layout_string
av_get_channel_name
av_get_colorspace_name
av_get_cpu_flags
av_get_default_channel_layout
av_get_double
av_get_int
av_get_known_color_name
av_get_media_type_string
av_get_packed_sample_fmt
av_get_padded_bits_per_pixel
av_get_picture_type_char
av_get_pix_fmt
av_get_pix_fmt_loss
av_get_pix_fmt_name
av_get_pix_fmt_string
av_get_planar_sample_fmt
av_get_q
av_get_random_seed
av_get_sample_fmt
av_get_sample_fmt_name
av_get_sample_fmt_string
av_get_standard_channel_layout
av_get_string
av_get_time_base_q
av_get_token
av_gettime
av_gettime_relative
av_gettime_relative_is_monotonic
av_hash_alloc
av_hash_final
av_hash_final_b64
av_hash_final_bin
av_hash_final_hex
av_hash_freep
av_hash_get_name
av_hash_get_size
av_hash_init
av_hash_names
av_hash_update
av_hmac_alloc
av_hmac_calc
av_hmac_final
av_hmac_free
av_hmac_init
av_hmac_update
av_image_alloc
av_image_check_sar
av_image_check_size
av_image_copy
av_image_copy_plane
av_image_copy_to_buffer
av_image_fill_arrays
av_image_fill_linesizes
av_image_fill_max_pixsteps
av_image_fill_pointers
av_image_get_buffer_size
av_image_get_linesize
av_int_list_length_for_size
av_isdigit
av_isgraph
av_isspace
av_isxdigit
av_lfg_init
av_log
av_log2
av_log2_16bit
av_log_default_callback
av_log_format_line
av_log_get_flags
av_log_get_level
av_log_set_callback
av_log_set_flags
av_log_set_level
av_lzo1x_decode
av_malloc
av_mallocz
av_match_list
av_match_name
av_max_alloc
av_md5_alloc
av_md5_final
av_md5_init
av_md5_size DATA
av_md5_sum
av_md5_update
av_memcpy_backptr
av_memdup
av_mul_q
av_murmur3_alloc
av_murmur3_final
av_murmur3_init
av_murmur3_init_seeded
av_murmur3_update
av_nearer_q
av_next_option
av_opt_child_class_next
av_opt_child_next
av_opt_copy
av_opt_eval_double
av_opt_eval_flags
av_opt_eval_float
av_opt_eval_int
av_opt_eval_int64
av_opt_eval_q
av_opt_find
av_opt_find2
av_opt_flag_is_set
av_opt_free
av_opt_freep_ranges
av_opt_get
av_opt_get_channel_layout
av_opt_get_dict_val
av_opt_get_double
av_opt_get_image_size
av_opt_get_int
av_opt_get_key_value
av_opt_get_pixel_fmt
av_opt_get_q
av_opt_get_sample_fmt
av_opt_get_video_rate
av_opt_is_set_to_default
av_opt_is_set_to_default_by_name
av_opt_next
av_opt_ptr
av_opt_query_ranges
av_opt_query_ranges_default
av_opt_serialize
av_opt_set
av_opt_set_bin
av_opt_set_channel_layout
av_opt_set_defaults
av_opt_set_defaults2
av_opt_set_dict
av_opt_set_dict2
av_opt_set_dict_val
av_opt_set_double
av_opt_set_from_string
av_opt_set_image_size
av_opt_set_int
av_opt_set_pixel_fmt
av_opt_set_q
av_opt_set_sample_fmt
av_opt_set_video_rate
av_opt_show2
av_parse_color
av_parse_cpu_caps
av_parse_cpu_flags
av_parse_ratio
av_parse_time
av_parse_video_rate
av_parse_video_size
av_pix_fmt_count_planes
av_pix_fmt_desc_get
av_pix_fmt_desc_get_id
av_pix_fmt_desc_next
av_pix_fmt_descriptors DATA
av_pix_fmt_get_chroma_sub_sample
av_pix_fmt_swap_endianness
av_pixelutils_get_sad_fn
av_rc4_crypt
av_rc4_init
av_read_image_line
av_realloc
av_realloc_array
av_realloc_f
av_reallocp
av_reallocp_array
av_reduce
av_rescale
av_rescale_delta
av_rescale_q
av_rescale_q_rnd
av_rescale_rnd
av_reverse DATA
av_ripemd_alloc
av_ripemd_final
av_ripemd_init
av_ripemd_size DATA
av_ripemd_update
av_sample_fmt_is_planar
av_samples_alloc
av_samples_alloc_array_and_samples
av_samples_copy
av_samples_fill_arrays
av_samples_get_buffer_size
av_samples_set_silence
av_set_cpu_flags_mask
av_set_double
av_set_int
av_set_options_string
av_set_q
av_set_string3
av_sha512_alloc
av_sha512_final
av_sha512_init
av_sha512_size DATA
av_sha512_update
av_sha_alloc
av_sha_final
av_sha_init
av_sha_size DATA
av_sha_update
av_small_strptime
av_stereo3d_alloc
av_stereo3d_create_side_data
av_strcasecmp
av_strdup
av_strerror
av_stristart
av_stristr
av_strlcat
av_strlcatf
av_strlcpy
av_strncasecmp
av_strndup
av_strnstr
av_strstart
av_strtod
av_strtok
av_sub_q
av_tempfile
av_thread_message_queue_alloc
av_thread_message_queue_free
av_thread_message_queue_recv
av_thread_message_queue_send
av_thread_message_queue_set_err_recv
av_thread_message_queue_set_err_send
av_timecode_adjust_ntsc_framenum2
av_timecode_check_frame_rate
av_timecode_get_smpte_from_framenum
av_timecode_init
av_timecode_init_from_string
av_timecode_make_mpeg_tc_string
av_timecode_make_smpte_tc_string
av_timecode_make_string
av_timegm
av_tree_destroy
av_tree_enumerate
av_tree_find
av_tree_insert
av_tree_node_alloc
av_tree_node_size DATA
av_usleep
av_utf8_decode
av_util_ffversion DATA
av_vbprintf
av_vlog
av_write_image_line
av_xtea_crypt
av_xtea_init
avpriv_alloc_fixed_dsp
avpriv_cga_font DATA
avpriv_emms_yasm DATA
avpriv_float_dsp_alloc
avpriv_float_dsp_init
avpriv_frame_get_metadatap
avpriv_init_lls
avpriv_open
avpriv_report_missing_feature
avpriv_request_sample
avpriv_scalarproduct_float_c
avpriv_set_systematic_pal2
avpriv_solve_lls
avpriv_vga16_font DATA
avutil_configuration
avutil_license
avutil_version

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EXPORTS
postproc_configuration
postproc_ffversion DATA
postproc_license
postproc_version
pp_free_context
pp_free_mode
pp_get_context
pp_get_mode_by_name_and_quality
pp_help DATA
pp_postprocess

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EXPORTS
swr_alloc
swr_alloc_set_opts
swr_close
swr_config_frame
swr_convert
swr_convert_frame
swr_drop_output
swr_ffversion DATA
swr_free
swr_get_class
swr_get_delay
swr_init
swr_inject_silence
swr_is_initialized
swr_next_pts
swr_set_channel_mapping
swr_set_compensation
swr_set_matrix
swresample_configuration
swresample_license
swresample_version

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EXPORTS
sws_addVec
sws_allocVec
sws_alloc_context
sws_cloneVec
sws_context_class DATA
sws_convVec
sws_convertPalette8ToPacked24
sws_convertPalette8ToPacked32
sws_freeContext
sws_freeFilter
sws_freeVec
sws_getCachedContext
sws_getCoefficients
sws_getColorspaceDetails
sws_getConstVec
sws_getContext
sws_getDefaultFilter
sws_getGaussianVec
sws_getIdentityVec
sws_get_class
sws_init_context
sws_isSupportedEndiannessConversion
sws_isSupportedInput
sws_isSupportedOutput
sws_normalizeVec
sws_printVec2
sws_rgb2rgb_init
sws_scale
sws_scaleVec
sws_setColorspaceDetails
sws_shiftVec
sws_subVec
swscale_configuration
swscale_license
swscale_version

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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by GNU Texinfo 5.2, http://www.gnu.org/software/texinfo/ -->
<head>
<meta charset="utf-8">
<title>
Developer Documentation
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div style="width: 95%; margin: auto">
<h1>
Developer Documentation
</h1>
<div align="center">
</div>
<a name="SEC_Top"></a>
<a name="SEC_Contents"></a>
<h2 class="contents-heading">Table of Contents</h2>
<div class="contents">
<ul class="no-bullet">
<li><a name="toc-Developers-Guide" href="#Developers-Guide">1 Developers Guide</a>
<ul class="no-bullet">
<li><a name="toc-Notes-for-external-developers" href="#Notes-for-external-developers">1.1 Notes for external developers</a></li>
<li><a name="toc-Contributing" href="#Contributing">1.2 Contributing</a></li>
<li><a name="toc-Coding-Rules-1" href="#Coding-Rules-1">1.3 Coding Rules</a>
<ul class="no-bullet">
<li><a name="toc-Code-formatting-conventions" href="#Code-formatting-conventions">1.3.1 Code formatting conventions</a></li>
<li><a name="toc-Comments" href="#Comments">1.3.2 Comments</a></li>
<li><a name="toc-C-language-features" href="#C-language-features">1.3.3 C language features</a></li>
<li><a name="toc-Naming-conventions" href="#Naming-conventions">1.3.4 Naming conventions</a></li>
<li><a name="toc-Miscellaneous-conventions" href="#Miscellaneous-conventions">1.3.5 Miscellaneous conventions</a></li>
<li><a name="toc-Editor-configuration" href="#Editor-configuration">1.3.6 Editor configuration</a></li>
</ul></li>
<li><a name="toc-Development-Policy" href="#Development-Policy">1.4 Development Policy</a></li>
<li><a name="toc-Submitting-patches-1" href="#Submitting-patches-1">1.5 Submitting patches</a></li>
<li><a name="toc-New-codecs-or-formats-checklist" href="#New-codecs-or-formats-checklist">1.6 New codecs or formats checklist</a></li>
<li><a name="toc-patch-submission-checklist" href="#patch-submission-checklist">1.7 patch submission checklist</a></li>
<li><a name="toc-Patch-review-process" href="#Patch-review-process">1.8 Patch review process</a></li>
<li><a name="toc-Regression-tests-1" href="#Regression-tests-1">1.9 Regression tests</a>
<ul class="no-bullet">
<li><a name="toc-Adding-files-to-the-fate_002dsuite-dataset" href="#Adding-files-to-the-fate_002dsuite-dataset">1.9.1 Adding files to the fate-suite dataset</a></li>
<li><a name="toc-Visualizing-Test-Coverage" href="#Visualizing-Test-Coverage">1.9.2 Visualizing Test Coverage</a></li>
<li><a name="toc-Using-Valgrind" href="#Using-Valgrind">1.9.3 Using Valgrind</a></li>
</ul></li>
<li><a name="toc-Release-process-1" href="#Release-process-1">1.10 Release process</a>
<ul class="no-bullet">
<li><a name="toc-Criteria-for-Point-Releases-1" href="#Criteria-for-Point-Releases-1">1.10.1 Criteria for Point Releases</a></li>
<li><a name="toc-Release-Checklist" href="#Release-Checklist">1.10.2 Release Checklist</a></li>
</ul></li>
</ul></li>
</ul>
</div>
<a name="Developers-Guide"></a>
<h2 class="chapter">1 Developers Guide<span class="pull-right"><a class="anchor hidden-xs" href="#Developers-Guide" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Developers-Guide" aria-hidden="true">TOC</a></span></h2>
<a name="Notes-for-external-developers"></a>
<h3 class="section">1.1 Notes for external developers<span class="pull-right"><a class="anchor hidden-xs" href="#Notes-for-external-developers" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Notes-for-external-developers" aria-hidden="true">TOC</a></span></h3>
<p>This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in <samp>doc/examples</samp> and in the source code to
see how the public API is employed.
</p>
<p>You can use the FFmpeg libraries in your commercial program, but you
are encouraged to <em>publish any patch you make</em>. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
</p>
<p>For more detailed legal information about the use of FFmpeg in
external programs read the <samp>LICENSE</samp> file in the source tree and
consult <a href="http://ffmpeg.org/legal.html">http://ffmpeg.org/legal.html</a>.
</p>
<a name="Contributing"></a>
<h3 class="section">1.2 Contributing<span class="pull-right"><a class="anchor hidden-xs" href="#Contributing" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Contributing" aria-hidden="true">TOC</a></span></h3>
<p>There are 3 ways by which code gets into ffmpeg.
</p><ul>
<li> Submitting Patches to the main developer mailing list
see <a href="#Submitting-patches">Submitting patches</a> for details.
</li><li> Directly committing changes to the main tree.
</li><li> Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
</li></ul>
<p>Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the <a href="#Coding-Rules">Coding Rules</a>.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
</p>
<a name="Coding-Rules"></a><a name="Coding-Rules-1"></a>
<h3 class="section">1.3 Coding Rules<span class="pull-right"><a class="anchor hidden-xs" href="#Coding-Rules-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Coding-Rules-1" aria-hidden="true">TOC</a></span></h3>
<a name="Code-formatting-conventions"></a>
<h4 class="subsection">1.3.1 Code formatting conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Code-formatting-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Code-formatting-conventions" aria-hidden="true">TOC</a></span></h4>
<p>There are the following guidelines regarding the indentation in files:
</p>
<ul>
<li> Indent size is 4.
</li><li> The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
</li><li> You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
</li></ul>
<p>The presentation is one inspired by &rsquo;indent -i4 -kr -nut&rsquo;.
</p>
<p>The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
</p>
<a name="Comments"></a>
<h4 class="subsection">1.3.2 Comments<span class="pull-right"><a class="anchor hidden-xs" href="#Comments" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Comments" aria-hidden="true">TOC</a></span></h4>
<p>Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
</p>
<p>Avoid Qt-style and similar Doxygen syntax with <code>!</code> in it, i.e. replace
<code>//!</code> with <code>///</code> and similar. Also @ syntax should be employed
for markup commands, i.e. use <code>@param</code> and not <code>\param</code>.
</p>
<div class="example">
<pre class="example">/**
* @file
* MPEG codec.
* @author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar {
int var1; /**&lt; var1 description */
int var2; ///&lt; var2 description
/** var3 description */
int var3;
} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @param my_parameter description of my_parameter
* @return return value description
*/
int myfunc(int my_parameter)
...
</pre></div>
<a name="C-language-features"></a>
<h4 class="subsection">1.3.3 C language features<span class="pull-right"><a class="anchor hidden-xs" href="#C-language-features" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-C-language-features" aria-hidden="true">TOC</a></span></h4>
<p>FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
</p>
<ul>
<li> the &lsquo;<samp>inline</samp>&rsquo; keyword;
</li><li> &lsquo;<samp>//</samp>&rsquo; comments;
</li><li> designated struct initializers (&lsquo;<samp>struct s x = { .i = 17 };</samp>&rsquo;)
</li><li> compound literals (&lsquo;<samp>x = (struct s) { 17, 23 };</samp>&rsquo;)
</li></ul>
<p>These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
</p>
<p>All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
</p>
<ul>
<li> mixing statements and declarations;
</li><li> &lsquo;<samp>long long</samp>&rsquo; (use &lsquo;<samp>int64_t</samp>&rsquo; instead);
</li><li> &lsquo;<samp>__attribute__</samp>&rsquo; not protected by &lsquo;<samp>#ifdef __GNUC__</samp>&rsquo; or similar;
</li><li> GCC statement expressions (&lsquo;<samp>(x = ({ int y = 4; y; })</samp>&rsquo;).
</li></ul>
<a name="Naming-conventions"></a>
<h4 class="subsection">1.3.4 Naming conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Naming-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Naming-conventions" aria-hidden="true">TOC</a></span></h4>
<p>All names should be composed with underscores (_), not CamelCase. For example,
&lsquo;<samp>avfilter_get_video_buffer</samp>&rsquo; is an acceptable function name and
&lsquo;<samp>AVFilterGetVideo</samp>&rsquo; is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
</p>
<p>There are the following conventions for naming variables and functions:
</p>
<ul>
<li> For local variables no prefix is required.
</li><li> For file-scope variables and functions declared as <code>static</code>, no prefix
is required.
</li><li> For variables and functions visible outside of file scope, but only used
internally by a library, an <code>ff_</code> prefix should be used,
e.g. &lsquo;<samp>ff_w64_demuxer</samp>&rsquo;.
</li><li> For variables and functions visible outside of file scope, used internally
across multiple libraries, use <code>avpriv_</code> as prefix, for example,
&lsquo;<samp>avpriv_aac_parse_header</samp>&rsquo;.
</li><li> Each library has its own prefix for public symbols, in addition to the
commonly used <code>av_</code> (<code>avformat_</code> for libavformat,
<code>avcodec_</code> for libavcodec, <code>swr_</code> for libswresample, etc).
Check the existing code and choose names accordingly.
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
<code>lib&lt;name&gt;/lib&lt;name&gt;.v</code> files.
</li></ul>
<p>Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in <code>_t</code> are reserved by
<a href="http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02">POSIX</a>.
Also avoid names starting with <code>__</code> or <code>_</code> followed by an uppercase
letter as they are reserved by the C standard. Names starting with <code>_</code>
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with <code>_</code> altogether.
</p>
<a name="Miscellaneous-conventions"></a>
<h4 class="subsection">1.3.5 Miscellaneous conventions<span class="pull-right"><a class="anchor hidden-xs" href="#Miscellaneous-conventions" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Miscellaneous-conventions" aria-hidden="true">TOC</a></span></h4>
<ul>
<li> fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
</li><li> Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don&rsquo;t make the code easier to understand.
</li></ul>
<a name="Editor-configuration"></a>
<h4 class="subsection">1.3.6 Editor configuration<span class="pull-right"><a class="anchor hidden-xs" href="#Editor-configuration" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Editor-configuration" aria-hidden="true">TOC</a></span></h4>
<p>In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your <samp>.vimrc</samp>:
</p><div class="example">
<pre class="example">&quot; indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
&quot; Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
&quot; Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
&quot; Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@&lt;!$/
</pre></div>
<p>For Emacs, add these roughly equivalent lines to your <samp>.emacs.d/init.el</samp>:
</p><div class="example">
<pre class="example">(c-add-style &quot;ffmpeg&quot;
'(&quot;k&amp;r&quot;
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style &quot;ffmpeg&quot;)
</pre></div>
<a name="Development-Policy"></a>
<h3 class="section">1.4 Development Policy<span class="pull-right"><a class="anchor hidden-xs" href="#Development-Policy" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Development-Policy" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Contributions should be licensed under the
<a href="http://www.gnu.org/licenses/lgpl-2.1.html">LGPL 2.1</a>,
including an &quot;or any later version&quot; clause, or, if you prefer
a gift-style license, the
<a href="http://opensource.org/licenses/isc-license.txt">ISC</a> or
<a href="http://mit-license.org/">MIT</a> license.
<a href="http://www.gnu.org/licenses/gpl-2.0.html">GPL 2</a> including
an &quot;or any later version&quot; clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
</li><li> You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers&rsquo;
work.
</li><li> The commit message should have a short first line in the form of
a &lsquo;<samp>topic: short description</samp>&rsquo; as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
</li><li> You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
</li><li> Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
</li><li> Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
<p>Note: Redundant code can be removed.
</p>
</li><li> Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
</li><li> We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
<p>NOTE: If you had to put if(){ .. } over a large (&gt; 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
</p>
</li><li> Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as &quot;fixed!&quot; or &quot;Changed it.&quot; are unacceptable.
Recommended format:
<div class="example">
<pre class="example">area changed: Short 1 line description
details describing what and why and giving references.
</pre></div>
</li><li> Make sure the author of the commit is set correctly. (see git commit &ndash;author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
</li><li> When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
</li><li> Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
</li><li> Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
</li><li> Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
</li><li> Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
</li><li> Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
</li><li> Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
</li><li> Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
</li><li> Make sure that no parts of the codebase that you maintain are missing from the
<samp>MAINTAINERS</samp> file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help
finding a new maintainer and also don&rsquo;t forget updating the <samp>MAINTAINERS</samp> file.
</li></ol>
<p>We think our rules are not too hard. If you have comments, contact us.
</p>
<a name="Submitting-patches"></a><a name="Submitting-patches-1"></a>
<h3 class="section">1.5 Submitting patches<span class="pull-right"><a class="anchor hidden-xs" href="#Submitting-patches-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Submitting-patches-1" aria-hidden="true">TOC</a></span></h3>
<p>First, read the <a href="#Coding-Rules">Coding Rules</a> above if you did not yet, in particular
the rules regarding patch submission.
</p>
<p>When you submit your patch, please use <code>git format-patch</code> or
<code>git send-email</code>. We cannot read other diffs :-)
</p>
<p>Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
</p>
<p>Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
</p>
<p>Run the <a href="#Regression-tests">Regression tests</a> before submitting a patch in order to verify
it does not cause unexpected problems.
</p>
<p>It also helps quite a bit if you tell us what the patch does (for example
&rsquo;replaces lrint by lrintf&rsquo;), and why (for example &rsquo;*BSD isn&rsquo;t C99 compliant
and has no lrint()&rsquo;)
</p>
<p>Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
</p>
<p>Patches should be posted to the
<a href="http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel">ffmpeg-devel</a>
mailing list. Use <code>git send-email</code> when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission.
</p>
<p>Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
</p>
<p>Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
</p>
<a name="New-codecs-or-formats-checklist"></a>
<h3 class="section">1.6 New codecs or formats checklist<span class="pull-right"><a class="anchor hidden-xs" href="#New-codecs-or-formats-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-New-codecs-or-formats-checklist" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Did you use av_cold for codec initialization and close functions?
</li><li> Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
</li><li> Did you bump the minor version number (and reset the micro version
number) in <samp>libavcodec/version.h</samp> or <samp>libavformat/version.h</samp>?
</li><li> Did you register it in <samp>allcodecs.c</samp> or <samp>allformats.c</samp>?
</li><li> Did you add the AVCodecID to <samp>avcodec.h</samp>?
When adding new codec IDs, also add an entry to the codec descriptor
list in <samp>libavcodec/codec_desc.c</samp>.
</li><li> If it has a FourCC, did you add it to <samp>libavformat/riff.c</samp>,
even if it is only a decoder?
</li><li> Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you&rsquo;re just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
</li><li> Did you add an entry to the table of supported formats or codecs in
<samp>doc/general.texi</samp>?
</li><li> Did you add an entry in the Changelog?
</li><li> If it depends on a parser or a library, did you add that dependency in
configure?
</li><li> Did you <code>git add</code> the appropriate files before committing?
</li><li> Did you make sure it compiles standalone, i.e. with
<code>configure --disable-everything --enable-decoder=foo</code>
(or <code>--enable-demuxer</code> or whatever your component is)?
</li></ol>
<a name="patch-submission-checklist"></a>
<h3 class="section">1.7 patch submission checklist<span class="pull-right"><a class="anchor hidden-xs" href="#patch-submission-checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-patch-submission-checklist" aria-hidden="true">TOC</a></span></h3>
<ol>
<li> Does <code>make fate</code> pass with the patch applied?
</li><li> Was the patch generated with git format-patch or send-email?
</li><li> Did you sign off your patch? (git commit -s)
See <a href="http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches">http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches</a> for the meaning
of sign off.
</li><li> Did you provide a clear git commit log message?
</li><li> Is the patch against latest FFmpeg git master branch?
</li><li> Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
</li><li> Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
</li><li> If the change is to speed critical code, did you benchmark it?
</li><li> If you did any benchmarks, did you provide them in the mail?
</li><li> Have you checked that the patch does not introduce buffer overflows or
other security issues?
</li><li> Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
<a href="http://caca.zoy.org/wiki/zzuf">zzuf</a>. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
</li><li> Does the patch not mix functional and cosmetic changes?
</li><li> Did you add tabs or trailing whitespace to the code? Both are forbidden.
</li><li> Is the patch attached to the email you send?
</li><li> Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
</li><li> If the patch fixes a bug, did you provide a verbose analysis of the bug?
</li><li> If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples &gt;100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
</li><li> Did you provide a verbose summary about what the patch does change?
</li><li> Did you provide a verbose explanation why it changes things like it does?
</li><li> Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
</li><li> Did you provide an example so we can verify the new feature added by the
patch easily?
</li><li> If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
</li><li> You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
</li><li> Lines with similar content should be aligned vertically when doing so
improves readability.
</li><li> Consider to add a regression test for your code.
</li><li> If you added YASM code please check that things still work with &ndash;disable-yasm
</li><li> Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like <code>av_malloc()</code>
are notoriously left unchecked, which is a serious problem.
</li><li> Test your code with valgrind and or Address Sanitizer to ensure it&rsquo;s free
of leaks, out of array accesses, etc.
</li></ol>
<a name="Patch-review-process"></a>
<h3 class="section">1.8 Patch review process<span class="pull-right"><a class="anchor hidden-xs" href="#Patch-review-process" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Patch-review-process" aria-hidden="true">TOC</a></span></h3>
<p>All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
</p>
<p>We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
</p>
<p>If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
</p>
<p>When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
</p>
<a name="Regression-tests"></a><a name="Regression-tests-1"></a>
<h3 class="section">1.9 Regression tests<span class="pull-right"><a class="anchor hidden-xs" href="#Regression-tests-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Regression-tests-1" aria-hidden="true">TOC</a></span></h3>
<p>Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
</p>
<p>Running &rsquo;make fate&rsquo; accomplishes this, please see <a href="fate.html">fate.html</a> for details.
</p>
<p>[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
</p>
<a name="Adding-files-to-the-fate_002dsuite-dataset"></a>
<h4 class="subsection">1.9.1 Adding files to the fate-suite dataset<span class="pull-right"><a class="anchor hidden-xs" href="#Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Adding-files-to-the-fate_002dsuite-dataset" aria-hidden="true">TOC</a></span></h4>
<p>When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
</p>
<a name="Visualizing-Test-Coverage"></a>
<h4 class="subsection">1.9.2 Visualizing Test Coverage<span class="pull-right"><a class="anchor hidden-xs" href="#Visualizing-Test-Coverage" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Visualizing-Test-Coverage" aria-hidden="true">TOC</a></span></h4>
<p>The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools <code>gcov</code>/<code>lcov</code>. This involves
the following steps:
</p>
<ol>
<li> Configure to compile with instrumentation enabled:
<code>configure --toolchain=gcov</code>.
</li><li> Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
</li><li> Run <code>make lcov</code> to generate coverage data in HTML format.
</li><li> View <code>lcov/index.html</code> in your preferred HTML viewer.
</li></ol>
<p>You can use the command <code>make lcov-reset</code> to reset the coverage
measurements. You will need to rerun <code>make lcov</code> after running a
new test.
</p>
<a name="Using-Valgrind"></a>
<h4 class="subsection">1.9.3 Using Valgrind<span class="pull-right"><a class="anchor hidden-xs" href="#Using-Valgrind" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Using-Valgrind" aria-hidden="true">TOC</a></span></h4>
<p>The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
<code>--toolchain=valgrind-memcheck</code> or <code>--toolchain=valgrind-massif</code>
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the <strong>memcheck</strong> or the
<strong>massif</strong> tool of the valgrind suite.
</p>
<p>In case you need finer control over how valgrind is invoked, use the
<code>--target-exec='valgrind &lt;your_custom_valgrind_options&gt;</code> option in
your configure line instead.
</p>
<a name="Release-process"></a><a name="Release-process-1"></a>
<h3 class="section">1.10 Release process<span class="pull-right"><a class="anchor hidden-xs" href="#Release-process-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-process-1" aria-hidden="true">TOC</a></span></h3>
<p>FFmpeg maintains a set of <strong>release branches</strong>, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a <strong>release
manager</strong> prepares, tests and publishes tarballs on the
<a href="http://ffmpeg.org">http://ffmpeg.org</a> website.
</p>
<p>There are two kinds of releases:
</p>
<ol>
<li> <strong>Major releases</strong> always include the latest and greatest
features and functionality.
</li><li> <strong>Point releases</strong> are cut from <strong>release</strong> branches,
which are named <code>release/X</code>, with <code>X</code> being the release
version number.
</li></ol>
<p>Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been <strong>compiled</strong> against
previous versions of <strong>the same release series</strong> in any case!
</p>
<p>However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the <strong>ffmpeg-devel</strong> mailing list in time to allow forward planning.
</p>
<a name="Criteria-for-Point-Releases"></a><a name="Criteria-for-Point-Releases-1"></a>
<h4 class="subsection">1.10.1 Criteria for Point Releases<span class="pull-right"><a class="anchor hidden-xs" href="#Criteria-for-Point-Releases-1" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Criteria-for-Point-Releases-1" aria-hidden="true">TOC</a></span></h4>
<p>Changes that match the following criteria are valid candidates for
inclusion into a point release:
</p>
<ol>
<li> Fixes a security issue, preferably identified by a <strong>CVE
number</strong> issued by <a href="http://cve.mitre.org/">http://cve.mitre.org/</a>.
</li><li> Fixes a documented bug in <a href="https://trac.ffmpeg.org">https://trac.ffmpeg.org</a>.
</li><li> Improves the included documentation.
</li><li> Retains both source code and binary compatibility with previous
point releases of the same release branch.
</li></ol>
<p>The order for checking the rules is (1 OR 2 OR 3) AND 4.
</p>
<a name="Release-Checklist"></a>
<h4 class="subsection">1.10.2 Release Checklist<span class="pull-right"><a class="anchor hidden-xs" href="#Release-Checklist" aria-hidden="true">#</a> <a class="anchor hidden-xs"href="#toc-Release-Checklist" aria-hidden="true">TOC</a></span></h4>
<p>The release process involves the following steps:
</p>
<ol>
<li> Ensure that the <samp>RELEASE</samp> file contains the version number for
the upcoming release.
</li><li> Add the release at <a href="https://trac.ffmpeg.org/admin/ticket/versions">https://trac.ffmpeg.org/admin/ticket/versions</a>.
</li><li> Announce the intent to do a release to the mailing list.
</li><li> Make sure all relevant security fixes have been backported. See
<a href="https://ffmpeg.org/security.html">https://ffmpeg.org/security.html</a>.
</li><li> Ensure that the FATE regression suite still passes in the release
branch on at least <strong>i386</strong> and <strong>amd64</strong>
(cf. <a href="#Regression-tests">Regression tests</a>).
</li><li> Prepare the release tarballs in <code>bz2</code> and <code>gz</code> formats, and
supplementing files that contain <code>gpg</code> signatures
</li><li> Publish the tarballs at <a href="http://ffmpeg.org/releases">http://ffmpeg.org/releases</a>. Create and
push an annotated tag in the form <code>nX</code>, with <code>X</code>
containing the version number.
</li><li> Propose and send a patch to the <strong>ffmpeg-devel</strong> mailing list
with a news entry for the website.
</li><li> Publish the news entry.
</li><li> Send announcement to the mailing list.
</li></ol>
<p style="font-size: small;">
This document was generated on <em>January 14, 2015</em> using <a href="http://www.gnu.org/software/texinfo/"><em>makeinfo</em></a>.
</p>
</div>
</body>
</html>

44
Externals/ffmpeg/shared/doc/examples/Makefile vendored
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@ -1,44 +0,0 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_reading \
decoding_encoding \
demuxing_decoding \
extract_mvs \
filtering_video \
filtering_audio \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

23
Externals/ffmpeg/shared/doc/examples/README vendored
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@ -1,23 +0,0 @@
FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then just run "make examples".
This will build the examples using the FFmpeg build system. You can clean those
examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make.

134
Externals/ffmpeg/shared/doc/examples/avio_reading.c vendored
View File

@ -1,134 +0,0 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_filename = argv[1];
/* register codecs and formats and other lavf/lavc components*/
av_register_all();
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx) {
av_freep(&avio_ctx->buffer);
av_freep(&avio_ctx);
}
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

665
Externals/ffmpeg/shared/doc/examples/decoding_encoding.c vendored
View File

@ -1,665 +0,0 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavcodec API use example.
*
* @example decoding_encoding.c
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
if (buffer_size < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
exit(1);
}
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
fclose(f);
av_freep(&samples);
av_frame_free(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int i, ch;
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_get_bytes_per_sample(c->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i=0; i<decoded_frame->nb_samples; i++)
for (ch=0; ch<c->channels; ch++)
fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&decoded_frame);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1,25};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
c->pix_fmt, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
fflush(stdout);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
av_frame_free(&frame);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for (;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
exit(1);
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
fclose(f);
avcodec_close(c);
av_free(c);
av_frame_free(&frame);
printf("\n");
}
int main(int argc, char **argv)
{
const char *output_type;
/* register all the codecs */
avcodec_register_all();
if (argc < 2) {
printf("usage: %s output_type\n"
"API example program to decode/encode a media stream with libavcodec.\n"
"This program generates a synthetic stream and encodes it to a file\n"
"named test.h264, test.mp2 or test.mpg depending on output_type.\n"
"The encoded stream is then decoded and written to a raw data output.\n"
"output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
output_type = argv[1];
if (!strcmp(output_type, "h264")) {
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.pcm", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");
} else {
fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
output_type);
return 1;
}
return 0;
}

386
Externals/ffmpeg/shared/doc/examples/demuxing_decoding.c vendored
View File

@ -1,386 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
/* The different ways of decoding and managing data memory. You are not
* supposed to support all the modes in your application but pick the one most
* appropriate to your needs. Look for the use of api_mode in this example to
* see what are the differences of API usage between them */
enum {
API_MODE_OLD = 0, /* old method, deprecated */
API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */
API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
};
static int api_mode = API_MODE_OLD;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
return ret;
}
/* Some audio decoders decode only part of the packet, and have to be
* called again with the remainder of the packet data.
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
* Also, some decoders might over-read the packet. */
decoded = FFMIN(ret, pkt.size);
if (*got_frame) {
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
}
}
/* If we use the new API with reference counting, we own the data and need
* to de-reference it when we don't use it anymore */
if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
av_frame_unref(frame);
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the decoders, with or without reference counting */
if (api_mode == API_MODE_NEW_API_REF_COUNT)
av_dict_set(&opts, "refcounted_frames", "1", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4 && argc != 5) {
fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
"input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n\n"
"If the -refcount option is specified, the program use the\n"
"reference counting frame system which allows keeping a copy of\n"
"the data for longer than one decode call. If unset, it's using\n"
"the classic old method.\n"
"\n", argv[0]);
exit(1);
}
if (argc == 5) {
const char *mode = argv[1] + strlen("-refcount=");
if (!strcmp(mode, "old")) api_mode = API_MODE_OLD;
else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT;
else {
fprintf(stderr, "unknow mode '%s'\n", mode);
exit(1);
}
argv++;
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
ret = av_image_alloc(video_dst_data, video_dst_linesize,
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
/* When using the new API, you need to use the libavutil/frame.h API, while
* the classic frame management is available in libavcodec */
if (api_mode == API_MODE_OLD)
frame = avcodec_alloc_frame();
else
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_close(video_dec_ctx);
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
if (api_mode == API_MODE_OLD)
avcodec_free_frame(&frame);
else
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}

185
Externals/ffmpeg/shared/doc/examples/extract_mvs.c vendored
View File

@ -1,185 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
static const char *src_filename = NULL;
static int video_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int decoded = pkt.size;
*got_frame = 0;
if (pkt.stream_index == video_stream_idx) {
int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
return ret;
}
if (*got_frame) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
}
}
return decoded;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
src_filename = argv[1];
av_register_all();
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
}
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
AVPacket orig_pkt = pkt;
do {
ret = decode_packet(&got_frame, 0);
if (ret < 0)
break;
pkt.data += ret;
pkt.size -= ret;
} while (pkt.size > 0);
av_free_packet(&orig_pkt);
}
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
end:
avcodec_close(video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

365
Externals/ffmpeg/shared/doc/examples/filter_audio.c vendored
View File

@ -1,365 +0,0 @@
/*
* copyright (c) 2013 Andrew Kelley
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include <inttypes.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink)
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
AVFilter *abuffer;
AVFilterContext *volume_ctx;
AVFilter *volume;
AVFilterContext *aformat_ctx;
AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
static int get_input(AVFrame *frame, int frame_num)
{
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float*)frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
int main(int argc, char *argv[])
{
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
return 1;
}
duration = atof(argv[1]);
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
avfilter_register_all();
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}

280
Externals/ffmpeg/shared/doc/examples/filtering_audio.c vendored
View File

@ -1,280 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet0, packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
packet0.data = NULL;
packet.data = NULL;
while (1) {
if (!packet0.data) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
packet0 = packet;
}
if (packet.stream_index == audio_stream_index) {
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
packet.size -= ret;
packet.data += ret;
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
}
if (packet.size <= 0)
av_free_packet(&packet0);
} else {
/* discard non-wanted packets */
av_free_packet(&packet0);
}
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

262
Externals/ffmpeg/shared/doc/examples/filtering_video.c vendored
View File

@ -1,262 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
int got_frame;
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

56
Externals/ffmpeg/shared/doc/examples/metadata.c vendored
View File

@ -1,56 +0,0 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}

670
Externals/ffmpeg/shared/doc/examples/muxing.c vendored
View File

@ -1,670 +0,0 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, *codec);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = ost->st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->st->codec;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->st->codec;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->st->codec;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i, ret;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(pict);
if (ret < 0)
exit(1);
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->st->codec;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
c = ost->st->codec;
frame = get_video_frame(ost);
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* a hack to avoid data copy with some raw video muxers */
AVPacket pkt;
av_init_packet(&pkt);
if (!frame)
return 1;
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = ost->st->index;
pkt.data = (uint8_t *)frame;
pkt.size = sizeof(AVPicture);
pkt.pts = pkt.dts = frame->pts;
av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
AVPacket pkt = { 0 };
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_close(ost->st->codec);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
if (argc > 3 && !strcmp(argv[2], "-flags")) {
av_dict_set(&opt, argv[2]+1, argv[3], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}

165
Externals/ffmpeg/shared/doc/examples/remuxing.c vendored
View File

@ -1,165 +0,0 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
in_filename = argv[1];
out_filename = argv[2];
av_register_all();
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *in_stream = ifmt_ctx->streams[i];
AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
if (ret < 0) {
fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
goto end;
}
out_stream->codec->codec_tag = 0;
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_free_packet(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

214
Externals/ffmpeg/shared/doc/examples/resampling_audio.c vendored
View File

@ -1,214 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

140
Externals/ffmpeg/shared/doc/examples/scaling_video.c vendored
View File

@ -1,140 +0,0 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

755
Externals/ffmpeg/shared/doc/examples/transcode_aac.c vendored
View File

@ -1,755 +0,0 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 48000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
/**
* Convert an error code into a text message.
* @param error Error code to be converted
* @return Corresponding error text (not thread-safe)
*/
static const char *get_error_text(const int error)
{
static char error_buffer[255];
av_strerror(error, error_buffer, sizeof(error_buffer));
return error_buffer;
}
/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodec *input_codec;
int error;
/** Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
filename, get_error_text(error));
*input_format_context = NULL;
return error;
}
/** Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/** Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
get_error_text(error));
avformat_close_input(input_format_context);
return error;
}
/** Save the decoder context for easier access later. */
*input_codec_context = (*input_format_context)->streams[0]->codec;
return 0;
}
/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/** Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
filename, get_error_text(error));
return error;
}
/** Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/** Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/** Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
av_strlcpy((*output_format_context)->filename, filename,
sizeof((*output_format_context)->filename));
/** Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/** Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/** Save the encoder context for easiert access later. */
*output_codec_context = stream->codec;
/**
* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion.
*/
(*output_codec_context)->channels = OUTPUT_CHANNELS;
(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
(*output_codec_context)->sample_fmt = AV_SAMPLE_FMT_S16;
(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
/**
* Some container formats (like MP4) require global headers to be present
* Mark the encoder so that it behaves accordingly.
*/
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
(*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
/** Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
get_error_text(error));
goto cleanup;
}
return 0;
cleanup:
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/** Initialize one data packet for reading or writing. */
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/** Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/**
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/**
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/** Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int init_fifo(AVAudioFifo **fifo)
{
/** Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/** Write the header of the output file container. */
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/** Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/** Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/** If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
get_error_text(error));
return error;
}
}
/**
* Decode the audio frame stored in the temporary packet.
* The input audio stream decoder is used to do this.
* If we are at the end of the file, pass an empty packet to the decoder
* to flush it.
*/
if ((error = avcodec_decode_audio4(input_codec_context, frame,
data_present, &input_packet)) < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&input_packet);
return error;
}
/**
* If the decoder has not been flushed completely, we are not finished,
* so that this function has to be called again.
*/
if (*finished && *data_present)
*finished = 0;
av_free_packet(&input_packet);
return 0;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/**
* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/**
* Allocate memory for the samples of all channels in one consecutive
* block for convenience.
*/
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
get_error_text(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/** Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Add converted input audio samples to the FIFO buffer for later processing. */
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/**
* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples.
*/
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/** Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/** Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/** Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present;
int ret = AVERROR_EXIT;
/** Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/** Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/**
* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error.
*/
if (*finished && !data_present) {
ret = 0;
goto cleanup;
}
/** If there is decoded data, convert and store it */
if (data_present) {
/** Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/**
* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples.
*/
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/** Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/** Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/**
* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity.
*/
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/**
* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified.
*/
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
get_error_text(error));
av_frame_free(frame);
return error;
}
return 0;
}
/** Encode one frame worth of audio to the output file. */
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/** Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/**
* Encode the audio frame and store it in the temporary packet.
* The output audio stream encoder is used to do this.
*/
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
frame, data_present)) < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
/** Write one audio frame from the temporary packet to the output file. */
if (*data_present) {
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
get_error_text(error));
av_free_packet(&output_packet);
return error;
}
av_free_packet(&output_packet);
}
return 0;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/** Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/**
* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size
*/
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/** Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/**
* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily.
*/
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/** Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/** Write the trailer of the output file container. */
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
get_error_text(error));
return error;
}
return 0;
}
/** Convert an audio file to an AAC file in an MP4 container. */
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/** Register all codecs and formats so that they can be used. */
av_register_all();
/** Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/** Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/** Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/** Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo))
goto cleanup;
/** Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/**
* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither.
*/
while (1) {
/** Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/**
* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples.
*/
while (av_audio_fifo_size(fifo) < output_frame_size) {
/**
* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer.
*/
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/**
* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file.
*/
if (finished)
break;
}
/**
* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder.
*/
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/**
* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file.
*/
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/**
* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish.
*/
if (finished) {
int data_written;
/** Flush the encoder as it may have delayed frames. */
do {
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/** Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_close(output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_close(input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

583
Externals/ffmpeg/shared/doc/examples/transcoding.c vendored
View File

@ -1,583 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream;
AVCodecContext *codec_ctx;
stream = ifmt_ctx->streams[i];
codec_ctx = stream->codec;
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* Open decoder */
ret = avcodec_open2(codec_ctx,
avcodec_find_decoder(codec_ctx->codec_id), NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = in_stream->codec;
enc_ctx = out_stream->codec;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
enc_ctx->pix_fmt = encoder->pix_fmts[0];
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = dec_ctx->time_base;
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
ifmt_ctx->streams[i]->codec);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
return ret;
}
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
AVFilter *buffersrc = NULL;
AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
ofmt_ctx->streams[i]->codec, filter_spec);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codec->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
ofmt_ctx->streams[stream_index]->codec->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ifmt_ctx->streams[stream_index]->codec->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = av_frame_get_best_effort_timestamp(frame);
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_free_packet(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_free_packet(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_close(ifmt_ctx->streams[i]->codec);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
avcodec_close(ofmt_ctx->streams[i]->codec);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

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