Commit Graph

401 Commits

Author SHA1 Message Date
Ryan C. Gordon
254eb67775 windows: Don't let Visual Studio insert an implicit dependency on memset().
Fixes Bugzilla #4662.
2019-06-11 02:08:31 -04:00
Sam Lantinga
d8da33c03f Fixed bug 4662 - SDL failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode with MSVC on Windows
LinGao

We build SDL with Visual studio 2017 compiler on Windows Server 2016, but it failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode on latest default branch. And we found that it can be first reproduced on ca7283111ad0 changeset. Could you please help have a look about this issue? Thanks in advance!
2019-06-10 08:49:26 -07:00
Sam Lantinga
762b788f67 Cleanup on bug 3894 - Fuzzing crashes for SDL_LoadWAV
Simon Hug

Attached is a minor cleanup patch. It changes the option name of one hint to something better, puts one or two more checks in, and adds explicit casting where warnings could appear otherwise.

I hope the naming of the hints and their options is acceptable. It would be kind of awkward to change them after they get released with an official SDL version.
2019-06-09 12:46:10 -07:00
Sam Lantinga
a21b5b3018 Fixed build 2019-06-08 19:09:43 -07:00
Sam Lantinga
990e166a3b Fixed bug 3894 - Fuzzing crashes for SDL_LoadWAV
Simon Hug

I had a look at this and made some additions to SDL_wave.c.

The attached patch adds many checks and error messages. For some reason I also added A-law and ?-law decoders. Forgot exactly why... but hey, they're small.

The WAVE format is seriously underspecified (at least by the documents that are publicly available on the internet) and it's a shame Microsoft never put something better out there. The language used in them is so loose at times, it's not surprising the encoders and decoders behave very differently. The Windows Media Player doesn't even support MS ADPCM correctly.

The patch also adds some hints to make the decoder more strict at the cost of compatibility with weird WAVE files.

I still think it needs a bit of cleaning up (Not happy with the MultiplySize function. Don't like the name and other SDL code may want to use something like this too.) and some duplicated code may be folded together. It does work in this state and I have thrown all kinds of WAVE files at it. The AFL files also pass with it and some even play (obviously just noise). Crafty little fuzzer.

Any critique would be welcome. I have a fork of SDL with a audio-loadwav branch over here if someone wants to use the commenting feature of Bitbucket:

https://bitbucket.org/ChliHug/SDL

I also cobbled some Lua scripts together to create WAVE test files:

https://bitbucket.org/ChliHug/gendat
2019-06-08 19:02:42 -07:00
Sam Lantinga
31765242d6 Fixed bug 4294 - Audio: perform more validation on conversion request
janisozaur

There are many cases which are not able to be handled by SDL's audio conversion routines, including too low (negative) rate, too high rate (impossible to allocate).

This patch aims to report such issues early and handle others in a graceful manner. The "INT32_MAX / RESAMPLER_SAMPLES_PER_ZERO_CROSSING" value is the conservative approach in terms of what can _technically_ be supported, but its value is 4'194'303, or just shy of 4.2MHz. I highly doubt any sane person would use such rates, especially in SDL2, so I would like to drive this limit further down, but would need some assistance to do that, as doing so would have to introduce an arbitrary value. Are you OK with such approach? What would a good value be? Wikipedia (https://en.wikipedia.org/wiki/High-resolution_audio) lists 96kHz as the highest sampling rate in use, even if I quadruple it for a good measure, to 384kHz it's still an order of magnitude lower than 4MHz.
2019-06-08 18:22:18 -07:00
Sam Lantinga
3f19a6d5e8 CVE-2019-7578: Fix a buffer overread in InitIMA_ADPCM
If IMA ADPCM format chunk was too short, InitIMA_ADPCM() parsing it
could read past the end of chunk data. This patch fixes it.

CVE-2019-7578
https://bugzilla.libsdl.org/show_bug.cgi?id=4494

Signed-off-by: Petr P?sa? <ppisar@redhat.com>
2019-06-08 18:07:58 -07:00
Sam Lantinga
8a37848de9 Fixed bug 4605 - WASAPI_WaitDevice hang
Matt Brocklehurst

We've noticed that if you are playing audio on Windows via the WASAPI interface and you unplug and reconnect the device a few times the program hangs.

We've debugged the problem down to

static void
WASAPI_WaitDevice(_THIS)
{

   ... snip ...
 if (WaitForSingleObjectEx(this->hidden->event, INFINITE, FALSE) == WAIT_OBJECT_0) {
   ... snip ...
}

This WaitForSingleObjectEx does not havbe a time out defined, so it hangs there forever.

Our suggested fix we found was to include a time out of say 200mSec

We have done quite a bit of testing with this fix in place on various hardware configurations and it seems to have resolved the issue.
2019-06-08 13:41:46 -07:00
Sam Lantinga
15bae953b1 Fixed bug 4642 - Rework SDL_netbsdaudio to improve performance
Nia Alarie

The NetBSD audio driver has a few problems. Lots of obsolete code, and extremely bad performance and stuttering.

I have a patch in NetBSD's package system to improve it. This is my attempt to upstream it.

The changes include:

* Removing references to defines which are never used.
* Using the correct structures for playback and recording, previously they were the wrong way around.
* Using the correct types ('struct audio_prinfo' in contrast to 'audio_prinfo')
* Removing the use of non-blocking I/O, as suggested in #3177.
* Removing workarounds for driver bugs on systems that don't exist or use this driver any more.
* Removing all usage of SDL_Delay(1)
* Removing pointless use of AUDIO_INITINFO and tests that expect AUDIO_SETINFO to fail when it can't.

These changes bring its performance in line with the DSP audio driver.
2019-06-08 13:03:36 -07:00
Sam Lantinga
03cf24162f OpenSL ES audio cleanup and added a note with low latency audio discussion 2019-06-08 10:21:38 -07:00
Sam Lantinga
166d15fd75 Fixed surround sound channel setup for Android OpenSL ES audio driver 2019-06-07 15:09:15 -07:00
Sam Lantinga
723d014336 Fixed bug 4171 - SDL_GetQueuedAudioSize is broken with WASAPI
Cameron Gutman

I was trying to use SDL_GetQueuedAudioSize() to ensure my audio latency didn't get too high while streaming data in from the network. If I get more than N frames of audio queued, I know that the network is giving me more data than I can play and I need to drop some to keep latency low.

This doesn't work well on WASAPI out of the box, due to the addition of GetPendingBytes() to the amount of queued data. As a terrible hack, I loop 100 times calling SDL_Delay(10) and SDL_GetQueuedAudioSize() before I ever call SDL_QueueAudio() to get a "baseline" amount that I then subtract from SDL_GetQueuedAudioSize() later. However, because this value isn't actually a constant, this hack can cause SDL_GetQueuedAudioSize() - baselineSize to be < 0. This means I have no accurate way of determining how much data is actually queued in SDL's audio buffer queue.

The SDL_GetQueuedAudioSize() documentation says: "This is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware." Yet, SDL_GetQueuedAudioSize() returns > 0 value when SDL_QueueAudio() has never been called.

Based on that documentation, I believe the current behavior contradicts the documented behavior of this function and should be changed in line with Boris's patch.

I understand that exposing the IAudioClient::GetCurrentPadding() value is useful, but a solution there needs to take into account what of that data is silence inserted by SDL and what is actual data queued by the user with SDL_QueueAudio(). Until that happens, I think the best approach is to remove the GetPendingBytes() call until SDL is able to keep track of queued data to make sense of it. This would make SDL_GetQueuedAudioSize() possible to use accurately with WASAPI.
2019-06-04 17:32:15 -07:00
Sam Lantinga
f3e76ea1d0 Use the OpenSL ES audio driver by default on Android, as it has the lowest latency. 2019-05-23 13:47:30 -07:00
Sam Lantinga
02f9667a08 Fixed static and buzzing when trying to use floating point audio on the OpenSL ES audio driver. 2019-05-23 13:47:27 -07:00
Sam Lantinga
abcfe80480 [SDL] iOS fix bug with audio interrupted by a phone call not restoring. 2019-05-14 14:20:54 -07:00
Ryan C. Gordon
2fbfe8b912 coreaudio: Set audio callback thread priority.
Fixes Bugzilla #4155.
2019-03-25 12:59:30 -04:00
Ryan C. Gordon
6a3356ab3f Backed out changeset cec31de4e126
This was meant to migrate CoreAudio onto the same SDL_RunAudio() path that
most other audio drivers are on, but it introduced a bug because it doesn't
deal with dropped audio buffers...and fixing that properly just introduces
latency.

I might revisit this later, perhaps by reworking SDL_RunAudio to allow for
this sort of API better, or redesigning the whole subsystem or something, I
don't know. I'm not super-thrilled that this has to exist outside of the usual
codepaths, though.

Fixes Bugzilla #4481.
2019-03-25 12:24:38 -04:00
Sam Lantinga
35255342cd Fixed bug 4525 - Fix crash in ALSA_HotplugThread caused by bad return value check
Anthony Pesch

Fix snd_device_name_hint return value check

According to the ALSA documentation, snd_device_name_hint returns 0 on
success, otherwise a negative error code. The code previously only
considered -1 to be an error, which let other error codes through
resulting in a segfault when hints (which was NULL) was dereferenced
2019-03-16 18:48:21 -07:00
Sylvain Becker
03cbac4040 Android/openslES: fix warnings, comment out un-used interface 2019-02-05 15:14:15 +01:00
Sylvain Becker
614c8aea20 Android/openslES: set number of buffers of DATALOCATOR to internal NUM_BUFFER
If we increase NUM_BUFFER, Enqueue won't fail with SL_RESULT_BUFFER_INSUFFICIENT
2019-02-05 15:09:41 +01:00
Sylvain Becker
bf823bf2dc Android/openslES: prevent to run out of buffers if Enqueue() fails. 2019-02-05 15:05:32 +01:00
Alon Zakai
3b4e369365 Emscripten: No need for Runtime. for dynCalls 2019-01-29 12:21:22 +00:00
Alon Zakai
53ead95e1d Emscripten: Avoid SDL2 in JS global scope
After this fix, closure works with the LLVM wasm backend on SDL2.
2019-01-29 12:19:36 +00:00
Sylvain Becker
1b24b2eca5 Android/openslES: fix Pause/ResumeDevices when openslES is not used 2019-01-14 22:56:57 +01:00
Sylvain Becker
647b1f6a6d Android/openslES: check for non NULL variable, some intialization.
use the previous naming
2019-01-14 14:36:13 +01:00
Sylvain Becker
7b1cc441dd Android/openslES: start playing, after creating ressources 2019-01-14 14:31:06 +01:00
Sylvain Becker
955d87894b Android/openslES: set audio in paused/resumed state for Android event loop
And also in "stopped" state before closing the device.
2019-01-14 12:33:29 +01:00
Sylvain Becker
59c8c7b684 Android/openslES: move a few static variables to SDL_PrivateAudioData structure 2019-01-14 10:58:57 +01:00
Sylvain Becker
5aeeaaab70 Android/openslES: register and use CloseDevice function. 2019-01-14 10:16:26 +01:00
Sylvain Becker
365fd9c602 Android/openslES: some space and indentation to match SDL conventions 2019-01-14 10:04:54 +01:00
Sam Lantinga
7dc92a7669 Initial Android OpenSL ES implementation, contributed by ANTA 2019-01-12 12:18:44 -08:00
Sylvain Becker
d23c2f07e3 Fixed bug 3930 - Android, set thread priorities and names
SDLActivity thread priority is unchanged, by default -10 (THREAD_PRIORITY_VIDEO).

SDLAudio thread priority was -4 (SDL_SetThreadPriority was ignored) and is now -16 (THREAD_PRIORITY_AUDIO).

SDLThread thread priority was 0 (THREAD_PRIORITY_DEFAULT) and is -4 (THREAD_PRIORITY_DISPLAY).
2019-01-10 18:05:56 +01:00
Sam Lantinga
5e13087b0f Updated copyright for 2019 2019-01-04 22:01:14 -08:00
Sylvain Becker
aea7e56a24 android: use __ARM_NEON instead of __ARM_NEON__ to include <arm_neon.h>
Only __ARM_NEON is defined with Android NDK and arm64-v8a
Tested on ndk-r18, ndk-r13 and also Xcode.
(Visual Studio needs a different fix).

Fixes Bugzilla #4409.
2018-12-04 12:34:45 +01:00
Sylvain Beucler
1f6bd95110 Emscripten: make CloseAudio actually close audio
cf. https://bugzilla.libsdl.org/show_bug.cgi?id=4176
2018-11-15 18:22:30 +00:00
Micha? Janiszewski
91820998fc Add and update include guards
Include guards in most changed files were missing, I added them keeping
the same style as other SDL files. In some cases I moved the include
guards around to be the first thing the header has to take advantage of
any possible improvements compiler may have for inclusion guards.
2018-10-28 21:36:48 +01:00
Ryan C. Gordon
4a50a04213 wasapi/win32: Sort initial device lists by device GUID.
This makes an unchanged set of hardware always report devices in the same
order on each run.
2018-10-21 22:40:17 -04:00
Ryan C. Gordon
04cbf13261 audio: All device names reported by SDL must be unique.
This means that if you have two devices named "Soundblaster Pro" in your
machine, one will be reported as "Soundblaster Pro" and the other as
"Soundblaster Pro (2)".

This makes it so you can't into a position where one of your devices can't
be opened because another is sitting on the same name.
2018-10-10 15:20:56 -04:00
Ryan C. Gordon
0378529e1e audio: clean_out_device_list() already sets this flag to false for us. 2018-10-10 14:55:24 -04:00
Sam Lantinga
f5a21ebf0c Added support for surround sound and float audio on Android 2018-10-09 20:12:43 -07:00
Sam Lantinga
b251876126 commit c6b28f46b8116552ec2b38d1d3c8535df28ba7a1
Author: Anthony Pesch <inolen@gmail.com>
Date:   Fri May 4 20:21:21 2018 -0400

    Added SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag enabling users of SDL_OpenAudioDevice to get
    the sample size of the actual hardware buffer vs having a stream created to handle the
    delta
2018-10-01 09:47:10 -07:00
Ryan C. Gordon
56ec349d2a audio: disable NEON converters for now.
To be revisited after 2.0.9 ships!

(doesn't fix Bugzilla #4186, but stops the regression for the time being.)
2018-09-29 16:48:15 -04:00
Ethan Lee
7f9854b9b2 WinRT: Wait until audio device activation is complete and PrepDevice during OpenAudio 2018-09-25 01:45:12 -04:00
Sam Lantinga
5febdfcece Fixed whitespace 2018-09-24 11:49:25 -07:00
Ryan C. Gordon
623a6defd3 alsa: optionally run entire pipeline non-blocking. 2018-08-07 16:49:18 -04:00
Ryan C. Gordon
56f44cfa0f audio: Deal with device shutdown more carefully.
This would cause problems in various ways, but specifically triggers an
assert when you close a WASAPI capture device in an app running over RDP.

Related to (but not the actual bug) in Bugzilla #3924.
2018-08-07 13:04:15 -04:00
Wohlstand
ff8c62f227 Fixed bug 4210 - SSE2-based converter makes junk result of S32 -> Float
At the HG state abdd17144682, 64-bit assemblies are using SSE2-based resampler, produces junk sound when converting the S32 -> Float32 -> S16 chain. The `NEED_SCALAR_CONVERTER_FALLBACKS` thing works perfectly.

If I will find a reason that caused this mistake, I'll send a patch by myself.
2018-07-02 03:53:57 +03:00
Ryan C. Gordon
4773690d0f Deal with possible malloc(0) calls, as pointed out by static analysis. 2018-06-25 12:55:23 -04:00
Anthony Pesch
c591429542 alsa: avoid hardware parameters with an excessive number of periods.
The previous code attempted to use set_buffer_size / set_period_size
discretely, favoring the parameters which generated a buffer size that was
exactly 2x the requested buffer size. This solution ultimately prioritizes
only the buffer size, which comes at a large performance cost on some machines
where this results in an excessive number of periods. In my case, for a 4096
sample buffer, this configured the device to use 37 periods with a period size
of 221 samples and a buffer size of 8192 samples. With 37 periods, the SDL
Audio thread was consuming 25% of the CPU.

This code has been refactored to use set_period_size and set_buffer_size
together. set_period_size is called first to attempt to set the period to
exactly match the requested buffer size, and set_buffer_size is called second
to further refine the parameters to attempt to use only 2 periods. The
fundamental change here is that the period size / count won't go to extreme
values if the buffer size can't be exactly matched, the buffer size should
instead just increase to the next closest multiple of the target period size
that is supported. After changing this, for a 4096 sample buffer, the device
is configured to use 3 periods with a period size of 4096 samples and a buffer
size of 12288 samples. With only 3 periods, the SDL Audio thread doesn't even
show up when profiling.

Fixes Bugzilla #4156.
2018-05-04 21:21:32 -04:00
Sam Lantinga
1d25135b71 Fixed bug 4184 - jack audio driver fails in presence of midi ports
Martin ?irokov

Launching an SDL application with SDL_AUDIODRIVER=jack, and then calling SDL_OpenAudioDevice() with whatever parameters fails with an error like this one:

SDL_OpenAudioDevice: Couldn't connect JACK ports: SDL:sdl_jack_output_0 => system:midi_playback_1

This happens because JACK_OpenDevice in src/audio/jack/SDL_jackaudio.c blindly tries to connect to all input ports without checking whether they are for audio or midi.

The fix is to check port types and ignore all non audio ports. Also I removed devports field from struct SDL_PrivateAudioData, because it's never really used and removing unused ports from it would be PITA.
2018-06-01 19:43:53 -07:00
Sam Lantinga
8325df25aa Fixed bug 4169 - Crash due to audio session observer race condition
Jona

The following explains why this bug was happening:
This crash was caused because the audio session was being set as active [session setActive:YES error:&err] when the audio device was actually being CLOSED. Certain cases the audio session being set to active would fail and the method would return right away. Because of the way the error was handled we never removed the SDLInterruptionListener thus leaking it. Later when an interruption was received the THIS_ object would contain a pointer to an already released device causing the crash.

The fix:
When only one device remained open and it was being closed we needed to set the audio session as NOT active and completely ignore the returned error to successfully release the SDLInterruptionListener. I think the user assumed that the open_playback_devices and open_capture_devices would equal 0 when all of them where closed but the truth is that at the end of the closing process that the open devices count is decremented.
2018-05-24 07:30:24 -07:00
Ryan C. Gordon
101544d6f0 audio: Needed to fix two more instances for Visual Studio. 2018-05-21 12:05:17 -04:00
Ryan C. Gordon
49881861b1 audio: Patched to compile on Visual Studio.
(It gets upset at the -2147483648, thinking this should be an unsigned value
because 2147483648 is too large for an int32, so the negative sign upsets the
compiler.)
2018-05-21 11:54:09 -04:00
Ryan C. Gordon
b7e88aaae0 audio: Added ARM NEON versions of audio converters.
These are _much_ faster than the scalar equivalents on the Raspberry Pi that
I tested on. Often 3x to 4x as fast!
2018-05-16 02:03:06 -04:00
Ryan C. Gordon
cb0e614fb1 audio: SSE2 float-to-int converters should clamp input.
The scalar versions already do this.
2018-05-15 02:29:35 -04:00
Ryan C. Gordon
a07e5815a5 audio: Fix range on float-to-int data clamping.
I can't tell if there was a good reason for this or it was just me getting
numbers wrong due to exhaustion.
2018-05-15 01:40:05 -04:00
Ryan C. Gordon
7832cb652e audio: float to int converters should clamp inclusively.
If we have to test if a sample is > 1.0f anyhow, we might as well use this
to avoid the unnecessary multiplication when it's == 1.0f, too. (etc).
2018-05-15 01:35:53 -04:00
Ryan C. Gordon
e2ec1eb12e audio: converting int32 to/from float shouldn't use doubles.
The concern is that a massive int sample, like 0x7FFFFFFF, won't fit in a
float32, which doesn't have enough bits to hold a whole number this large,
just to divide it to get a value between 0 and 1.
Previously we would convert to double, to get more bits, do the division, and
cast back to a float, but this is expensive.

Casting to double is more accurate, but it's 2x to 3x slower. Shifting out
the least significant byte of an int32, so it'll definitely fit in a float,
and dividing by 0x7FFFFF is still accurate to about 5 decimal places, and the
difference doesn't appear to be perceptable.
2018-05-15 01:04:11 -04:00
Sam Lantinga
f521b22eb5 Added SDL_THREAD_PRIORITY_TIME_CRITICAL 2018-04-23 22:07:56 -07:00
Ryan C. Gordon
dc8b55e50b coreaudio: Use the standard SDL audio thread instead of spinning a new one.
Fixes corner cases, like the audio callback not firing if the device is
disconnected, etc.
2018-04-16 02:11:09 -04:00
Sam Lantinga
99a0c0f0e2 Fixed MinGW-w64 build 2018-02-24 08:23:44 -08:00
Ryan C. Gordon
c7e4366530 wasapi: let Windows do the resampling for us if possible. 2018-02-21 21:34:06 -05:00
Ryan C. Gordon
7e1fa0ce53 wasapi: fixed typo in an assert message. 2018-02-21 21:34:35 -05:00
Ryan C. Gordon
97494f5374 pulseaudio: Just read/dump captured data in FlushCapture.
Apparently pa_stream_flush() doesn't work as expected:

https://lists.freedesktop.org/archives/pulseaudio-discuss/2012-April/013328.html

Fixes Bugzilla #4087.
2018-02-17 18:30:21 -05:00
sezero
ba0ecc6712 fix building SDL_audiotypecvt.c with gcc < 4.0 2018-02-12 10:47:00 +03:00
sezero
40b27fd51b revert the recent typecast assignment changes (see bug #4079)
also change the void* typedefs for the two vulkan function
pointers added in vulkan_internal.h  into generic function
pointer typedefs.
2018-02-12 17:00:00 +03:00
Sam Lantinga
90e72bf4e2 Fixed ISO C99 compatibility
SDL now builds with gcc 7.2 with the following command line options:
-Wall -pedantic-errors -Wno-deprecated-declarations -Wno-overlength-strings --std=c99
2018-01-30 18:08:34 -08:00
Ryan C. Gordon
488824017a wasapi: Fixed some compiler warnings. 2018-01-22 09:36:40 -05:00
Sam Lantinga
e3cc5b2c6b Updated copyright for 2018 2018-01-03 10:03:25 -08:00
Ryan C. Gordon
77bb49b7a7 wasapi: Patched to compile on non-UWP WinRT builds. 2017-12-31 03:34:16 -05:00
Ryan C. Gordon
ab4695f48f wasapi: switched to event-driven interface.
This reduces latency and improves battery life.
2017-12-13 14:35:55 -05:00
Ryan C. Gordon
351d6d4784 audio: Port WASAPI to WinRT, remove XAudio2 backend.
XAudio2 doesn't have capture support, so WASAPI was to replace it; the holdout
was WinRT, which still needed it as its primary audio target until the WASAPI
code code be made to work.

The support matrix now looks like:

WinXP: directsound by default, winmm as a fallback for buggy drivers.
Vista+: WASAPI (directsound and winmm as fallbacks for debugging).
WinRT: WASAPI
2017-12-06 12:24:32 -05:00
Sam Lantinga
e830ef3458 Fixed typo converting 4 channel audio to 2 channel 2017-10-20 16:53:42 -07:00
Sam Lantinga
9a291c1e59 Added a note about adjusting channel weights when converting to fewer channels 2017-10-20 14:51:22 -07:00
Ryan C. Gordon
729329068b audio: Added SDL_AudioStreamFlush(). 2017-10-19 18:05:42 -04:00
Ryan C. Gordon
e98920f5f3 Check correct variable for malloc() results. 2017-10-18 23:49:46 -04:00
Sam Lantinga
afefcbfeba Fixed bug 3876 - Resampling of certain sounds adds heavy distortion
Simon Hug

Patch that adds [-1, 1] clamping to the scalar audio type conversions.

This may come from the SDL_Convert_F32_to_X_Scalar functions. They don't clamp the float value to [-1, 1] and when they cast it to the target integer it may be too large or too small for the type and get truncated, causing horrible noise.

The attached patch throws clamping in, but I don't know if that's the preferred way to fix this. For x86 (without SSE) the compiler (I tested MSVC) seems to throw a horrible amount of x87 code in it. It's a bit better with SSE, but probably still quite the performance hit. And SSE2 uses a branchless approach with maxss and minss.
2017-10-18 19:30:47 -07:00
Sam Lantinga
653ab5d9c4 Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling 2017-10-18 19:26:36 -07:00
Sam Lantinga
80f8464d97 Added audio stream conversion functions:
SDL_NewAudioStream
    SDL_AudioStreamPut
    SDL_AudioStreamGet
    SDL_AudioStreamAvailable
    SDL_AudioStreamClear
    SDL_FreeAudioStream
2017-10-18 15:54:05 -07:00
Ryan C. Gordon
fa15674134 coreaudio: changed device close procedure to prevent long hangs in some cases.
The audioqueue thread needs to keep running, and processing the CFRunLoop
until the AudioQueue is disposed of, otherwise CoreAudio will hang waiting for
final data to feed the device.

At least, I think this is how it all works. It definitely fixes the bug here!

Since AudioQueueDispose() calls AudioQueueStop() internally, there's no need
for our thread to handle this, either, which is good because the AudioQueue
would be disposed by this point. So now the AudioQueue is disposed first, and
then our thread is joined, and everything works out okay.

Just in case, we mark the device "paused" before setting everything in motion,
so any further callbacks from CoreAudio will write silence and not fire the
app's audio callback again.

Fixes Bugzilla #3868.
2017-10-13 01:15:29 -04:00
Sam Lantinga
ba10d2b654 Fixed compiler warning 2017-10-12 13:55:35 -07:00
Ryan C. Gordon
5e5f2290f2 audio: Turns out the accumulation errors sound better. :/
Moving to double fixed the overflows, but using "time = i * incr" instead of
"time += incr" causes clicks in the output.
2017-10-11 12:07:43 -04:00
Ryan C. Gordon
9bd2c6b491 audio: Moved the resampler state up to double precision.
Fixes more buffer overflows.
2017-10-11 11:51:14 -04:00
Ryan C. Gordon
b2f5123b65 audio: calculate resampling time directly, don't increment (thanks, Eric!).
Fixes buffer overruns as floating point errors accumulate.

Partially fixes Bugzilla #3848.
2017-10-11 11:43:35 -04:00
Ryan C. Gordon
763c387149 audio: clamp resampler interpolation values to prevent buffer overflow.
Partially fixes Bugzilla #3848.
2017-10-11 02:33:55 -04:00
Ryan C. Gordon
0085f917e0 audio: Moved unchanging variable out of loop. 2017-10-11 02:31:58 -04:00
Ryan C. Gordon
cb8bf6bbaf audio: Make sure audio stream resampling doesn't overflow buffers. 2017-10-11 02:03:05 -04:00
Ryan C. Gordon
459e2b0bbe audio: Fixed check for minimum audio stream put size. 2017-10-11 01:37:11 -04:00
Ryan C. Gordon
903ff6414e audio: SDL_ResampleCVT() should use memmove instead of memcpy.
This copy can overlap.

Fixes Bugzilla #3849.
2017-10-10 22:31:02 -04:00
Ryan C. Gordon
42fff7ce2b audio: Don't stack-allocate resampler padding.
(I thought padding size ranged from 5 frames to ~30 frames (based around
RESAMPLER_ZERO_CROSSINGS, which is 5), but it's actually between 512 and
several thousands (based on RESAMPLER_SAMPLES_PER_ZERO_CROSSING)). It gets
big fast when downsampling.
2017-10-10 22:18:46 -04:00
Ryan C. Gordon
37d89aa10f audio: reworked audio streams to have right-hand resampling padding available.
Fixes Bugzilla #3851.
2017-10-10 16:12:56 -04:00
Ryan C. Gordon
099ae43e81 audio: Fixed compiler warning on Visual Studio. 2017-09-22 22:28:21 -04:00
Sam Lantinga
fe6b8f1c31 Fixed Mac OS X build 2017-09-22 11:25:52 -07:00
Sam Lantinga
407e1693ae Fixed audio being silent on older iOS devices
Tested on an iPod running iOS 6.1
2017-09-22 11:15:14 -07:00
Sam Lantinga
d74c00e67d Fixed memory leak when HAVE_ALLOCA isn't defined 2017-09-22 08:51:45 -07:00
Ryan C. Gordon
6d206a7b28 audio: Stream resampling now saves some samples from previous run for padding.
Previously, the padding was silence, which was a problem when streaming since
you would sample a little bit of this silence between each buffer.

We still need a means to get padding data for the right hand side, but this
patch makes the resampler output more correct.
2017-09-22 07:42:24 -04:00
Sam Lantinga
8b660c5046 Added some missing "extern" declarations 2017-09-21 00:55:29 -07:00
Ryan C. Gordon
f40bd5ee24 audio: removed my perl experiment script. 2017-09-21 02:06:53 -04:00
Ryan C. Gordon
1a3b95a11e audio: Replaced the resampler. Again.
This time it's using real math from a real whitepaper instead of my previous
amateur, fast-but-low-quality attempt. The new resampler does "bandlimited
interpolation," as described here: https://ccrma.stanford.edu/~jos/resample/

The output appears to sound cleaner, especially at high frequencies, and of
course works with non-power-of-two rate conversions.

There are some obvious optimizations to be done to this still, and there is
other fallout: this doesn't resample a buffer in-place, the 2-channels-Sint16
fast path is gone because this resampler does a _lot_ of floating point math.
There is a nasty hack to make it work with SDL_AudioCVT.

It's possible these issues are solvable, but they aren't solved as of yet.
Still, I hope this effort is slouching in the right direction.
2017-09-21 02:51:14 -04:00
Sam Lantinga
c08a7a74a5 Added a hint SDL_HINT_AUDIO_CATEGORY to control the audio category,
determining whether the phone mute switch affects the audio
2017-09-15 17:27:32 -07:00