Commit Graph

236 Commits

Author SHA1 Message Date
Tobias
aa84022704
Port yuzu-emu/yuzu#4164: "hotkeys: Add a "Mute Audio" hotkey" (#5463)
Co-authored-by: Kewlan <colin_rehn@hotmail.com>
2022-11-04 20:25:57 +01:00
GPUCode
cbd5d1c15c
Upgrade codebase to C++ 20 + fix warnings + update submodules (#6115) 2022-09-21 18:36:12 +02:00
scribblemaniac
a6e7a81de9
Use cubic mapping for volume control (#6020)
* Use cubic mapping for volume control

* Update comment for hardware volume slider
2022-05-20 22:47:37 +05:30
SachinVin
ac98458e0b
audio_core\lle\lle.cpp: Add 16 bit and 32 bit read/write callbacks (#5968) 2022-03-05 12:38:46 +05:30
liushuyu
c7869ff332
audio_core/hle/ffmpeg_decoder: make avcodec const 2022-02-21 00:51:17 -07:00
SachinVin
6183b5d76c
Merge pull request #5823 from SachinVin/dyn
Android: Backport easy stuff
2021-10-03 18:58:20 +05:30
bunnei
4f737c329e android: audio_core: Remove noisy log. 2021-09-29 22:51:11 +05:30
Pengfei Zhu
19617f7edb
dsp_interface: Move sink to the last in member list (#5844)
So that it is destructed first. Otherwise, the TimeStretcher will be destructed before the Sink, which might cause segfaults when the Sink tries to read data from the TimeStretcher afterwards.
2021-09-20 22:40:36 +05:30
Vitor Kiguchi
350c9c8d7d sdl2_sink: remove SDL_AUDIO_ALLOW_FREQUENCY_CHANGE flag
this is necessary for sdl audio to work properly in
sdl as of current dev version (2.0.15)
2021-04-29 20:59:41 -03:00
xperia64
5f1eb7146d
Merge generic part of Android microphone changes (#5624) 2020-12-30 19:21:03 -05:00
Marshall Mohror
ab6c605e59
Merge pull request #5609 from gal20/enable-fdk-patch
Enable fdk decoder for flatpak version
2020-12-30 11:59:18 -06:00
Tobias
3f13e1cc24
cubeb_sink: Use static_cast instead of reinterpret_cast in DataCallback() (#5573)
Conversions from void* to the proper data type are well-defined and
supported by static_cast. We don't need to use reinterpret_cast here.

Co-Authored-By: LC <712067+lioncash@users.noreply.github.com>

Co-authored-by: LC <712067+lioncash@users.noreply.github.com>
2020-12-07 16:06:16 +01:00
Tobias
f0e3637c7a
codec: Make lookup table static constexpr (#5572)
Allows compilers to elide needing to push these values on the stack
every time the function is called.

Co-authored-by: Lioncash <mathew1800@gmail.com>
2020-12-07 16:05:45 +01:00
xperia64
1aaec7938f
Initial implementation of partial_embedded_buffer_dirty handling (#5548)
* Initial implementation of partial_embedded_buffer_dirty handling

* Apply suggestions from code review

Co-authored-by: Marshall Mohror <mohror64@gmail.com>

* Serialize physical address, fix LOG_TRACE

* Add bracket

* Avoid crash in partial update behavior

Co-authored-by: Marshall Mohror <mohror64@gmail.com>
2020-11-17 17:31:05 -05:00
gal20
5683f86ed3
Remove pessimistic sanity check
This check creates false positive when using the flatpak runtime library
2020-11-14 20:06:25 +02:00
Vitor Kiguchi
1efe80bd10 Update cubeb and request a persistent stream session 2020-10-20 11:19:58 -03:00
FearlessTobi
51d348b087 General: Make use of std::nullopt where applicable
Allows some implementations to avoid completely zeroing out the internal
buffer of the optional, and instead only set the validity byte within
the structure.

This also makes it consistent how we return empty optionals.

Co-Authored-By: LC <712067+lioncash@users.noreply.github.com>
2020-10-03 17:25:54 +02:00
tywald
3d9d071262 cubeb_sink.cpp: Change one log from INFO to DEBUG level. 2020-08-20 20:22:50 +02:00
xperia64
b4ec50ba21
Actually return true if InitMFDLL succeeded (#5470) 2020-07-23 00:46:10 -04:00
Pengfei Zhu
5b245aafd3
Merge pull request #5402 from xperia64/update_teakra_sync
Update teakra, adjust TeakraSlice for new audio frame period
2020-06-11 22:41:01 +08:00
Pengfei Zhu
2632b421c2
Merge pull request #5266 from xperia64/audio_ticks_tweak
Adjust audio_frame_ticks
2020-06-11 22:37:30 +08:00
xperia64
b0a20180ee Update comments after hardware testing 2020-06-10 23:05:02 -04:00
xperia64
62e2cd6239 Use samples_per_frame instead of hardcoded 160 2020-06-10 17:10:50 -04:00
xperia64
20d823a42a Fix WMF AAC decoder bug 2020-06-10 16:58:09 -04:00
xperia64
daf0e750d2 Update teakra, adjust TeakraSlice for new audio frame period 2020-06-07 20:06:22 -04:00
xperia64
21159dd83a clang-format, and avoid another potential leak 2020-05-21 21:05:03 -04:00
xperia64
f9750875e3 Avoid leaking the cubeb input stream 2020-05-21 20:34:00 -04:00
xperia64
a0e8255b65 Update cycles and explanation 2020-04-26 03:14:54 -04:00
xperia64
3a1601a534 Change audio_frame_ticks with length explanation 2020-04-21 23:40:34 -04:00
xperia64
38c3c9c74b
Add sample rate field to AAC decoder (#5195)
* Add sample rate field to AAC decoder

* Fix TODO comment

* Remove unneeded conversion
2020-04-21 20:34:50 -05:00
Hamish Milne
828f88d20a Merge branch 'master' into feature/savestates-2 2020-04-12 11:24:06 +01:00
Marshall Mohror
9c7da35382
Merge pull request #5083 from zhaowenlan1779/video-dumping-update
video_core, citra_qt: Video dumping updates
2020-04-03 21:15:32 -05:00
Hamish Milne
92640fc29c Code review actions (plus hopefully fix the linux CI) 2020-03-31 17:54:28 +01:00
Hamish Milne
841255cd16 Attempt to fix the linux builds 2020-03-28 21:40:18 +00:00
James Rowe
a6ee1bf913
HLE Audio: Increase frame position by input buffer sample rate (#5049)
* HLE Audio: Increase frame position by input buffer sample rate

Currently the frame position moves ahead by the number of samples
output, but thats a fixed number based on the 3ds native sample rate.
Instead, based on a homebrew by cyuubi and looking at the lle audio,
this sample position should be moved forward by the number of samples
from the input buffer that was read, based on the buffer's sample rate.
2020-03-28 10:39:50 -05:00
Hamish Milne
025960bcdd Attempt to fix flatpak CI 2020-03-28 15:10:35 +00:00
Hamish Milne
7049af744f Merge remote-tracking branch 'upstream/master' into feature/savestates-2 2020-03-28 12:33:21 +00:00
Hamish Milne
232b52a27d Minor cleanup 2020-03-28 09:59:45 +00:00
Hamish Milne
3d1180ee21 DSP now works... committing this!! 2020-03-27 22:19:29 +00:00
BreadFish64
37384174d9 fix formatting for media-ndk 2020-03-17 21:15:33 -05:00
zhang wei
3410b96400
mediandk for android (#4921) 2020-03-16 21:07:22 -05:00
Hamish Milne
da3ab3d56e Merge branch 'master' into feature/savestates-2 2020-03-07 21:23:08 +00:00
zhupengfei
3c6765e87c
core: Properly std::move things around 2020-02-27 16:55:09 +08:00
liushuyu
cff00f38c5
Implements fdk_aac decoder (#4764)
* audio_core: dsp_hle: implements fdk_aac decoder

* audio_core: dsp_hle: clean up and add comments

* audio_core: dsp_hle: move fdk include to cpp file

* audio_core: dsp_hle: detects broken fdk_aac...

... and refuses to initialize if that's the case

* audio_core: dsp_hle: fdk_aac: address comments...

... and rebase commits

* fdk_decoder: move fdk header to cpp file
2020-02-23 11:01:21 +01:00
Hamish Milne
c983528862 Reworked DSP serialization 2020-02-13 17:42:12 +08:00
Hamish Milne
6917eaf53b Use load_construct_data for kernel objects 2020-02-13 17:38:25 +08:00
Hamish Milne
ee2cae2093 Added core serialization 2020-02-13 17:34:13 +08:00
James Rowe
5fd1ff08d7
Merge pull request #5024 from jroweboy/temp-hle-audio-fix
Prevent out of memory errors when the game passes in an improper length value
2020-01-21 15:30:20 -07:00
James Rowe
e53a2ac411 Reenable AAC FFMPEG decoding
Simple cut/paste issue where initialized is only set to true when the
emulation attempts to init the Binary Pipe, but we used it to test if
the FFMPEG decoder was valid and disabled it if it wasn't. Just return
the value of have_ffmpeg_dl instead so when dynamic loading is added
it'll still work.
2019-12-16 19:23:43 -07:00
James Rowe
87facaa2e2 Prevent out of memory errors when the game passes in an improper length value
HACK

In Luigi's Mansion Dark Moon in HLE audio, the game mysteriously passes
in an extremely large value for length, which without any checks, causes
HLE audio to allocate an extremely large buffer.

This value seemingly is caused by some other HLE audio feature is missing,
and Luigi's Mansion subtracts two values to get a length, without
checking for overflow first. This appears to be caused by an incorrect
HLE audio emulation, as its fixed entirely by only changing to LLE. As
such, further investigation is required, but in the meantime, completely
eating up our users RAM is unacceptable.
2019-12-14 18:18:59 -07:00