aa84022704
Port yuzu-emu/yuzu#4164: "hotkeys: Add a "Mute Audio" hotkey" ( #5463 )
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Co-authored-by: Kewlan <colin_rehn@hotmail.com >
2022-11-04 20:25:57 +01:00
GPUCode and GitHub
cbd5d1c15c
Upgrade codebase to C++ 20 + fix warnings + update submodules ( #6115 )
2022-09-21 18:36:12 +02:00
scribblemaniac and GitHub
a6e7a81de9
Use cubic mapping for volume control ( #6020 )
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* Use cubic mapping for volume control
* Update comment for hardware volume slider
2022-05-20 22:47:37 +05:30
SachinVin and GitHub
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 and GitHub
6183b5d76c
Merge pull request #5823 from SachinVin/dyn
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Android: Backport easy stuff
2021-10-03 18:58:20 +05:30
bunnei and SachinVin
4f737c329e
android: audio_core: Remove noisy log.
2021-09-29 22:51:11 +05:30
Pengfei Zhu and GitHub
19617f7edb
dsp_interface: Move sink to the last in member list ( #5844 )
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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
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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 and GitHub
5f1eb7146d
Merge generic part of Android microphone changes ( #5624 )
2020-12-30 19:21:03 -05:00
Marshall Mohror and GitHub
ab6c605e59
Merge pull request #5609 from gal20/enable-fdk-patch
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Enable fdk decoder for flatpak version
2020-12-30 11:59:18 -06:00
3f13e1cc24
cubeb_sink: Use static_cast instead of reinterpret_cast in DataCallback() ( #5573 )
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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
f0e3637c7a
codec: Make lookup table static constexpr ( #5572 )
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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
1aaec7938f
Initial implementation of partial_embedded_buffer_dirty handling ( #5548 )
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* 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 and GitHub
5683f86ed3
Remove pessimistic sanity check
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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 and LC
51d348b087
General: Make use of std::nullopt where applicable
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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 and GitHub
b4ec50ba21
Actually return true if InitMFDLL succeeded ( #5470 )
2020-07-23 00:46:10 -04:00
Pengfei Zhu and GitHub
5b245aafd3
Merge pull request #5402 from xperia64/update_teakra_sync
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Update teakra, adjust TeakraSlice for new audio frame period
2020-06-11 22:41:01 +08:00
Pengfei Zhu and GitHub
2632b421c2
Merge pull request #5266 from xperia64/audio_ticks_tweak
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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 and GitHub
38c3c9c74b
Add sample rate field to AAC decoder ( #5195 )
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* 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 and GitHub
9c7da35382
Merge pull request #5083 from zhaowenlan1779/video-dumping-update
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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 and GitHub
a6ee1bf913
HLE Audio: Increase frame position by input buffer sample rate ( #5049 )
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* 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 and GitHub
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 and GitHub
cff00f38c5
Implements fdk_aac decoder ( #4764 )
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* 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 and zhupengfei
c983528862
Reworked DSP serialization
2020-02-13 17:42:12 +08:00
Hamish Milne and zhupengfei
6917eaf53b
Use load_construct_data for kernel objects
2020-02-13 17:38:25 +08:00
Hamish Milne and zhupengfei
ee2cae2093
Added core serialization
2020-02-13 17:34:13 +08:00
James Rowe and GitHub
5fd1ff08d7
Merge pull request #5024 from jroweboy/temp-hle-audio-fix
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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
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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
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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