snd_pcm_writei() is meant to block block until all samples are written,
but apparently in some situations it can block for much longer, prehaps
even a infinite time, in the case of virtual machine FifoCI runs in.
Because it grabed a mutex before blocking, it could also block the
Clear() call for an infinite length of time, blocking dolphin's emu
thread.
snd_pcm_writei() also takes 10-15 seconds if you run dolphin under GDB
and can randomly take 5 or so seconds during normal usage.
By moving all the pause code to the ALSA thread, Clear() no-longer
blocks and everyone keeps their sanity.
When the emulation is paused and the ALSA backend is used, make the audio
thread wait on a condition variable instead of busy-waiting. This commit
fixes bug #7729
Since the ALSA API is not thread-safe, calls to snd_pcm_drop() and snd_pcm_prepare()
in AlsaSound::Clear() are protected by the same mutex as the condition variable in AlsaSound::SoundLoop()
to make sure that we do not call these functions while a call to
snd_pcm_writei() is ongoing.
This fixes a race condition:
Before this commit, there was a race condition when starting a game:
Core::EmuThread(), after having started (but not necessarily completed)
the initialization of the audio thread, calls Core::SetState() which calls
CCPU::EnableStepping(), which in turns calls AudioCommon::ClearAudioBuffer().
This means that SoundStream::Clear() can be called before
AlsaSound::AlsaInit() has completed.
If the selected audio backend fails to Start() (which could happen for
example if there is no audio device), we currently still use the backend
anyway. This can lead to crashes on some platforms (such as Windows) and
is outright wrong anyway.
This commit fallbacks to the Null audio backend if the selected backend
couldn't be started.
This fixes bug #6001
We had to lock audiocommon with the old asynchron HLE audio emulation,
now our Mixer is just a plain FIFO which may underrun.
Of course, this will stutter, but underruning the audio backend is likely worse.
For more information:
https://docs.google.com/document/d/1tBEgsJh7QiwNwepXI0eobfK3U8LkJButSyeuFt1degM/edit?usp=sharing
removed: SSE includes (not used)
added: 16bit -> float -> 16bit functions
added: linear interpolator and high-quality (windowed-sinc) interpolator functions (including Resampler class)
added: dithering
changed: renamed variables and reformatted a few things to fit with style guide (braces, #include->const)
changed: use s16, u16, s32, u32 etc
changed: store samples and do all computations as floats
changed: volume from 0 - 255
Code was only using front-left and front-right to calculate bass, but
HRTF code - which this was once based on - uses all five channels and
this sounds fuller.
This change was done because with the previous method of dumping audio, the mixer would handle switching the RL being emitted by the DSP to LR, and thus would provide the proper channel orientation. Because we're now dumping directly from PushSamples() and PushStreamingSamples(), it was writing the right channel to the left channel of the wave file and vice versa.
Each emulated Wiimote can have its speaker routed from left to right via the "Speaker Pan" setting in the emulated wiimote settings dialog. Use any value from -127 for leftmost to 127 for rightmost with 0 being the centre.
Added code in the InputConfig to use a spin control for non-boolean values.
Defaulted the setting of "Enable Speaker Data" to disabled.
The Wiimotes are positioned as follows:
Wiimote 0 = Center
Wiimote 1 = Left
Wiimote 2 = Right
Wiimote 3 = Center
The Wiimote speaker output can be disabled via the "Enable Speaker Data" checkbox in the Wiimote settings.
It was only used for Windows XP and lower.
This also bumps the _WIN32_WINNT define in the stdafx precompiled headers to set the minimum version as Windows Vista.
The two instances of this class were sharing a frac variable causing
audio glitches when both were running (which is now all the time).
Fixes issue 7463 (Since DTK merge, audio has staic in it).
The code actually handles this case correctly; the algorithm is linear
interpolation between the two closest samples, and the way it is written
should work correctly with any ratio.
The primary motivation here is to make sure we submit samples from the
CPU thread. This makes sure the timing of related interrupts accurate,
and generally keeps the different kinds of audio synchronized. This will also
allow improvements to audio dumping functionality.
The new code is also more concise because it gets rid of some duplicated
audio mixing code.
This reverts commit 4990b8910b.
The commit is causing substantial performance issues for the DSound
backend which I somehow didn't catch during testing.
Pretty straightforward; IDirectSoundNotify lets you register for
notifications after a certain amount of sound has played, so use that
instead of depending on Update() notifications from the CPU thread.
Also, while I'm here, reduce the buffer size by a factor of 4; this seems
to reduce the latency, although the difference is sort of subtle.
This is good for a couple of reasons: one, it gets rid of duplicated code,
and two, DSP emulation shouldn't need to interact with audio in the first
place.
- remove unused variables
- reduce the scope where it makes sense
- correct limits (did you know that strcat()'s last parameter does not
include the \0 that is always added?)
- set some free()'d pointers to NULL
The default async api allow us to set some latency options. The old one (simple API) was the lazy way to go for usual audio where latency doesn't matter.
This also streams audio, so it should be a bit faster then the old one.
* Currently there is no DEBUGFAST configuration. Defining DEBUGFAST as a preprocessor definition in Base.props (or a global header) enables it for now, pending a better method. This was done to make managing the build harder to screw up. However it may not even be an issue anymore with the new .props usage.
* D3DX11SaveTextureToFile usage is dropped and not replaced.
* If you have $(DXSDK_DIR) in your global property sheets (Microsoft.Cpp.$(PlatformName).user), you need to remove it. The build will error out with a message if it's configured incorrectly.
* If you are on Windows 8 or above, you no longer need the June 2010 DirectX SDK installed to build dolphin. If you are in this situation, it is still required if you want your built binaries to be able to use XAudio2 and XInput on previous Windows versions.
* GLew updated to 1.10.0
* compiler switches added: /volatile:iso, /d2Zi+
* LTCG available via msbuild property: DolphinRelease
* SDL updated to 2.0.0
* All Externals (excl. OpenAL and SDL) are built from source.
* Now uses STL version of std::{mutex,condition_variable,thread}
* Now uses Build as root directory for *all* intermediate files
* Binary directory is populated as post-build msbuild action
* .gitignore is simplified
* UnitTests project is no longer compiled
This fix the 1h32 audio bug which outputs static sound after 1h32.
The mixer is used for 32->48kHz resampling and as output buffer for the async audio backends.
So this buffer was indiced by a writing and a reading pointer and the count of samples in it.
As this is redundant and the sample count isn't accurate calculateable because of the interpolation,
both indices gets out of sync. So after some time (~92min), this buffer overflows and return only garbage.
thx @ moosehunter + delroth for debugging on this issue. You did the most work :-)
Also, some tab/space mismatches removed from VideoOGL, and some places I missed in VideoDX[number] projects.
Now, the Core is literally the only project with tab/space mismatches (on a large scale).
This commit mainly elaborates on some messages a little more. Also fixes some typos that slipped through the last commit.
A large change in text can be seen in EXI_DeviceMemoryCard.cpp. I added more info as to why a write to a memory card may fail. (This actually was a reason I was unable to write to a memcard recently).
Elaborations can be seen in WGL.cpp
I did change some comments in some files that I was correcting logging messages in, however this is only if I spot a typo or if an abbreviation is lower-cased. Even in that case, the amount of changes done to comments is very minimal.
* OpenAL:
Changed SoundTouch to use float samples, allowing SSE to be used. Made the DPL2 decoder disabled by default. Re-added the audio hack used by the Accurate VBeam emulation option.
Added a latency setting to the audio settings. Removed the Sample Rate setting. It is now hardcoded to 48000hz (accurate audio timing).
Skipped timestretching if the emulator is running below 10% speed to prevent buffer overflows.
Removed the synchronisation between the CPU thread and the audio thread. Added code to detect and resume from buffer underruns. Disabled the ability to change the DPL2 option after the game has started. Fixed a memory leak that occurred in the DPL2 decoder. Fixed the OSX build.
Build fix
Added a Dolby Pro Logic II (DPL2) decoder in the OpenAL backend. DPL2 audio is decoded to 5.1. Code adapted from ffdshow. Added an option in the DSP settings to disable the DPL2 decoder in case Dolphin incorrectly detects a 5.1 audio system. Updated the OpenAL files to OpenAL Soft 1.15.1 in the Windows build.
Removed the system timing hack which was activated when the Accurate VBeam option was enabled.
Fixed the include directories in Audio Common for the Windows 32bit build.
Fixed the include directories in Audio Common for the Windows build.
Messed up the static include line
Fix include paths and compiling in Linux. Externals soundtouch is 1.7.1, while Ubuntu 12.10 is 1.6.x. Externals soundtouch is compiled with integer samples, while ubuntu is compiled with float samples. Float samples is probably the more common route. If you're going to use soundtouch, you should probably use SAMPLETYPE instead of explicitly choosing short. This probably breaks the windows build since its includes aren't setup.
OSX: typedef signed char BOOL
OSX build fix
Build fix
Added audio time stretching by using the SoundTouch library.
Implemented correct audio timing.
OpenAL for Windows initial commit
Added code to detect and resume from buffer underruns.
Disabled the ability to change the DPL2 option after the game has started.
Fixed a memory leak that occurred in the DPL2 decoder.
Fixed the OSX build.
Added an option in the DSP settings to disable the DPL2 decoder in case Dolphin incorrectly detects a 5.1 audio system.
Updated the OpenAL files to OpenAL Soft 1.15.1 in the Windows build.
Fixes issue 3023.
They are currently broken and cause sound issues which are not present in other
backends:
* OpenAL plays music 2x too fast in Zelda UCode games with HLE
* Pulse backend uses a lot of CPU power and slows down emulation significantly
Both backends are currently being re-implemented in separate branches of
Dolphin, so this should be a temporary removal.
the intent is to replace the haphazard scheduling and finger-crossing associated with saving/loading with the correct and minimal necessary wait for each thread to reach a known safe location before commencing the savestate operation, and for any already-paused components to not need to be resumed to do so.
XGRA Extreme-G Racing Association/Conduit 2... Enabled 48kHz output for LLE (before it was just 32kHz). I hope it works, if there are serious problems, it will be reverted.
git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@7511 8ced0084-cf51-0410-be5f-012b33b47a6e
Fix a few more dialogs to close with escape.
Fix an issue with the pulse audio sound backend when the dsp emulation method is changed, and the frequency setting was not respected.
git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@7358 8ced0084-cf51-0410-be5f-012b33b47a6e
Uses the last sample from the ppc buffer to fill the samples the ppc
didn't deliver data for, avoids clicking on underruns.
git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@7338 8ced0084-cf51-0410-be5f-012b33b47a6e
Ryan Houdek