Fix Frame Advance and FifoPlayer pause/unpause/stop.
CPU::EnableStepping is not atomic but is called from multiple threads
which races and leaves the system in a random state; also instruction
stepping was unstable, m_StepEvent had an almost random value because
of the dual purpose it served which could cause races where CPU::Run
would SingleStep when it was supposed to be sleeping.
FifoPlayer never FinishStateMove()d which was causing it to deadlock.
Rather than partially reimplementing CPU::Run, just use CPUCoreBase
and then call CPU::Run(). More DRY and less likely to have weird bugs
specific to the player (i.e the previous freezing on pause/stop).
Refactor PowerPC::state into CPU since it manages the state of the
CPU Thread which is controlled by CPU, not PowerPC. This simplifies
the architecture somewhat and eliminates races that can be caused by
calling PowerPC state functions directly instead of using CPU's
(because they bypassed the EnableStepping lock).
We don't throttle by frames, we throttle by coretiming speed.
So looking up VI for calculating the speed was just very wrong.
The new ini option is a float, 1.0f for fullspeed.
In the GUI, percentual values are used.
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.
Chris Burgener