Now renders some of the cutscene audio in Zelda TWW. Most of the work around
sample sources is in a good enough state, even though it is still missing
features like Dolby mixing, IIR, etc.
Now accurate control flow for DAC. Voice rendering is not implemented yet, but
framing seems correct. Not expected to work for anything than DAC UCode games.
Push the mail at UCode boot time, not after the first update. This avoids a lot
of crazy busy-looping on the CPU side while waiting for the DSP to be
initialized.
This change is meant to solve the following problem: how to translate the
following snippet to DSPHLE:
SendInterruptAndWaitRead(MAIL_A);
SendAndWaitRead(MAIL_B);
SendInterruptAndWaitRead(MAIL_C);
This should cause the following actions on the CPU side:
---> Woken up by interrupt
Reads MAIL_A
Reads MAIL_B
<--- Exits interrupt handler
---> Woken up by interrupt
Reads MAIL_C
<---
But with the current DSPHLE mail support, the following would happen because
the "AndWaitRead" part is not supported:
---> Woken up by interrupt
Reads MAIL_A
Reads MAIL_B
<--- Exits interrupt handler
[Never gets the second interrupt since it was triggered at the same time as
the first one! Misses MAIL_C.]
This changes fixes the issue by storing two values in the mail queue on the DSP
side: the value of the mail itself, and whether a read of that mail should
trigger a DSP interrupt. If nothing is in the queue yet and an interrupt is
requested, just trigger the interrupt. In the present example, the queue will
look like this:
Mail value Interrupt requested
MAIL_A No <-- Interrupt was triggered when
pushing the mail to the queue.
MAIL_B Yes
MAIL_C No
When the CPU will read MAIL_B, this will cause MailHandler to trigger the
interrupt, which will be handled by the CPU when coming back from the exception
handler. MAIL_C is then successfully read.
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
Pierre Bourdon