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Make DSound2 audio player use dynamic latency trickery and only do streaming (looping buffer) playback for long playback durations. This should fix most if not all "could not lock buffer" errors and close #855. 

Originally committed to SVN as r2997.
This commit is contained in:
Niels Martin Hansen 2009-06-01 14:53:18 +00:00
parent db27d770f6
commit da78539ab2
1 changed files with 46 additions and 6 deletions
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48
aegisub/src/audio_player_dsound2.cpp
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@ -236,10 +236,12 @@ void DirectSoundPlayer2Thread::Run()
int64_t next_input_frame = 0;
DWORD buffer_offset = 0;
bool playback_should_be_running = false;
int current_latency = wanted_latency;
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*provider->GetBytesPerSample()/1000;
while (running)
{
DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait)/sizeof(HANDLE), events_to_wait, FALSE, wanted_latency);
DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait)/sizeof(HANDLE), events_to_wait, FALSE, current_latency);
switch (wait_result)
{
@ -255,6 +257,9 @@ void DirectSoundPlayer2Thread::Run()
void *buf;
buffer_offset = 0;
if (FAILED(bfr->SetCurrentPosition(0)))
REPORT_ERROR("Could not reset playback buffer cursor before filling first buffer.")
HRESULT res = bfr->Lock(buffer_offset, 0, &buf, &buf_size, 0, 0, DSBLOCK_ENTIREBUFFER);
while (FAILED(res)) // yes, while, so I can break out of it without a goto!
{
@ -274,14 +279,30 @@ void DirectSoundPlayer2Thread::Run()
REPORT_ERROR("Could not lock buffer for playback.")
}
buffer_offset += FillAndUnlockBuffers(buf, buf_size, 0, 0, next_input_frame, bfr.obj);
// Clear the buffer in case we can't fill it completely
memset(buf, 0, buf_size);
DWORD bytes_filled = FillAndUnlockBuffers(buf, buf_size, 0, 0, next_input_frame, bfr.obj);
buffer_offset += bytes_filled;
if (buffer_offset >= bufSize) buffer_offset -= bufSize;
if (FAILED(bfr->SetCurrentPosition(0)))
REPORT_ERROR("Could not reset playback buffer cursor before playback.")
if (bytes_filled < wanted_latency_bytes)
{
// Very short playback length, do without streaming playback
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
if (FAILED(bfr->Play(0, 0, 0)))
REPORT_ERROR("Could not start single-buffer playback.")
}
else
{
// We filled the entire buffer so there's reason to do streaming playback
current_latency = wanted_latency;
if (FAILED(bfr->Play(0, 0, DSBPLAY_LOOPING)))
REPORT_ERROR("Could not start looping playback.")
}
SetEvent(is_playing);
playback_should_be_running = true;
@ -342,7 +363,8 @@ void DirectSoundPlayer2Thread::Run()
if (!(status & DSBSTATUS_LOOPING))
{
// Not really what we expected...
// Not looping playback...
// hopefully we only triggered timeout after being done with the buffer
bfr->Stop();
ResetEvent(is_playing);
playback_should_be_running = false;
@ -380,9 +402,27 @@ void DirectSoundPlayer2Thread::Run()
REPORT_ERROR("Could not lock buffer for filling.")
}
buffer_offset += FillAndUnlockBuffers(buf1, buf1sz, buf2, buf2sz, next_input_frame, bfr.obj);
DWORD bytes_filled = FillAndUnlockBuffers(buf1, buf1sz, buf2, buf2sz, next_input_frame, bfr.obj);
buffer_offset += bytes_filled;
if (buffer_offset >= bufSize) buffer_offset -= bufSize;
if (bytes_filled < 1024)
{
// Arbitrary low number, we filled in very little so better get back to filling in the rest with silence
// really fast... set latency to zero in this case.
current_latency = 0;
}
else if (bytes_filled < wanted_latency_bytes)
{
// Didn't fill as much as we wanted to, let's get back to filling sooner than normal
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
}
else
{
// Plenty filled in, do regular latency
current_latency = wanted_latency;
}
break;
}