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wasapi: Fixed glitchy audio in case of Exclusive mode, 24-bit packed format and USB DAC (other audio devices using the same audio format could be affected too).

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dmitrykos 2021-02-16 12:15:00 +02:00
commit 7926e3a8ee
1 changed files with 38 additions and 42 deletions
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78
src/hostapi/wasapi/pa_win_wasapi.c
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@ -1026,27 +1026,26 @@ static UINT32 ALIGN_NEXT_POW2(UINT32 v)
// Aligns WASAPI buffer to 128 byte packet boundary. HD Audio will fail to play if buffer
// is misaligned. This problem was solved in Windows 7 were AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED
// is thrown although we must align for Vista anyway.
static UINT32 AlignFramesPerBuffer(UINT32 nFrames, UINT32 nSamplesPerSec, UINT32 nBlockAlign,
ALIGN_FUNC pAlignFunc)
static UINT32 AlignFramesPerBuffer(UINT32 nFrames, UINT32 nBlockAlign, ALIGN_FUNC pAlignFunc)
{
#define HDA_PACKET_SIZE (128)
long frame_bytes = nFrames * nBlockAlign;
long packets;
(void)nSamplesPerSec;
UINT32 bytes = nFrames * nBlockAlign;
UINT32 packets;
// align to packet size
frame_bytes = pAlignFunc(frame_bytes, HDA_PACKET_SIZE); // use ALIGN_FWD if bigger but safer period is more desired
// align to a HD Audio packet size
bytes = pAlignFunc(bytes, HDA_PACKET_SIZE);
// atlest 1 frame must be available
if (frame_bytes < HDA_PACKET_SIZE)
frame_bytes = HDA_PACKET_SIZE;
if (bytes < HDA_PACKET_SIZE)
bytes = HDA_PACKET_SIZE;
nFrames = frame_bytes / nBlockAlign;
packets = frame_bytes / HDA_PACKET_SIZE;
packets = bytes / HDA_PACKET_SIZE;
bytes = packets * HDA_PACKET_SIZE;
nFrames = bytes / nBlockAlign;
frame_bytes = packets * HDA_PACKET_SIZE;
nFrames = frame_bytes / nBlockAlign;
// WASAPI frames are always aligned to at least 8
nFrames = ALIGN_FWD(nFrames, 8);
return nFrames;
@ -3161,26 +3160,32 @@ static PaUint32 _GetFramesPerHostBuffer(PaUint32 userFramesPerBuffer, PaTime sug
}
// ------------------------------------------------------------------------------------------
static void _RecalculateBuffersCount(PaWasapiSubStream *sub, UINT32 userFramesPerBuffer, UINT32 framesPerLatency, BOOL fullDuplex)
static void _RecalculateBuffersCount(PaWasapiSubStream *sub, UINT32 userFramesPerBuffer, UINT32 framesPerLatency,
BOOL fullDuplex, BOOL output)
{
// Count buffers (must be at least 1)
sub->buffers = (userFramesPerBuffer ? framesPerLatency / userFramesPerBuffer : 0);
sub->buffers = (userFramesPerBuffer != 0 ? framesPerLatency / userFramesPerBuffer : 1);
if (sub->buffers == 0)
sub->buffers = 1;
// Determine amount of buffers used:
// - Full-duplex mode will lead to period difference, thus only 1.
// - Input mode, only 1, as WASAPI allows extraction of only 1 packet.
// - For Shared mode we use double buffering.
// Determine number of buffers used:
// - Full-duplex mode will lead to period difference, thus only 1
// - Input mode, only 1, as WASAPI allows extraction of only 1 packet
// - For Shared mode we use double buffering
if ((sub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) || fullDuplex)
{
BOOL eventMode = ((sub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) == AUDCLNT_STREAMFLAGS_EVENTCALLBACK);
// Exclusive mode does not allow >1 buffers be used for Event interface, e.g. GetBuffer
// call must acquire max buffer size and it all must be processed.
if (sub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK)
if (eventMode)
sub->userBufferAndHostMatch = 1;
// Use paUtilBoundedHostBufferSize because exclusive mode will starve and produce
// bad quality of audio
// Full-duplex or Event mode: prefer paUtilBoundedHostBufferSize because exclusive mode will starve
// and produce glitchy audio
// Output Polling mode: prefer paUtilFixedHostBufferSize (buffers != 1) for polling mode is it allows
// to consume user data by fixed size data chunks and thus lowers memory movement (less CPU usage)
if (fullDuplex || eventMode || !output)
sub->buffers = 1;
}
}
@ -3195,7 +3200,7 @@ static void _CalculateAlignedPeriod(PaWasapiSubStream *pSub, UINT32 *nFramesPerL
if (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
{
(*nFramesPerLatency) = AlignFramesPerBuffer((*nFramesPerLatency),
pSub->wavex.Format.nSamplesPerSec, pSub->wavex.Format.nBlockAlign, pAlignFunc);
pSub->wavex.Format.nBlockAlign, pAlignFunc);
}
// Calculate period
@ -3205,7 +3210,7 @@ static void _CalculateAlignedPeriod(PaWasapiSubStream *pSub, UINT32 *nFramesPerL
// ------------------------------------------------------------------------------------------
static void _CalculatePeriodicity(PaWasapiSubStream *pSub, BOOL output, REFERENCE_TIME *periodicity)
{
// Note: according Microsoft docs for IAudioClient::Initialize we can set periodicity of the buffer
// Note: according to Microsoft docs for IAudioClient::Initialize we can set periodicity of the buffer
// only for Exclusive mode. By setting periodicity almost equal to the user buffer frames we can
// achieve high quality (less glitchy) low-latency audio.
if (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
@ -3216,7 +3221,7 @@ static void _CalculatePeriodicity(PaWasapiSubStream *pSub, BOOL output, REFERENC
(*periodicity) = pSub->period;
// Try make buffer ready for I/O once we request the buffer readiness for it. Only Polling mode
// because for Event mode buffer size and periodicity must be equal according Microsoft
// because for Event mode buffer size and periodicity must be equal according to Microsoft
// documentation for IAudioClient::Initialize.
//
// TO-DO: try spread to capture and full-duplex cases (not tested and therefore disabled)
@ -3230,7 +3235,7 @@ static void _CalculatePeriodicity(PaWasapiSubStream *pSub, BOOL output, REFERENC
// Align frames backwards, so device will likely make buffer read ready when we are ready
// to read it (our scheduling will wait for amount of millisoconds of frames_per_buffer)
alignedFrames = AlignFramesPerBuffer(pSub->params.frames_per_buffer,
pSub->wavex.Format.nSamplesPerSec, pSub->wavex.Format.nBlockAlign, ALIGN_BWD);
pSub->wavex.Format.nBlockAlign, ALIGN_BWD);
userPeriodicity = MakeHnsPeriod(alignedFrames, pSub->wavex.Format.nSamplesPerSec);
@ -3429,7 +3434,7 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
}
}
// Set device scheduling period (always 0 in Shared mode according Microsoft docs)
// Set device scheduling period (always 0 in Shared mode according to Microsoft docs)
_CalculatePeriodicity(pSub, output, &eventPeriodicity);
// Open the stream and associate it with an audio session
@ -3483,7 +3488,7 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
goto done;
}
// Set device scheduling period (always 0 in Shared mode according Microsoft docs)
// Set device scheduling period (always 0 in Shared mode according to Microsoft docs)
_CalculatePeriodicity(pSub, output, &eventPeriodicity);
// Open the stream and associate it with an audio session
@ -3523,7 +3528,7 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
goto done;
}
// Set device scheduling period (always 0 in Shared mode according Microsoft docs)
// Set device scheduling period (always 0 in Shared mode according to Microsoft docs)
_CalculatePeriodicity(pSub, output, &eventPeriodicity);
// Open the stream and associate it with an audio session
@ -3556,7 +3561,7 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
goto done;
}
// Set device scheduling period (always 0 in Shared mode according Microsoft docs)
// Set device scheduling period (always 0 in Shared mode according to Microsoft docs)
_CalculatePeriodicity(pSub, output, &eventPeriodicity);
// Open the stream and associate it with an audio session
@ -3583,7 +3588,7 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
// Recalculate buffers count
_RecalculateBuffersCount(pSub, userFramesPerBuffer, MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec),
fullDuplex);
fullDuplex, output);
// No error, client is successfully created
(*pa_error) = paNoError;
@ -3631,15 +3636,6 @@ static PaError ActivateAudioClientOutput(PaWasapiStream *stream)
// Correct buffer to max size if it maxed out result of GetBufferSize
stream->out.bufferSize = maxBufferSize;
// Get interface latency (actually unneeded as we calculate latency from the size of maxBufferSize)
if (FAILED(hr = IAudioClient_GetStreamLatency(stream->out.clientParent, &stream->out.deviceLatency)))
{
LogHostError(hr);
LogPaError(result = paInvalidDevice);
goto error;
}
//stream->out.latencySeconds = nano100ToSeconds(stream->out.deviceLatency);
// Number of frames that are required at each period
stream->out.framesPerHostCallback = maxBufferSize;
@ -6085,7 +6081,7 @@ static UINT32 ConfigureLoopSleepTimeAndScheduler(PaWasapiStream *stream, ThreadI
UINT32 userFramesOut = stream->out.framesPerBuffer;
// Adjust polling time for non-paUtilFixedHostBufferSize, input stream is not adjustable as it is being
// polled according its packet length
// polled according to its packet length
if (stream->bufferMode != paUtilFixedHostBufferSize)
{
userFramesOut = (stream->bufferProcessor.framesPerUserBuffer ? stream->bufferProcessor.framesPerUserBuffer :