// 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;
}
// ------------------------------------------------------------------------------------------
-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
- sub->buffers = 1;
+ // 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;
}
}
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
// ------------------------------------------------------------------------------------------
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)
(*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)
// 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);
}
}
- // 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
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
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
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
// 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;
// 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;
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 :
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