/*
* Portable Audio I/O Library WASAPI implementation
* Copyright (c) 2006-2010 David Viens
- * Copyright (c) 2010-2018 Dmitry Kostjuchenko
+ * Copyright (c) 2010-2019 Dmitry Kostjuchenko
*
* Based on the Open Source API proposed by Ross Bencina
- * Copyright (c) 1999-2018 Ross Bencina, Phil Burk
+ * Copyright (c) 1999-2019 Ross Bencina, Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
#define PA_WASAPI_MAX_CONST_DEVICE_COUNT 0 // Force basic behavior by defining 0 if not defined by user
#endif
+// Fallback from Event to the Polling method in case if latency is higher than 21.33ms, as it allows to use
+// 100% of CPU inside the PA's callback.
+// Note: Some USB DAC drivers are buggy when Polling method is forced in Exclusive mode, audio output becomes
+// unstable with a lot of interruptions, therefore this define is optional. The default behavior is to
+// not change the Event mode to Polling and use the mode which user provided.
+//#define PA_WASAPI_FORCE_POLL_IF_LARGE_BUFFER
+
// WinRT
#if defined(WINAPI_FAMILY) && (WINAPI_FAMILY == WINAPI_FAMILY_APP)
#define PA_WINRT
#else
// Determine number of available devices by activating AudioClient for render and capture data flows
- if (!FAILED(ActivateAudioInterface_WINRT(eRender, GetAudioClientIID(), &tmpClient)))
+ if (SUCCEEDED(ActivateAudioInterface_WINRT(eRender, GetAudioClientIID(), &tmpClient)))
{
paWasapi->deviceCount++;
SAFE_RELEASE(tmpClient);
}
- if (!FAILED(ActivateAudioInterface_WINRT(eCapture, GetAudioClientIID(), &tmpClient)))
+ if (SUCCEEDED(ActivateAudioInterface_WINRT(eCapture, GetAudioClientIID(), &tmpClient)))
{
paWasapi->deviceCount++;
SAFE_RELEASE(tmpClient);
#ifndef PA_WINRT
if (!SetupAVRT())
{
- PRINT(("WASAPI: No AVRT! (not VISTA?)"));
+ PRINT(("WASAPI: No AVRT! (not VISTA?)\n"));
return paNoError;
}
#endif
}
// ------------------------------------------------------------------------------------------
-static void _CalculateAlignedPeriod(PaWasapiSubStream *pSub, UINT32 *nFramesPerLatency,
- ALIGN_FUNC pAlignFunc)
+static void _CalculateAlignedPeriod(PaWasapiSubStream *pSub, UINT32 *nFramesPerLatency, ALIGN_FUNC pAlignFunc)
{
// Align frames to HD Audio packet size of 128 bytes for Exclusive mode only.
// Not aligning on Windows Vista will cause Event timeout, although Windows 7 will
pSub->period = MakeHnsPeriod((*nFramesPerLatency), pSub->wavex.Format.nSamplesPerSec);
}
+// ------------------------------------------------------------------------------------------
+static void _CalculatePeriodicity(PaWasapiSubStream *pSub, BOOL output, REFERENCE_TIME *periodicity)
+{
+ // Note: according 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)
+ {
+ const PaWasapiDeviceInfo *pInfo = pSub->params.device_info;
+
+ // By default periodicity equals to the full buffer (legacy PA WASAPI's behavior)
+ (*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
+ // documentation for IAudioClient::Initialize.
+ //
+ // TO-DO: try spread to capture and full-duplex cases (not tested and therefore disabled)
+ //
+ if (((pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) == 0) &&
+ (output && !pSub->params.full_duplex))
+ {
+ UINT32 alignedFrames;
+ REFERENCE_TIME userPeriodicity;
+
+ // Align frames backwards, so device will likely make buffer read ready when we are ready
+ // to read it (our sheduling 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);
+
+ userPeriodicity = MakeHnsPeriod(alignedFrames, pSub->wavex.Format.nSamplesPerSec);
+
+ // Must not be larger than buffer size
+ if (userPeriodicity > pSub->period)
+ userPeriodicity = pSub->period;
+
+ // Must not be smaller than minimum supported by the device
+ if (userPeriodicity < pInfo->MinimumDevicePeriod)
+ userPeriodicity = pInfo->MinimumDevicePeriod;
+
+ (*periodicity) = userPeriodicity;
+ }
+ }
+ else
+ (*periodicity) = 0;
+}
+
// ------------------------------------------------------------------------------------------
static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSub, BOOL output, PaError *pa_error)
{
PaError error;
HRESULT hr;
-
const PaWasapiDeviceInfo *pInfo = pSub->params.device_info;
const PaStreamParameters *params = &pSub->params.stream_params;
- UINT32 framesPerLatency = pSub->params.frames_per_buffer;
- double sampleRate = pSub->params.sample_rate;
- //BOOL blocking = pSub->params.blocking;
- BOOL fullDuplex = pSub->params.full_duplex;
-
- const UINT32 userFramesPerBuffer = framesPerLatency;
- IAudioClient *audioClient = NULL;
+ const double sampleRate = pSub->params.sample_rate;
+ const BOOL fullDuplex = pSub->params.full_duplex;
+ const UINT32 userFramesPerBuffer = pSub->params.frames_per_buffer;
+ UINT32 framesPerLatency = userFramesPerBuffer;
+ IAudioClient *audioClient = NULL;
+ REFERENCE_TIME eventPeriodicity = 0;
// Assume default failure due to some reason
(*pa_error) = paInvalidDevice;
}
// Get the audio client
- hr = ActivateAudioInterface(pInfo, &audioClient);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioInterface(pInfo, &audioClient)))
{
(*pa_error) = paInsufficientMemory;
LogHostError(hr);
// Check for Mono <<>> Stereo workaround
if ((params->channelCount == 1) && (pSub->wavex.Format.nChannels == 2))
{
- /*if (blocking)
- {
- LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT);
- goto done; // fail, blocking mode not supported
- }*/
-
// select mixer
pSub->monoMixer = GetMonoToStereoMixer(&pSub->wavex, (pInfo->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1_L));
if (pSub->monoMixer == NULL)
}
}
-#if 0
- // Add suggestd latency
- framesPerLatency += MakeFramesFromHns(SecondsTonano100(params->suggestedLatency), pSub->wavex.Format.nSamplesPerSec);
-#else
// Calculate host buffer size
if ((pSub->shareMode != AUDCLNT_SHAREMODE_EXCLUSIVE) &&
(!pSub->streamFlags || ((pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) == 0)))
}
else
{
+ #ifdef PA_WASAPI_FORCE_POLL_IF_LARGE_BUFFER
REFERENCE_TIME overall;
+ #endif
// Work 1:1 with user buffer (only polling allows to use >1)
framesPerLatency += MakeFramesFromHns(SecondsTonano100(params->suggestedLatency), pSub->wavex.Format.nSamplesPerSec);
- // Use Polling if overall latency is >= 21.33ms as it allows to use 100% CPU in a callback,
- // or user specified latency parameter
+ // Force Polling if overall latency is >= 21.33ms as it allows to use 100% CPU in a callback,
+ // or user specified latency parameter.
+ #ifdef PA_WASAPI_FORCE_POLL_IF_LARGE_BUFFER
overall = MakeHnsPeriod(framesPerLatency, pSub->wavex.Format.nSamplesPerSec);
if (overall >= (106667 * 2)/*21.33ms*/)
{
pSub->streamFlags &= ~AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
PRINT(("WASAPI: CreateAudioClient: forcing POLL mode\n"));
}
+ #endif
}
-#endif
// For full-duplex output resize buffer to be the same as for input
if (output && fullDuplex)
if (pSub->period < pInfo->DefaultDevicePeriod)
{
pSub->period = pInfo->DefaultDevicePeriod;
+
// Recalculate aligned period
framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
if (pSub->period < pInfo->MinimumDevicePeriod)
{
pSub->period = pInfo->MinimumDevicePeriod;
+
// Recalculate aligned period
framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_FWD);
static const REFERENCE_TIME MAX_BUFFER_EVENT_DURATION = 500 * 10000;
static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
- if (pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) // pull mode, max 500ms
+ // Pull mode, max 500ms
+ if (pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK)
{
if (pSub->period > MAX_BUFFER_EVENT_DURATION)
{
pSub->period = MAX_BUFFER_EVENT_DURATION;
+
// Recalculate aligned period
framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
}
}
- else // push mode, max 2000ms
+ // Push mode, max 2000ms
+ else
{
if (pSub->period > MAX_BUFFER_POLL_DURATION)
{
pSub->period = MAX_BUFFER_POLL_DURATION;
+
// Recalculate aligned period
framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
break;
}
- hr = IAudioClient2_SetClientProperties((IAudioClient2 *)audioClient, (AudioClientProperties *)&audioProps);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient2_SetClientProperties((IAudioClient2 *)audioClient, (AudioClientProperties *)&audioProps)))
{
PRINT(("WASAPI: IAudioClient2_SetClientProperties(IsOffload = %d, Category = %d, Options = %d) failed\n", audioProps.bIsOffload, audioProps.eCategory, audioProps.Options));
LogHostError(hr);
}
#endif
+ // Set device scheduling period (always 0 in Shared mode according Microsoft docs)
+ _CalculatePeriodicity(pSub, output, &eventPeriodicity);
+
// Open the stream and associate it with an audio session
hr = IAudioClient_Initialize(audioClient,
pSub->shareMode,
pSub->streamFlags,
pSub->period,
- (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
+ eventPeriodicity,
&pSub->wavex.Format,
NULL);
// [Output only] Check if buffer size is the one we requested in Exclusive mode, for UAC1 USB DACs WASAPI
- // allocates internal buffer equal to 8 times of pSub->period that has to be corrected in order to match
+ // can allocate internal buffer equal to 8 times of pSub->period that has to be corrected in order to match
// the requested latency
- if (output && (hr == S_OK) && (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE))
+ if (output && SUCCEEDED(hr) && (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE))
{
- UINT32 maxBufferSize = 0;
+ UINT32 maxBufferFrames;
- hr = IAudioClient_GetBufferSize(audioClient, &maxBufferSize);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetBufferSize(audioClient, &maxBufferFrames)))
{
(*pa_error) = paInvalidDevice;
LogHostError(hr);
goto done;
}
- // For Exclusive mode for UAC1 devices maxBufferSize will be framesPerLatency * 8 but check any difference
+ // For Exclusive mode for UAC1 devices maxBufferFrames may be framesPerLatency * 8 but check any difference
// to be able to guarantee the latency user requested and also resulted framesPerLatency may be bigger than
// 2 seconds that will cause audio client not operational (GetCurrentPadding() will return always 0)
- if (maxBufferSize >= (framesPerLatency * 2))
+ if (maxBufferFrames >= (framesPerLatency * 2))
{
- UINT32 ratio = maxBufferSize / framesPerLatency;
+ UINT32 ratio = maxBufferFrames / framesPerLatency;
PRINT(("WASAPI: CreateAudioClient: detected %d times larger buffer than requested, correct to match user latency\n", ratio));
SAFE_RELEASE(audioClient);
// Create a new audio client
- hr = ActivateAudioInterface(pInfo, &audioClient);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioInterface(pInfo, &audioClient)))
{
(*pa_error) = paInsufficientMemory;
LogHostError(hr);
goto done;
}
+ // Set device scheduling period (always 0 in Shared mode according Microsoft docs)
+ _CalculatePeriodicity(pSub, output, &eventPeriodicity);
+
// Open the stream and associate it with an audio session
hr = IAudioClient_Initialize(audioClient,
pSub->shareMode,
pSub->streamFlags,
pSub->period,
- (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
+ eventPeriodicity,
&pSub->wavex.Format,
NULL);
}
SAFE_RELEASE(audioClient);
// Create a new audio client
- hr = ActivateAudioInterface(pInfo, &audioClient);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioInterface(pInfo, &audioClient)))
{
(*pa_error) = paInsufficientMemory;
LogHostError(hr);
goto done;
}
+ // Set device scheduling period (always 0 in Shared mode according Microsoft docs)
+ _CalculatePeriodicity(pSub, output, &eventPeriodicity);
+
// Open the stream and associate it with an audio session
hr = IAudioClient_Initialize(audioClient,
pSub->shareMode,
pSub->streamFlags,
pSub->period,
- (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
+ eventPeriodicity,
&pSub->wavex.Format,
NULL);
}
- /*! WASAPI buffer size failure. Fallback to using default size.
+ /*! WASAPI buffer size or alignment failure. Fallback to using default size and alignment.
*/
- if (hr == AUDCLNT_E_BUFFER_SIZE_ERROR)
+ if ((hr == AUDCLNT_E_BUFFER_SIZE_ERROR) || (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED))
{
// Use default
pSub->period = pInfo->DefaultDevicePeriod;
- PRINT(("WASAPI: CreateAudioClient: correcting buffer size to device default\n"));
+ PRINT(("WASAPI: CreateAudioClient: correcting buffer size/alignment to device default\n"));
// Release the previous allocations
SAFE_RELEASE(audioClient);
// Create a new audio client
- hr = ActivateAudioInterface(pInfo, &audioClient);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioInterface(pInfo, &audioClient)))
{
(*pa_error) = paInsufficientMemory;
LogHostError(hr);
goto done;
}
+ // Set device scheduling period (always 0 in Shared mode according Microsoft docs)
+ _CalculatePeriodicity(pSub, output, &eventPeriodicity);
+
// Open the stream and associate it with an audio session
hr = IAudioClient_Initialize(audioClient,
pSub->shareMode,
pSub->streamFlags,
pSub->period,
- (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
+ eventPeriodicity,
&pSub->wavex.Format,
NULL);
}
- /*! If the requested buffer size is not aligned. Can be triggered by Windows 7 and up.
- Should not be be triggered ever as we do align buffers always with _CalculateAlignedPeriod.
- */
- if (hr == AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED)
- {
- UINT32 frames = 0;
-
- // Get the next aligned frame
- hr = IAudioClient_GetBufferSize(audioClient, &frames);
- if (hr != S_OK)
- {
- (*pa_error) = paInvalidDevice;
- LogHostError(hr);
- goto done;
- }
-
- PRINT(("WASAPI: CreateAudioClient: aligning buffer size to % frames\n", frames));
-
- // Release the previous allocations
- SAFE_RELEASE(audioClient);
-
- // Create a new audio client
- hr = ActivateAudioInterface(pInfo, &audioClient);
- if (hr != S_OK)
- {
- (*pa_error) = paInsufficientMemory;
- LogHostError(hr);
- goto done;
- }
-
- // Get closest format
- if ((error = GetClosestFormat(audioClient, sampleRate, params, pSub->shareMode, &pSub->wavex, output)) != paFormatIsSupported)
- {
- (*pa_error) = error;
- LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT); // fail, format not supported
- goto done;
- }
-
- // Check for Mono >> Stereo workaround
- if ((params->channelCount == 1) && (pSub->wavex.Format.nChannels == 2))
- {
- /*if (blocking)
- {
- LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT);
- goto done; // fail, blocking mode not supported
- }*/
-
- // Select mixer
- pSub->monoMixer = GetMonoToStereoMixer(&pSub->wavex, (pInfo->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1_L));
- if (pSub->monoMixer == NULL)
- {
- (*pa_error) = paInvalidChannelCount;
- LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT);
- goto done; // fail, no mixer for format
- }
- }
-
- // Calculate period
- pSub->period = MakeHnsPeriod(frames, pSub->wavex.Format.nSamplesPerSec);
-
- // Open the stream and associate it with an audio session
- hr = IAudioClient_Initialize(audioClient,
- pSub->shareMode,
- pSub->streamFlags,
- pSub->period,
- (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
- &pSub->wavex.Format,
- NULL);
- if (hr != S_OK)
- {
- (*pa_error) = paInvalidDevice;
- LogHostError(hr);
- goto done;
- }
- }
- else
- if (hr != S_OK)
+ // Error has no workaround, fail completely
+ if (FAILED(hr))
{
(*pa_error) = paInvalidDevice;
LogHostError(hr);
IAudioClient_AddRef(pSub->clientParent);
// Recalculate buffers count
- _RecalculateBuffersCount(pSub,
- userFramesPerBuffer,
- MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec),
+ _RecalculateBuffersCount(pSub, userFramesPerBuffer, MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec),
fullDuplex);
// No error, client is succesfully created
{
HRESULT hr;
PaError result;
-
- UINT32 maxBufferSize = 0;
- PaTime buffer_latency = 0;
- UINT32 framesPerBuffer = stream->out.params.frames_per_buffer;
+ UINT32 maxBufferSize;
+ PaTime bufferLatency;
+ const UINT32 framesPerBuffer = stream->out.params.frames_per_buffer;
// Create Audio client
- hr = CreateAudioClient(stream, &stream->out, TRUE, &result);
- if (hr != S_OK)
+ if (FAILED(hr = CreateAudioClient(stream, &stream->out, TRUE, &result)))
{
LogPaError(result);
goto error;
return paInvalidDevice;*/
// Get max possible buffer size to check if it is not less than that we request
- hr = IAudioClient_GetBufferSize(stream->out.clientParent, &maxBufferSize);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetBufferSize(stream->out.clientParent, &maxBufferSize)))
{
LogHostError(hr);
LogPaError(result = paInvalidDevice);
// Correct buffer to max size if it maxed out result of GetBufferSize
stream->out.bufferSize = maxBufferSize;
- // Get interface latency (actually uneeded as we calculate latency from the size
- // of maxBufferSize).
- hr = IAudioClient_GetStreamLatency(stream->out.clientParent, &stream->out.deviceLatency);
- if (hr != S_OK)
+ // Get interface latency (actually uneeded 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);
(stream->out.userBufferAndHostMatch ? stream->out.framesPerHostCallback : framesPerBuffer);
// Calculate buffer latency
- buffer_latency = (PaTime)maxBufferSize / stream->out.wavex.Format.nSamplesPerSec;
+ bufferLatency = (PaTime)maxBufferSize / stream->out.wavex.Format.nSamplesPerSec;
// Append buffer latency to interface latency in shared mode (see GetStreamLatency notes)
- stream->out.latencySeconds = buffer_latency;
+ stream->out.latencySeconds = bufferLatency;
PRINT(("WASAPI::OpenStream(output): framesPerUser[ %d ] framesPerHost[ %d ] latency[ %.02fms ] exclusive[ %s ] wow64_fix[ %s ] mode[ %s ]\n", (UINT32)framesPerBuffer, (UINT32)stream->out.framesPerHostCallback, (float)(stream->out.latencySeconds*1000.0f), (stream->out.shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? "YES" : "NO"), (stream->out.params.wow64_workaround ? "YES" : "NO"), (stream->out.streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK ? "EVENT" : "POLL")));
{
HRESULT hr;
PaError result;
-
- UINT32 maxBufferSize = 0;
- PaTime buffer_latency = 0;
- UINT32 framesPerBuffer = stream->in.params.frames_per_buffer;
+ UINT32 maxBufferSize;
+ PaTime bufferLatency;
+ const UINT32 framesPerBuffer = stream->in.params.frames_per_buffer;
// Create Audio client
- hr = CreateAudioClient(stream, &stream->in, FALSE, &result);
- if (hr != S_OK)
+ if (FAILED(hr = CreateAudioClient(stream, &stream->in, FALSE, &result)))
{
LogPaError(result);
goto error;
return paInvalidDevice;*/
// Get max possible buffer size to check if it is not less than that we request
- hr = IAudioClient_GetBufferSize(stream->in.clientParent, &maxBufferSize);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetBufferSize(stream->in.clientParent, &maxBufferSize)))
{
LogHostError(hr);
LogPaError(result = paInvalidDevice);
// Get interface latency (actually uneeded as we calculate latency from the size
// of maxBufferSize).
- hr = IAudioClient_GetStreamLatency(stream->in.clientParent, &stream->in.deviceLatency);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetStreamLatency(stream->in.clientParent, &stream->in.deviceLatency)))
{
LogHostError(hr);
LogPaError(result = paInvalidDevice);
(stream->in.userBufferAndHostMatch ? stream->in.framesPerHostCallback : framesPerBuffer);
// Calculate buffer latency
- buffer_latency = (PaTime)maxBufferSize / stream->in.wavex.Format.nSamplesPerSec;
+ bufferLatency = (PaTime)maxBufferSize / stream->in.wavex.Format.nSamplesPerSec;
// Append buffer latency to interface latency in shared mode (see GetStreamLatency notes)
- stream->in.latencySeconds = buffer_latency;
+ stream->in.latencySeconds = bufferLatency;
PRINT(("WASAPI::OpenStream(input): framesPerUser[ %d ] framesPerHost[ %d ] latency[ %.02fms ] exclusive[ %s ] wow64_fix[ %s ] mode[ %s ]\n", (UINT32)framesPerBuffer, (UINT32)stream->in.framesPerHostCallback, (float)(stream->in.latencySeconds*1000.0f), (stream->in.shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? "YES" : "NO"), (stream->in.params.wow64_workaround ? "YES" : "NO"), (stream->in.streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK ? "EVENT" : "POLL")));
stream->in.params.wow64_workaround = paWasapi->useWOW64Workaround;
// Create and activate audio client
- hr = ActivateAudioClientInput(stream);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioClientInput(stream)))
{
LogPaError(result = paInvalidDevice);
goto error;
}
// Only get IAudioCaptureClient input once here instead of getting it at multiple places based on the use
- hr = IAudioClient_GetService(stream->in.clientParent, &pa_IID_IAudioCaptureClient, (void **)&stream->captureClientParent);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetService(stream->in.clientParent, &pa_IID_IAudioCaptureClient, (void **)&stream->captureClientParent)))
{
LogHostError(hr);
LogPaError(result = paUnanticipatedHostError);
stream->out.params.wow64_workaround = paWasapi->useWOW64Workaround;
// Create and activate audio client
- hr = ActivateAudioClientOutput(stream);
- if (hr != S_OK)
+ if (FAILED(hr = ActivateAudioClientOutput(stream)))
{
LogPaError(result = paInvalidDevice);
goto error;
}
// Only get IAudioCaptureClient output once here instead of getting it at multiple places based on the use
- hr = IAudioClient_GetService(stream->out.clientParent, &pa_IID_IAudioRenderClient, (void **)&stream->renderClientParent);
- if (hr != S_OK)
+ if (FAILED(hr = IAudioClient_GetService(stream->out.clientParent, &pa_IID_IAudioRenderClient, (void **)&stream->renderClientParent)))
{
LogHostError(hr);
LogPaError(result = paUnanticipatedHostError);
// Get buffer
if ((hr = IAudioRenderClient_GetBuffer(stream->renderClient, frames, &data)) != S_OK)
{
+ // Both modes, Shared and Exclusive, can fail with AUDCLNT_E_BUFFER_TOO_LARGE error
+ #if 0
if (stream->out.shareMode == AUDCLNT_SHAREMODE_SHARED)
{
// Using GetCurrentPadding to overcome AUDCLNT_E_BUFFER_TOO_LARGE in
// shared mode results in no sound in Event-driven mode (MSDN does not
// document this, or is it WASAPI bug?), thus we better
// try to acquire buffer next time when GetBuffer allows to do so.
-#if 0
+ #if 0
// Get Read position
UINT32 padding = 0;
hr = IAudioClient_GetCurrentPadding(stream->out.clientProc, &padding);
if ((hr = IAudioRenderClient_GetBuffer(stream->renderClient, frames, &data)) != S_OK)
return LogHostError(hr);
-#else
+ #else
if (hr == AUDCLNT_E_BUFFER_TOO_LARGE)
return S_OK; // be silent in shared mode, try again next time
-#endif
+ #endif
}
else
return LogHostError(hr);
+ #else
+ if (hr == AUDCLNT_E_BUFFER_TOO_LARGE)
+ return S_OK; // try again next time
+
+ return LogHostError(hr);
+ #endif
}
// Process data
ThreadIdleScheduler scheduler;
// Calculate the actual duration of the allocated buffer.
- DWORD sleep_ms = 0;
+ DWORD sleep_ms = 0;
DWORD sleep_ms_in;
DWORD sleep_ms_out;
}
else
{
- if (frames != 0)
- {
- // USB DACs report large buffer in Exclusive mode and if it is filled fully will stuck in
- // non playing state, e.g. IAudioClient_GetCurrentPadding() will start reporting max buffer size
- // constantly, thus preload data size equal to the user buffer to allow process going
- if ((stream->out.shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) && (frames > stream->out.framesPerBuffer))
- frames = stream->out.framesPerBuffer;
+ // Some devices may not start (will get stuck with 0 ready frames) if data not prefetched
+ if (frames == 0)
+ frames = stream->out.framesPerBuffer;
- if ((hr = ProcessOutputBuffer(stream, processor, frames)) != S_OK)
- {
- LogHostError(hr); // not fatal, just log
- }
+ // USB DACs report large buffer in Exclusive mode and if it is filled fully will stuck in
+ // non playing state, e.g. IAudioClient_GetCurrentPadding() will start reporting max buffer size
+ // constantly, thus preload data size equal to the user buffer to allow process going
+ if ((stream->out.shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) && (frames >= (stream->out.framesPerBuffer * 2)))
+ frames -= stream->out.framesPerBuffer;
+
+ if ((hr = ProcessOutputBuffer(stream, processor, frames)) != S_OK)
+ {
+ LogHostError(hr); // not fatal, just log
}
}
}
if (!PA_WASAPI__IS_FULLDUPLEX(stream))
{
- // Processing Loop
UINT32 next_sleep = sleep_ms;
+
+ // Processing Loop
while (WaitForSingleObject(stream->hCloseRequest, next_sleep) == WAIT_TIMEOUT)
{
// Get next sleep time
if (sleep_ms == 0)
- {
next_sleep = ThreadIdleScheduler_NextSleep(&scheduler);
- }
for (i = 0; i < S_COUNT; ++i)
{
if (stream->renderClient == NULL)
break;
- // get available frames
+ // Get available frames
if ((hr = _PollGetOutputFramesAvailable(stream, &frames)) != S_OK)
{
LogHostError(hr);
goto thread_error;
}
- // output
+ // Output data to the user callback
if (stream->bufferMode == paUtilFixedHostBufferSize)
{
while (frames >= stream->out.framesPerBuffer)
}
}
else
+ if (frames != 0)
{
- if (frames != 0)
+ if ((hr = ProcessOutputBuffer(stream, processor, frames)) != S_OK)
{
- if ((hr = ProcessOutputBuffer(stream, processor, frames)) != S_OK)
- {
- LogHostError(hr);
- goto thread_error;
- }
+ LogHostError(hr);
+ goto thread_error;
}
}
projects / portaudio / commitdiff