wasapi:
- fixed Exclusive/Event mode if latency used exceeds 100ms resulting in bad audio - fixed Exclusive mode if overall stream latency exceeds 1400-2000ms resulting in E_OUTOFMEMORY even under Vista, now PA WASAPI will gradually decrease overall latency until acceptable level - applied device period aligning if period was corrected - improved robustness of audio client creation
This commit is contained in:
dmitrykos
parent
0a4cf33379
commit
f7ee62732e
1 changed files with 211 additions and 130 deletions
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@ -708,6 +708,9 @@ static UINT32 MakeFramesFromHns(REFERENCE_TIME hnsPeriod, UINT32 nSamplesPerSec)
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return nFrames;
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}
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// Aligning function type
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typedef UINT32 (*ALIGN_FUNC) (UINT32 v, UINT32 align);
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// ------------------------------------------------------------------------------------------
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// Aligns 'v' backwards
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static UINT32 ALIGN_BWD(UINT32 v, UINT32 align)
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@ -729,7 +732,8 @@ static UINT32 ALIGN_FWD(UINT32 v, UINT32 align)
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// Aligns WASAPI buffer to 128 byte packet boundary. HD Audio will fail to play if buffer
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// is misaligned. This problem was solved in Windows 7 were AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED
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// is thrown although we must align for Vista anyway.
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static UINT32 AlignFramesPerBuffer(UINT32 nFrames, UINT32 nSamplesPerSec, UINT32 nBlockAlign)
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static UINT32 AlignFramesPerBuffer(UINT32 nFrames, UINT32 nSamplesPerSec, UINT32 nBlockAlign,
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ALIGN_FUNC pAlignFunc)
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{
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#define HDA_PACKET_SIZE (128)
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@ -737,7 +741,7 @@ static UINT32 AlignFramesPerBuffer(UINT32 nFrames, UINT32 nSamplesPerSec, UINT32
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long packets;
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// align to packet size
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frame_bytes = ALIGN_FWD(frame_bytes, HDA_PACKET_SIZE); // use ALIGN_FWD if bigger but safer period is more desired
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frame_bytes = pAlignFunc(frame_bytes, HDA_PACKET_SIZE); // use ALIGN_FWD if bigger but safer period is more desired
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nFrames = frame_bytes / nBlockAlign;
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packets = frame_bytes / HDA_PACKET_SIZE;
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@ -1937,24 +1941,66 @@ static PaUint32 PaUtil_GetFramesPerHostBuffer(PaUint32 userFramesPerBuffer, PaTi
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}
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// ------------------------------------------------------------------------------------------
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static HRESULT CreateAudioClient(PaWasapiSubStream *pSubStream, PaWasapiDeviceInfo *info,
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const PaStreamParameters *params, UINT32 framesPerLatency, double sampleRate, UINT32 streamFlags,
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BOOL blocking, BOOL output, BOOL full_duplex, PaError *pa_error)
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static void _RecalculateBuffersCount(PaWasapiSubStream *sub, UINT32 userFramesPerBuffer, UINT32 framesPerLatency, BOOL fullDuplex)
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{
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// Count buffers (must be at least 1)
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sub->buffers = (userFramesPerBuffer ? framesPerLatency / userFramesPerBuffer : 0);
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if (sub->buffers == 0)
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sub->buffers = 1;
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// Determine amount of buffers used:
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// - Full-duplex mode will lead to period difference, thus only 1.
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// - Input mode, only 1, as WASAPI allows extraction of only 1 packet.
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// - For Shared mode we use double buffering.
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if ((sub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) || fullDuplex)
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{
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// Exclusive mode does not allow >1 buffers be used for Event interface, e.g. GetBuffer
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// call must acquire max buffer size and it all must be processed.
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if (sub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK)
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sub->userBufferAndHostMatch = 1;
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// Use paUtilBoundedHostBufferSize because exclusive mode will starve and produce
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// bad quality of audio
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sub->buffers = 1;
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}
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}
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// ------------------------------------------------------------------------------------------
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static void _CalculateAlignedPeriod(PaWasapiSubStream *pSub, UINT32 *nFramesPerLatency,
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ALIGN_FUNC pAlignFunc)
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{
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// Align frames to HD Audio packet size of 128 bytes for Exclusive mode only.
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// Not aligning on Windows Vista will cause Event timeout, although Windows 7 will
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// return AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED error to realign buffer. Aligning is necessary
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// for Exclusive mode only! when audio data is feeded directly to hardware.
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if (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
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{
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(*nFramesPerLatency) = AlignFramesPerBuffer((*nFramesPerLatency),
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pSub->wavex.Format.nSamplesPerSec, pSub->wavex.Format.nBlockAlign, pAlignFunc);
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}
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// Calculate period
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pSub->period = MakeHnsPeriod((*nFramesPerLatency), pSub->wavex.Format.nSamplesPerSec);
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}
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// ------------------------------------------------------------------------------------------
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static HRESULT CreateAudioClient(PaWasapiSubStream *pSub, PaWasapiDeviceInfo *pInfo,
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const PaStreamParameters *params, UINT32 framesPerLatency, double sampleRate,
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BOOL blocking, BOOL output, BOOL fullDuplex, PaError *pa_error)
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{
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PaError error;
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HRESULT hr = S_OK;
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UINT32 nFrames = 0;
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UINT32 userFramesPerBuffer = framesPerLatency;
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IAudioClient *pAudioClient = NULL;
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double suggestedLatency = 0.0;
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HRESULT hr;
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const UINT32 userFramesPerBuffer = framesPerLatency;
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IAudioClient *audioClient = NULL;
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if (!pSubStream || !info || !params)
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// Validate parameters
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if (!pSub || !pInfo || !params)
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return E_POINTER;
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if ((UINT32)sampleRate == 0)
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return E_INVALIDARG;
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// Get the audio client
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hr = IMMDevice_Activate(info->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&pAudioClient);
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hr = IMMDevice_Activate(pInfo->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void **)&audioClient);
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if (hr != S_OK)
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{
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LogHostError(hr);
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@ -1962,7 +2008,7 @@ static HRESULT CreateAudioClient(PaWasapiSubStream *pSubStream, PaWasapiDeviceIn
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}
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// Get closest format
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if ((error = GetClosestFormat(pAudioClient, sampleRate, params, pSubStream->shareMode, &pSubStream->wavex, output)) != paFormatIsSupported)
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if ((error = GetClosestFormat(audioClient, sampleRate, params, pSub->shareMode, &pSub->wavex, output)) != paFormatIsSupported)
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{
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if (pa_error)
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(*pa_error) = error;
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@ -1972,7 +2018,7 @@ static HRESULT CreateAudioClient(PaWasapiSubStream *pSubStream, PaWasapiDeviceIn
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}
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// Check for Mono >> Stereo workaround
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if ((params->channelCount == 1) && (pSubStream->wavex.Format.nChannels == 2))
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if ((params->channelCount == 1) && (pSub->wavex.Format.nChannels == 2))
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{
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if (blocking)
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{
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@ -1981,8 +2027,8 @@ static HRESULT CreateAudioClient(PaWasapiSubStream *pSubStream, PaWasapiDeviceIn
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}
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// select mixer
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pSubStream->monoMixer = _GetMonoToStereoMixer(params->sampleFormat, (info->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1));
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if (pSubStream->monoMixer == NULL)
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pSub->monoMixer = _GetMonoToStereoMixer(params->sampleFormat, (pInfo->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1));
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if (pSub->monoMixer == NULL)
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{
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LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT);
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goto done; // fail, no mixer for format
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@ -1990,122 +2036,129 @@ static HRESULT CreateAudioClient(PaWasapiSubStream *pSubStream, PaWasapiDeviceIn
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}
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#if 0
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// Correct latency to default device period (in Exclusive mode this is 10ms) if user selected 0
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suggestedLatency = params->suggestedLatency;
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// Add suggestd latency
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framesPerLatency += MakeFramesFromHns(SecondsTonano100(suggestedLatency), pSubStream->wavex.Format.nSamplesPerSec);
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framesPerLatency += MakeFramesFromHns(SecondsTonano100(params->suggestedLatency), pSub->wavex.Format.nSamplesPerSec);
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#else
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// Calculate host buffer size
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if ((pSubStream->shareMode != AUDCLNT_SHAREMODE_EXCLUSIVE) &&
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(!streamFlags || ((streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) == 0)))
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if ((pSub->shareMode != AUDCLNT_SHAREMODE_EXCLUSIVE) &&
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(!pSub->streamFlags || ((pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) == 0)))
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{
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framesPerLatency = PaUtil_GetFramesPerHostBuffer(framesPerLatency,
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params->suggestedLatency, pSubStream->wavex.Format.nSamplesPerSec, 0/*,
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(streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK ? 0 : 1)*/);
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framesPerLatency = PaUtil_GetFramesPerHostBuffer(userFramesPerBuffer,
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params->suggestedLatency, pSub->wavex.Format.nSamplesPerSec, 0/*,
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(pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK ? 0 : 1)*/);
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}
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else
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{
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// do nothing, work 1:1 with buffer (only polling allows to work with >1)
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REFERENCE_TIME overall;
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// Work 1:1 with user buffer (only polling allows to use >1)
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framesPerLatency += MakeFramesFromHns(SecondsTonano100(params->suggestedLatency), pSub->wavex.Format.nSamplesPerSec);
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// Use Polling if overall latency is > 5ms as it allows to use 100% CPU in a callback,
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// or user specified latency parameter
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overall = MakeHnsPeriod(framesPerLatency, pSub->wavex.Format.nSamplesPerSec);
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if ((overall > 50000) || (params->suggestedLatency > 0))
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{
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framesPerLatency = PaUtil_GetFramesPerHostBuffer(userFramesPerBuffer,
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params->suggestedLatency, pSub->wavex.Format.nSamplesPerSec, 0/*,
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(streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK ? 0 : 1)*/);
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// Use Polling interface
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pSub->streamFlags &= ~AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
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}
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}
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#endif
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// Count buffers (must be at least 1)
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pSubStream->buffers = (userFramesPerBuffer ? framesPerLatency / userFramesPerBuffer : 0);
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if (pSubStream->buffers == 0)
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pSubStream->buffers = 1;
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// Determine amount of buffers to be used:
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// - Exclusive mode does not allow >1 buffers be used, e.g. GetBuffer call must acquire
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// max buffer size and it all must be processed.
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// - Full-duplex mode will lead to period difference, thus only 1.
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// - Input mode, only 1, as WASAPI allows extraction of only 1 packet.
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// - For Shared mode we use safe double buffering (2 buffers).
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if ((pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE) || full_duplex/* && output*/)
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{
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pSubStream->userBufferAndHostMatch = 1;
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pSubStream->buffers = 1;
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}
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// Avoid 0 frames
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if (framesPerLatency == 0)
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framesPerLatency = MakeFramesFromHns(info->DefaultDevicePeriod, pSubStream->wavex.Format.nSamplesPerSec);
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framesPerLatency = MakeFramesFromHns(pInfo->DefaultDevicePeriod, pSub->wavex.Format.nSamplesPerSec);
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// Align frames to HD Audio packet size of 128 bytes for Exclusive mode only.
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// Not aligning on Windows Vista will cause Event timeout, although Windows 7 will
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// return AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED error to realign buffer. Aligning is necessary
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// for Exclusive mode only! when audio data is feeded directly to hardware.
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if (pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
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// Calculate aligned period
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_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_FWD);
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/*! Enforce min/max period for device in Shared mode to avoid bad audio quality.
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Avoid doing so for Exclusive mode as alignment will suffer.
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*/
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if (pSub->shareMode == AUDCLNT_SHAREMODE_SHARED)
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{
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framesPerLatency = AlignFramesPerBuffer(framesPerLatency,
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pSubStream->wavex.Format.nSamplesPerSec, pSubStream->wavex.Format.nBlockAlign);
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if (pSub->period < pInfo->DefaultDevicePeriod)
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pSub->period = pInfo->DefaultDevicePeriod;
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}
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else
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{
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if (pSub->period < pInfo->MinimumDevicePeriod)
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pSub->period = pInfo->MinimumDevicePeriod;
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}
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// Calculate period
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pSubStream->period = MakeHnsPeriod(framesPerLatency, pSubStream->wavex.Format.nSamplesPerSec);
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// Enforce min/max period for device in Shared mode to avoid distorted sound.
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// Avoid doing so for Exclusive mode as alignment will suffer. Exclusive mode processes
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// big buffers without problem. Push Exclusive beyond limits if possible.
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if (pSubStream->shareMode == AUDCLNT_SHAREMODE_SHARED)
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/*! Windows 7 does not allow to set latency lower than minimal device period and will
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return error: AUDCLNT_E_INVALID_DEVICE_PERIOD. Under Vista we enforce the same behavior
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manually for unified behavior on all platforms.
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*/
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{
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if (pSubStream->period < info->DefaultDevicePeriod)
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pSubStream->period = info->DefaultDevicePeriod;
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}
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// Windows 7 does not allow to set latency lower than minimal device period and will
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// return error: AUDCLNT_E_INVALID_DEVICE_PERIOD. Under Vista though this error is not implemented
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// allowing to hack WASAPI device with lower period resulting in possibly lower latency.
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if (GetWindowsVersion() & WINDOWS_7_SERVER2008R2_AND_UP)
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{
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if (pSubStream->period < info->MinimumDevicePeriod)
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pSubStream->period = info->MinimumDevicePeriod;
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/*
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AUDCLNT_E_BUFFER_SIZE_ERROR: Applies to Windows 7 and later.
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/*! AUDCLNT_E_BUFFER_SIZE_ERROR: Applies to Windows 7 and later.
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Indicates that the buffer duration value requested by an exclusive-mode client is
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out of range. The requested duration value for pull mode must not be greater than
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500 milliseconds; for push mode the duration value must not be greater than 2 seconds.
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*/
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if (pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
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if (pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
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{
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static const REFERENCE_TIME MAX_BUFFER_EVENT_DURATION = 500 * 10000;
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static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
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static const REFERENCE_TIME MAX_BUFFER_EVENT_DURATION = 500 * 10000;
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static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
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if (streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) // pull mode, max 500ms
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if (pSub->streamFlags & AUDCLNT_STREAMFLAGS_EVENTCALLBACK) // pull mode, max 500ms
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{
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if (pSubStream->period > MAX_BUFFER_EVENT_DURATION)
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pSubStream->period = MAX_BUFFER_EVENT_DURATION;
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if (pSub->period > MAX_BUFFER_EVENT_DURATION)
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{
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pSub->period = MAX_BUFFER_EVENT_DURATION;
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// Recalculate aligned period
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framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
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_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
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}
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}
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else // push mode, max 2000ms
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else // push mode, max 2000ms
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{
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if (pSubStream->period > MAX_BUFFER_POLL_DURATION)
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pSubStream->period = MAX_BUFFER_POLL_DURATION;
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if (pSub->period > MAX_BUFFER_POLL_DURATION)
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{
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pSub->period = MAX_BUFFER_POLL_DURATION;
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// Recalculate aligned period
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framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
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_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
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}
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}
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}
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}
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// Open the stream and associate it with an audio session
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hr = IAudioClient_Initialize(pAudioClient,
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pSubStream->shareMode,
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streamFlags/*AUDCLNT_STREAMFLAGS_EVENTCALLBACK*/,
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pSubStream->period,
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(pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSubStream->period : 0),
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&pSubStream->wavex.Format,
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hr = IAudioClient_Initialize(audioClient,
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pSub->shareMode,
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pSub->streamFlags,
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pSub->period,
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(pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
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&pSub->wavex.Format,
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NULL);
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// This would mean too low latency
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if (hr == AUDCLNT_E_BUFFER_SIZE_ERROR)
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/*! WASAPI is tricky on large device buffer, sometimes 2000ms can be allocated sometimes
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less. There is no known guaranteed level thus we make subsequent tries by decreasing
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buffer by 100ms per try.
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*/
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while ((hr == E_OUTOFMEMORY) && (pSub->period > (100 * 10000)))
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{
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PRINT(("WASAPI: CreateAudioClient: correcting buffer size to device minimum\n"));
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PRINT(("WASAPI: CreateAudioClient: decreasing buffer size to %d milliseconds\n", (pSub->period / 10000)));
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// User was trying to set lowest possible, so set lowest device possible
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pSubStream->period = info->MinimumDevicePeriod;
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// Decrease by 100ms and try again
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pSub->period -= (100 * 10000);
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// Recalculate aligned period
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framesPerLatency = MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec);
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_CalculateAlignedPeriod(pSub, &framesPerLatency, ALIGN_BWD);
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// Release the previous allocations
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SAFE_RELEASE(pAudioClient);
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SAFE_RELEASE(audioClient);
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// Create a new audio client
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hr = IMMDevice_Activate(info->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
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hr = IMMDevice_Activate(pInfo->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&audioClient);
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if (hr != S_OK)
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{
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LogHostError(hr);
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@ -2113,33 +2166,67 @@ static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
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}
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// Open the stream and associate it with an audio session
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hr = IAudioClient_Initialize(pAudioClient,
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pSubStream->shareMode,
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streamFlags/*AUDCLNT_STREAMFLAGS_EVENTCALLBACK*/,
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pSubStream->period,
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(pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSubStream->period : 0),
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&pSubStream->wavex.Format,
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hr = IAudioClient_Initialize(audioClient,
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pSub->shareMode,
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pSub->streamFlags,
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pSub->period,
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(pSub->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSub->period : 0),
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&pSub->wavex.Format,
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NULL);
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}
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// If the requested buffer size is not aligned...
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/*! WASAPI buffer size failure. Fallback to using default size.
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*/
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if (hr == AUDCLNT_E_BUFFER_SIZE_ERROR)
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{
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// Use default
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pSub->period = pInfo->DefaultDevicePeriod;
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PRINT(("WASAPI: CreateAudioClient: correcting buffer size to device default\n"));
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// Release the previous allocations
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SAFE_RELEASE(audioClient);
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// Create a new audio client
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hr = IMMDevice_Activate(pInfo->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&audioClient);
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if (hr != S_OK)
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{
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LogHostError(hr);
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goto done;
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}
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// 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 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)
|
||||
{
|
||||
PRINT(("WASAPI: CreateAudioClient: aligning buffer size\n"));
|
||||
UINT32 frames = 0;
|
||||
|
||||
// Get the next aligned frame
|
||||
hr = IAudioClient_GetBufferSize(pAudioClient, &nFrames);
|
||||
hr = IAudioClient_GetBufferSize(audioClient, &frames);
|
||||
if (hr != S_OK)
|
||||
{
|
||||
LogHostError(hr);
|
||||
goto done;
|
||||
}
|
||||
|
||||
PRINT(("WASAPI: CreateAudioClient: aligning buffer size to % frames\n", frames));
|
||||
|
||||
// Release the previous allocations
|
||||
SAFE_RELEASE(pAudioClient);
|
||||
SAFE_RELEASE(audioClient);
|
||||
|
||||
// Create a new audio client
|
||||
hr = IMMDevice_Activate(info->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
|
||||
hr = IMMDevice_Activate(pInfo->device, &pa_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&audioClient);
|
||||
if (hr != S_OK)
|
||||
{
|
||||
LogHostError(hr);
|
||||
|
|
@ -2147,7 +2234,7 @@ static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
|
|||
}
|
||||
|
||||
// Get closest format
|
||||
if ((error = GetClosestFormat(pAudioClient, sampleRate, params, pSubStream->shareMode, &pSubStream->wavex, output)) != paFormatIsSupported)
|
||||
if ((error = GetClosestFormat(audioClient, sampleRate, params, pSub->shareMode, &pSub->wavex, output)) != paFormatIsSupported)
|
||||
{
|
||||
if (pa_error)
|
||||
(*pa_error) = error;
|
||||
|
|
@ -2157,7 +2244,7 @@ static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
|
|||
}
|
||||
|
||||
// Check for Mono >> Stereo workaround
|
||||
if ((params->channelCount == 1) && (pSubStream->wavex.Format.nChannels == 2))
|
||||
if ((params->channelCount == 1) && (pSub->wavex.Format.nChannels == 2))
|
||||
{
|
||||
if (blocking)
|
||||
{
|
||||
|
|
@ -2165,37 +2252,25 @@ static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
|
|||
goto done; // fail, blocking mode not supported
|
||||
}
|
||||
|
||||
// select mixer
|
||||
pSubStream->monoMixer = _GetMonoToStereoMixer(params->sampleFormat, (info->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1));
|
||||
if (pSubStream->monoMixer == NULL)
|
||||
// Select mixer
|
||||
pSub->monoMixer = _GetMonoToStereoMixer(params->sampleFormat, (pInfo->flow == eRender ? MIX_DIR__1TO2 : MIX_DIR__2TO1));
|
||||
if (pSub->monoMixer == NULL)
|
||||
{
|
||||
LogHostError(hr = AUDCLNT_E_UNSUPPORTED_FORMAT);
|
||||
goto done; // fail, no mixer for format
|
||||
}
|
||||
}
|
||||
|
||||
// Reset buffers count to single
|
||||
pSubStream->buffers = 1;
|
||||
|
||||
// Calculate period
|
||||
pSubStream->period = MakeHnsPeriod(nFrames, pSubStream->wavex.Format.nSamplesPerSec);
|
||||
|
||||
// Windows 7 does not allow to set latency lower than minimal device period and will
|
||||
// return error: AUDCLNT_E_INVALID_DEVICE_PERIOD. Under Vista though this error is not implemented
|
||||
// allowing to hack WASAPI device with lower period resulting in possibly lower latency.
|
||||
if (GetWindowsVersion() & WINDOWS_7_SERVER2008R2_AND_UP)
|
||||
{
|
||||
if (pSubStream->period < info->MinimumDevicePeriod)
|
||||
pSubStream->period = info->MinimumDevicePeriod;
|
||||
}
|
||||
pSub->period = MakeHnsPeriod(frames, pSub->wavex.Format.nSamplesPerSec);
|
||||
|
||||
// Open the stream and associate it with an audio session
|
||||
hr = IAudioClient_Initialize(pAudioClient,
|
||||
pSubStream->shareMode,
|
||||
streamFlags/*AUDCLNT_STREAMFLAGS_EVENTCALLBACK*/,
|
||||
pSubStream->period,
|
||||
(pSubStream->shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE ? pSubStream->period : 0),
|
||||
&pSubStream->wavex.Format,
|
||||
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)
|
||||
{
|
||||
|
|
@ -2210,14 +2285,20 @@ static const REFERENCE_TIME MAX_BUFFER_POLL_DURATION = 2000 * 10000;
|
|||
goto done;
|
||||
}
|
||||
|
||||
// Set result
|
||||
pSubStream->client = pAudioClient;
|
||||
IAudioClient_AddRef(pSubStream->client);
|
||||
// Set client
|
||||
pSub->client = audioClient;
|
||||
IAudioClient_AddRef(pSub->client);
|
||||
|
||||
// Recalculate buffers count
|
||||
_RecalculateBuffersCount(pSub,
|
||||
userFramesPerBuffer,
|
||||
MakeFramesFromHns(pSub->period, pSub->wavex.Format.nSamplesPerSec),
|
||||
fullDuplex);
|
||||
|
||||
done:
|
||||
|
||||
// Clean up
|
||||
SAFE_RELEASE(pAudioClient);
|
||||
SAFE_RELEASE(audioClient);
|
||||
return hr;
|
||||
}
|
||||
|
||||
|
|
@ -2331,7 +2412,7 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
|
|||
|
||||
// Create Audio client
|
||||
hr = CreateAudioClient(&stream->in, info, inputParameters, framesPerBuffer/*framesPerLatency*/,
|
||||
sampleRate, stream->in.streamFlags, (streamCallback == NULL), FALSE, fullDuplex, &result);
|
||||
sampleRate, (streamCallback == NULL), FALSE, fullDuplex, &result);
|
||||
if (hr != S_OK)
|
||||
{
|
||||
LogHostError(hr);
|
||||
|
|
@ -2449,7 +2530,7 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
|
|||
|
||||
// Create Audio client
|
||||
hr = CreateAudioClient(&stream->out, info, outputParameters, framesPerBuffer/*framesPerLatency*/,
|
||||
sampleRate, stream->out.streamFlags, (streamCallback == NULL), TRUE, fullDuplex, &result);
|
||||
sampleRate, (streamCallback == NULL), TRUE, fullDuplex, &result);
|
||||
if (hr != S_OK)
|
||||
{
|
||||
LogHostError(hr);
|
||||
|
|
|
|||
Loading…
Reference in a new issue