wasapi:
- reimplemented full-duplex processing to address changes to latency calculations (please note Exclusive mode due to some unknown reason will likely not be able to provide good audio quality yet) - fixed Pa_GetStreamInfo for WASAPI device if stream is opened as full-duplex (thanks to Reid Bishop for pointing to the bug)
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dmitrykos
parent
1c289f924e
commit
a243ad431c
1 changed files with 150 additions and 7 deletions
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@ -924,7 +924,7 @@ typedef enum EMixerDir { MIX_DIR__1TO2, MIX_DIR__2TO1 } EMixerDir;
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TYPE * __restrict end = to + count;\
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while (to != end)\
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{\
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*to ++ = (TYPE)((float)(from[0] + from[1]) * 0.70710678118654752440084436210485f/*1/sqrt(2)*/);\
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*to ++ = (TYPE)((float)(from[0] + from[1]) * 0.5f);\
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from += 2;\
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}
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@ -2093,6 +2093,10 @@ static HRESULT CreateAudioClient(PaWasapiStream *pStream, PaWasapiSubStream *pSu
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}
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#endif
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// For full-duplex output resize buffer to be the same as for input
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if (output && fullDuplex)
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framesPerLatency = pStream->in.framesPerHostCallback;
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// Avoid 0 frames
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if (framesPerLatency == 0)
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framesPerLatency = MakeFramesFromHns(pInfo->DefaultDevicePeriod, pSub->wavex.Format.nSamplesPerSec);
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@ -2663,13 +2667,16 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
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if (outputParameters && inputParameters)
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{
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// serious problem #1
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if (stream->in.period != stream->out.period)
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// serious problem #1 - No, Not a problem, especially concerning Exclusive mode.
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// Input device in exclusive mode somehow is getting large buffer always, thus we
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// adjust Output latency to reflect it, thus period will differ but playback will be
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// normal.
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/*if (stream->in.period != stream->out.period)
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{
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PRINT(("WASAPI: OpenStream: period discrepancy\n"));
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LogPaError(result = paBadIODeviceCombination);
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goto error;
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}
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}*/
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// serious problem #2 - No, Not a problem, as framesPerHostCallback take into account
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// sample size while it is not a problem for PA full-duplex, we must care of
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@ -2723,12 +2730,12 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
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// Set Input latency
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stream->streamRepresentation.streamInfo.inputLatency =
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PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor)
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((double)PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor) / sampleRate)
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+ ((inputParameters)?stream->in.latency_seconds : 0);
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// Set Output latency
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stream->streamRepresentation.streamInfo.outputLatency =
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PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor)
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((double)PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor) / sampleRate)
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+ ((outputParameters)?stream->out.latency_seconds : 0);
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// Set SR
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@ -3948,8 +3955,9 @@ PA_THREAD_FUNC ProcThreadPoll(void *param)
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}
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else
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{
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#if 0
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// Processing Loop
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while (WaitForSingleObject(stream->hCloseRequest, sleep_ms) == WAIT_TIMEOUT)
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while (WaitForSingleObject(stream->hCloseRequest, 1) == WAIT_TIMEOUT)
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{
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UINT32 i_frames = 0, i_processed = 0;
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BYTE *i_data = NULL, *o_data = NULL, *o_data_host = NULL;
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@ -4056,6 +4064,141 @@ PA_THREAD_FUNC ProcThreadPoll(void *param)
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break;
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}
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}
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#else
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// Processing Loop
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//sleep_ms = 1;
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while (WaitForSingleObject(stream->hCloseRequest, sleep_ms) == WAIT_TIMEOUT)
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{
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UINT32 i_frames = 0, i_processed = 0;
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BYTE *i_data = NULL, *o_data = NULL, *o_data_host = NULL;
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DWORD i_flags = 0;
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UINT32 o_frames = 0;
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//BOOL repeat = FALSE;
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// going below 1 msec resolution, switching between 1 ms and no waiting
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//if (stream->in.shareMode == AUDCLNT_SHAREMODE_EXCLUSIVE)
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// sleep_ms = !sleep_ms;
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// get available frames
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if ((hr = PollGetOutputFramesAvailable(stream, &o_frames)) != S_OK)
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{
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LogHostError(hr);
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break;
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}
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while (o_frames != 0)
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{
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// get host input buffer
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if ((hr = IAudioCaptureClient_GetBuffer(stream->cclient, &i_data, &i_frames, &i_flags, NULL, NULL)) != S_OK)
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{
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if (hr == AUDCLNT_S_BUFFER_EMPTY)
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break; // no data in capture buffer
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LogHostError(hr);
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break;
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}
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//PA_DEBUG(("full-duplex: o_frames[%d] i_frames[%d] repeat[%d]\n", o_frames, i_frames, repeat));
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//repeat = TRUE;
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// process equal ammount of frames
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if (o_frames >= i_frames)
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{
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// process input ammount of frames
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UINT32 o_processed = i_frames;
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// get host output buffer
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if ((hr = IAudioRenderClient_GetBuffer(stream->rclient, o_processed, &o_data)) == S_OK)
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{
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// processed amount of i_frames
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i_processed = i_frames;
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o_data_host = o_data;
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// convert output mono
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if (stream->out.monoMixer)
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{
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#define __DIV_8(v) ((v) >> 3) //!< (v / 8)
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UINT32 mono_frames_size = o_processed * __DIV_8(stream->out.wavex.Format.wBitsPerSample);
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#undef __DIV_8
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// expand buffer (one way only for better performance due to no calls to realloc)
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if (mono_frames_size > stream->out.monoBufferSize)
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{
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stream->out.monoBuffer = realloc(stream->out.monoBuffer, (stream->out.monoBufferSize = mono_frames_size));
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if (stream->out.monoBuffer == NULL)
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{
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LogPaError(paInsufficientMemory);
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goto thread_error;
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}
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}
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// replace buffer pointer
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o_data = (BYTE *)stream->out.monoBuffer;
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}
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// convert input mono
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if (stream->in.monoMixer)
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{
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#define __DIV_8(v) ((v) >> 3) //!< (v / 8)
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UINT32 mono_frames_size = i_processed * __DIV_8(stream->in.wavex.Format.wBitsPerSample);
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#undef __DIV_8
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// expand buffer (one way only for better performance due to no calls to realloc)
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if (mono_frames_size > stream->in.monoBufferSize)
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{
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stream->in.monoBuffer = realloc(stream->in.monoBuffer, (stream->in.monoBufferSize = mono_frames_size));
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if (stream->in.monoBuffer == NULL)
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{
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LogPaError(paInsufficientMemory);
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goto thread_error;
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}
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}
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// mix 2 to 1 input channels
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stream->in.monoMixer(stream->in.monoBuffer, i_data, i_processed);
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// replace buffer pointer
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i_data = (BYTE *)stream->in.monoBuffer;
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}
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// process
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processor[S_FULLDUPLEX].processor(i_data, i_processed, o_data, o_processed, processor[S_FULLDUPLEX].userData);
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// mix 1 to 2 output channels
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if (stream->out.monoBuffer)
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stream->out.monoMixer(o_data_host, stream->out.monoBuffer, o_processed);
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// release host output buffer
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if ((hr = IAudioRenderClient_ReleaseBuffer(stream->rclient, o_processed, 0)) != S_OK)
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LogHostError(hr);
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o_frames -= o_processed;
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}
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else
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{
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if (stream->out.shareMode != AUDCLNT_SHAREMODE_SHARED)
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LogHostError(hr); // be silent in shared mode, try again next time
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}
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}
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else
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{
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i_processed = 0;
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goto fd_release_buffer_in;
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}
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fd_release_buffer_in:
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// release host input buffer
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if ((hr = IAudioCaptureClient_ReleaseBuffer(stream->cclient, i_processed)) != S_OK)
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{
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LogHostError(hr);
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break;
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}
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// break processing, input hasn't been accumulated yet
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if (i_processed == 0)
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break;
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}
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}
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#endif
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}
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thread_end:
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