Fix ring buffer handling in pa_mac_core.c that caused many spurious paInputOverflow flags.
Fixed size bug when input underflow detected. Code was confused between bytes and elements. Zero out the beginning of a recording when detecting pops to prevent start of audio from looking like a pop in qa loopback test.
This commit is contained in:
philburk
parent
44bdc45b2a
commit
b4a22b91ff
4 changed files with 55 additions and 42 deletions
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@ -515,6 +515,12 @@ void PaQa_FadeInRecording( PaQaRecording *recording, int startFrame, int count )
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assert( startFrame >= 0 );
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assert( count > 0 );
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/* Zero out initial part of the recording. */
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for( is=0; is<startFrame; is++ )
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{
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recording->buffer[ is ] = 0.0f;
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}
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/* Fade in where signal begins. */
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for( is=0; is<count; is++ )
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{
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double c = cos( phase );
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@ -989,8 +989,8 @@ static int PaQa_AnalyzeLoopbackConnection( UserOptions *userOptions, PaDeviceInd
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int framesPerBuffers[] = { 0, 16, 32, 40, 64, 100, 128, 256, 512, 1024 };
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int numBufferSizes = (sizeof(framesPerBuffers)/sizeof(int));
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PaSampleFormat sampleFormats[] = { paUInt8, paInt8, paInt16, paInt32 };
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const char *sampleFormatNames[] = { "paUInt8", "paInt8", "paInt16", "paInt32" };
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PaSampleFormat sampleFormats[] = { paFloat32, paUInt8, paInt8, paInt16, paInt32 };
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const char *sampleFormatNames[] = { "paFloat32", "paUInt8", "paInt8", "paInt16", "paInt32" };
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int numSampleFormats = (sizeof(sampleFormats)/sizeof(PaSampleFormat));
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printf( "=============== Analysing Loopback %d to %d =====================\n", outputDevice, inputDevice );
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@ -1774,25 +1774,16 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
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do is initialize everything so that if we fail, we know what hasn't
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been touched.
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*/
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bzero( stream, sizeof( PaMacCoreStream ) );
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stream->inputAudioBufferList.mBuffers[0].mData = NULL;
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stream->inputRingBuffer.buffer = NULL;
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bzero( &stream->blio, sizeof( PaMacBlio ) );
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/*
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/*
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stream->blio.inputRingBuffer.buffer = NULL;
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stream->blio.outputRingBuffer.buffer = NULL;
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stream->blio.inputSampleFormat = inputParameters?inputParameters->sampleFormat:0;
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stream->blio.inputSampleSize = computeSampleSizeFromFormat(stream->blio.inputSampleFormat);
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stream->blio.outputSampleFormat=outputParameters?outputParameters->sampleFormat:0;
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stream->blio.outputSampleSize = computeSampleSizeFromFormat(stream->blio.outputSampleFormat);
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*/
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stream->inputSRConverter = NULL;
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stream->inputUnit = NULL;
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stream->outputUnit = NULL;
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stream->inputFramesPerBuffer = 0;
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stream->outputFramesPerBuffer = 0;
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stream->bufferProcessorIsInitialized = FALSE;
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stream->timingInformationMutexIsInitialized = 0;
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*/
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/* assert( streamCallback ) ; */ /* only callback mode is implemented */
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if( streamCallback )
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@ -2251,6 +2242,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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*/
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OSStatus err = 0;
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unsigned long frames;
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long bytesPerFrame = sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
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/* -- start processing -- */
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PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
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@ -2261,8 +2253,8 @@ static OSStatus AudioIOProc( void *inRefCon,
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/* -- compute frames. do some checks -- */
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assert( ioData->mNumberBuffers == 1 );
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assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
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frames = ioData->mBuffers[0].mDataByteSize;
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frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
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frames = ioData->mBuffers[0].mDataByteSize / bytesPerFrame;
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/* -- copy and process input data -- */
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err= AudioUnitRender(stream->inputUnit,
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ioActionFlags,
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@ -2301,6 +2293,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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* is required on input, that is done here as well.
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*/
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unsigned long frames;
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long bytesPerFrame = sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
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/* Sometimes, when stopping a duplex stream we get erroneous
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xrun flags, so if this is our last run, clear the flags. */
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@ -2322,8 +2315,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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/* -- Copy and process output data -- */
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assert( ioData->mNumberBuffers == 1 );
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frames = ioData->mBuffers[0].mDataByteSize;
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frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
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frames = ioData->mBuffers[0].mDataByteSize / bytesPerFrame;
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assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
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PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames );
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PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor),
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@ -2337,6 +2329,8 @@ static OSStatus AudioIOProc( void *inRefCon,
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/* Here, we read the data out of the ring buffer, through the
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audio converter. */
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int inChan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels;
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long bytesPerFrame = flsz * inChan;
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if( stream->inputSRConverter )
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{
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OSStatus err;
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@ -2353,8 +2347,13 @@ static OSStatus AudioIOProc( void *inRefCon,
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{ /*the ring buffer callback underflowed */
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err = 0;
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bzero( ((char *)data) + size, sizeof(data)-size );
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/* The ring buffer can underflow normally when the stream is stopping.
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* So only report an error if the stream is active. */
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if( stream->state == ACTIVE )
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{
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stream->xrunFlags |= paInputUnderflow;
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}
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}
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ERR( err );
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assert( !err );
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@ -2374,7 +2373,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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AudioConverter would otherwise handle for us. */
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void *data1, *data2;
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ring_buffer_size_t size1, size2;
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PaUtil_GetRingBufferReadRegions( &stream->inputRingBuffer,
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ring_buffer_size_t framesReadable = PaUtil_GetRingBufferReadRegions( &stream->inputRingBuffer,
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frames,
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&data1, &size1,
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&data2, &size2 );
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@ -2389,14 +2388,21 @@ static OSStatus AudioIOProc( void *inRefCon,
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PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
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&callbackResult );
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PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1 );
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} else if( size1 + size2 < frames ) {
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} else if( framesReadable < frames ) {
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long sizeBytes1 = size1 * bytesPerFrame;
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long sizeBytes2 = size2 * bytesPerFrame;
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/*we underflowed. take what data we can, zero the rest.*/
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unsigned char data[frames*inChan*flsz];
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if( size1 )
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memcpy( data, data1, size1 );
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if( size2 )
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memcpy( data+size1, data2, size2 );
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bzero( data+size1+size2, frames*flsz*inChan - size1 - size2 );
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unsigned char data[ frames * bytesPerFrame ];
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if( size1 > 0 )
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{
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memcpy( data, data1, sizeBytes1 );
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}
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if( size2 > 0 )
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{
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memcpy( data+sizeBytes1, data2, sizeBytes2 );
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}
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bzero( data+sizeBytes1+sizeBytes2, (frames*bytesPerFrame) - sizeBytes1 - sizeBytes2 );
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PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
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PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
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@ -2407,7 +2413,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
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&callbackResult );
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PaUtil_AdvanceRingBufferReadIndex( &stream->inputRingBuffer,
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size1+size2 );
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framesReadable );
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/* flag underflow */
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stream->xrunFlags |= paInputUnderflow;
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} else {
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@ -2425,7 +2431,7 @@ static OSStatus AudioIOProc( void *inRefCon,
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framesProcessed =
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PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
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&callbackResult );
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PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1+size2 );
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PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, framesReadable );
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}
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}
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} else {
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@ -2463,14 +2469,14 @@ static OSStatus AudioIOProc( void *inRefCon,
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{
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/* If this is duplex or we use a converter, put the data
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into the ring buffer. */
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long bytesIn, bytesOut;
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bytesIn = sizeof( float ) * inNumberFrames * chan;
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bytesOut = PaUtil_WriteRingBuffer( &stream->inputRingBuffer,
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ring_buffer_size_t framesWritten = PaUtil_WriteRingBuffer( &stream->inputRingBuffer,
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stream->inputAudioBufferList.mBuffers[0].mData,
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inNumberFrames );
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if( bytesIn != bytesOut )
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if( framesWritten != inNumberFrames )
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{
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stream->xrunFlags |= paInputOverflow ;
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}
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}
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else
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{
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/* for simplex input w/o SR conversion,
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@ -359,8 +359,9 @@ int BlioCallback( const void *input, void *output, unsigned long frameCount,
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/* check for underflow */
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if( avail < frameCount * blio->inputSampleSizeActual * blio->inChan )
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{
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OSAtomicOr32( paInputOverflow, &blio->statusFlags );
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}
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toRead = MIN( avail, frameCount * blio->inputSampleSizeActual * blio->inChan );
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/* copy the data */
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