diff options
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 959 |
1 files changed, 643 insertions, 316 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp index 8d15a34..cd427f3 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -4,13 +4,13 @@ RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows (DirectSound and ASIO) operating systems. RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2008 Gary P. Scavone + Copyright (c) 2001-2009 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -38,10 +38,13 @@ */ /************************************************************************/ -// RtAudio: Version 4.0.4 +// RtAudio: Version 4.0.5 #include "RtAudio.h" #include <iostream> +#include <cstdlib> +#include <cstring> +#include <limits.h> // Static variable definitions. const unsigned int RtApi::MAX_SAMPLE_RATES = 14; @@ -359,6 +362,13 @@ double RtApi :: getStreamTime( void ) #endif } +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + // *************************************************** // // @@ -393,7 +403,9 @@ double RtApi :: getStreamTime( void ) // implementation. struct CoreHandle { AudioDeviceID id[2]; // device ids - UInt32 iStream[2]; // device stream index (first for mono mode) + AudioDeviceIOProcID procId[2]; + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use bool xrun[2]; char *deviceBuffer; pthread_cond_t condition; @@ -401,7 +413,7 @@ struct CoreHandle { bool internalDrain; // Indicates if stop is initiated from callback or not. CoreHandle() - :deviceBuffer(0), drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } }; RtApiCore :: RtApiCore() @@ -805,70 +817,72 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne return FAILURE; } - // Search for a stream that contains the desired number of + // Search for one or more streams that contain the desired number of // channels. CoreAudio devices can have an arbitrary number of // streams and each stream can have an arbitrary number of channels. // For each stream, a single buffer of interleaved samples is - // provided. RtAudio currently only supports the use of one stream - // of interleaved data or multiple consecutive single-channel - // streams. Thus, our search below is limited to these two - // contexts. - unsigned int streamChannels = 0, nStreams = 0; - UInt32 iChannel = 0, iStream = 0; - unsigned int offsetCounter = firstChannel; - stream_.deviceInterleaved[mode] = true; - nStreams = bufferList->mNumberBuffers; + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; bool foundStream = false; + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStream<nStreams; iStream++ ) + deviceChannels += bufferList->mBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; for ( iStream=0; iStream<nStreams; iStream++ ) { streamChannels = bufferList->mBuffers[iStream].mNumberChannels; if ( streamChannels >= channels + offsetCounter ) { - iChannel += offsetCounter; + firstStream = iStream; + channelOffset = offsetCounter; foundStream = true; break; } if ( streamChannels > offsetCounter ) break; offsetCounter -= streamChannels; - iChannel += streamChannels; } - // If we didn't find a single stream above, see if we can meet - // the channel specification in mono mode (i.e. using separate - // non-interleaved buffers). This can only work if there are N - // consecutive one-channel streams, where N is the number of - // desired channels (+ channel offset). + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. if ( foundStream == false ) { - unsigned int counter = 0; + monoMode = true; offsetCounter = firstChannel; - iChannel = 0; for ( iStream=0; iStream<nStreams; iStream++ ) { streamChannels = bufferList->mBuffers[iStream].mNumberChannels; - if ( offsetCounter ) { - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - else if ( streamChannels == 1 ) - counter++; - else - counter = 0; - if ( counter == channels ) { - iStream -= channels - 1; - iChannel -= channels - 1; - stream_.deviceInterleaved[mode] = false; - foundStream = true; - break; - } - iChannel += streamChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; } - } - free( bufferList ); - if ( foundStream == false ) { - errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels."; - errorText_ = errorStream_.str(); - return FAILURE; + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } } + free( bufferList ); + // Determine the buffer size. AudioValueRange bufferRange; dataSize = sizeof( AudioValueRange ); @@ -885,8 +899,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; - // Set the buffer size. For mono mode, I'm assuming we only need to - // make this setting for the master channel. + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. UInt32 theSize = (UInt32) *bufferSize; dataSize = sizeof( UInt32 ); result = AudioDeviceSetProperty( id, NULL, 0, isInput, @@ -911,8 +925,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.bufferSize = *bufferSize; stream_.nBuffers = 1; - // Get the stream ID(s) so we can set the stream format. In mono - // mode, we'll have to do this for each stream (channel). + // Get the stream ID(s) so we can set the stream format. We'll have + // to do this for each stream. AudioStreamID streamIDs[ nStreams ]; dataSize = nStreams * sizeof( AudioStreamID ); result = AudioDeviceGetProperty( id, 0, isInput, @@ -928,13 +942,11 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // device and change that if necessary. AudioStreamBasicDescription description; dataSize = sizeof( AudioStreamBasicDescription ); - if ( stream_.deviceInterleaved[mode] ) nStreams = 1; - else nStreams = channels; bool updateFormat; - for ( unsigned int i=0; i<nStreams; i++ ) { + for ( UInt32 i=0; i<streamCount; i++ ) { - result = AudioStreamGetProperty( streamIDs[iStream+i], 0, + result = AudioStreamGetProperty( streamIDs[firstStream+i], 0, kAudioStreamPropertyVirtualFormat, &dataSize, &description ); @@ -959,7 +971,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } if ( updateFormat ) { - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyVirtualFormat, dataSize, &description ); if ( result != noErr ) { @@ -970,7 +982,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // Now check the physical format. - result = AudioStreamGetProperty( streamIDs[iStream+i], 0, + result = AudioStreamGetProperty( streamIDs[firstStream+i], 0, kAudioStreamPropertyPhysicalFormat, &dataSize, &description ); if ( result != noErr ) { @@ -988,32 +1000,32 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne testDescription.mBitsPerChannel = 32; formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 32; formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat; testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 24; testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 16; testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result == noErr ) continue; testDescription = description; testDescription.mBitsPerChannel = 8; testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; errorText_ = errorStream_.str(); @@ -1026,14 +1038,12 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // and the stream. First, attempt to get the device latency on the // master channel or the first open channel. Errors that might // occur here are not deemed critical. + + // ***** CHECK THIS ***** // UInt32 latency, channel = 0; dataSize = sizeof( UInt32 ); AudioDevicePropertyID property = kAudioDevicePropertyLatency; - for ( int i=0; i<2; i++ ) { - if ( hasProperty( id, channel, isInput, property ) == true ) break; - channel = iChannel + 1 + i; - } - if ( channel <= iChannel + 1 ) { + if ( hasProperty( id, channel, isInput, property ) == true ) { result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency ); if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency; else { @@ -1043,9 +1053,9 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } } - // Now try to get the stream latency. For "mono" mode, I assume the - // latency is equal for all single-channel streams. - result = AudioStreamGetProperty( streamIDs[iStream], 0, property, &dataSize, &latency ); + // Now try to get the stream latency. For multiple streams, I assume the + // latency is equal for each. + result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency ); if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency; else { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ")."; @@ -1063,14 +1073,16 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.userFormat = format; stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - if ( stream_.deviceInterleaved[mode] ) + if ( streamCount == 1 ) stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; - else // mono mode + else // multiple streams stream_.nDeviceChannels[mode] = channels; stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = iChannel; // offset within a CoreAudio stream + stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; @@ -1078,8 +1090,12 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) stream_.doConvertBuffer[mode] = true; // Allocate our CoreHandle structure for the stream. @@ -1101,11 +1117,13 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } else handle = (CoreHandle *) stream_.apiHandle; - handle->iStream[mode] = iStream; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; handle->id[mode] = id; // Allocate necessary internal buffers. - unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; @@ -1113,9 +1131,9 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // If possible, we will make use of the CoreAudio stream buffers as - // "device buffers". However, we can't do this if the device - // buffers are non-interleaved ("mono" mode). - if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) { + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); @@ -1134,13 +1152,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; goto error; } - - // Save a pointer to our own device buffer in the CoreHandle - // structure because we may need to use the stream_.deviceBuffer - // variable to point to the CoreAudio buffer before buffer - // conversion (if we have a duplex stream with two different - // conversion schemes). - handle->deviceBuffer = stream_.deviceBuffer; } } @@ -1149,30 +1160,22 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.state = STREAM_STOPPED; stream_.callbackInfo.object = (void *) this; - // Setup the buffer conversion information structure. We override - // the channel offset value and perform our own setting for that - // here. + // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) { - setConvertInfo( mode, 0 ); - - // Add channel offset for interleaved channels. - if ( firstChannel > 0 && stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].outOffset[k] += firstChannel; - } - else { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].inOffset[k] += firstChannel; - } - } + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); } if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) // Only one callback procedure per device. stream_.mode = DUPLEX; else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; errorText_ = errorStream_.str(); @@ -1225,13 +1228,23 @@ void RtApiCore :: closeStream( void ) if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[0], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[1], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif } for ( int i=0; i<2; i++ ) { @@ -1241,8 +1254,8 @@ void RtApiCore :: closeStream( void ) } } - if ( handle->deviceBuffer ) { - free( handle->deviceBuffer ); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } @@ -1311,6 +1324,11 @@ void RtApiCore :: stopStream( void ) MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1338,10 +1356,11 @@ void RtApiCore :: stopStream( void ) } } + stream_.state = STREAM_STOPPED; + unlock: MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result == noErr ) return; error( RtError::SYSTEM_ERROR ); } @@ -1386,6 +1405,12 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } + AudioDeviceID outputDevice = handle->id[0]; // Invoke user callback to get fresh output data UNLESS we are @@ -1418,48 +1443,96 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, if ( handle->drainCounter > 1 ) { // write zeros to the output stream - if ( stream_.deviceInterleaved[0] ) { + if ( handle->nStreams[0] == 1 ) { memset( outBufferList->mBuffers[handle->iStream[0]].mData, 0, outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } - else { - for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { + else { // fill multiple streams with zeros + for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, 0, outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); } } } - else if ( stream_.doConvertBuffer[0] ) { - - if ( stream_.deviceInterleaved[0] ) - stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData; - else - stream_.deviceBuffer = handle->deviceBuffer; - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - - if ( !stream_.deviceInterleaved[0] ) { - UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { - memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData, - &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); } - - } - else { - if ( stream_.deviceInterleaved[0] ) { + else { // copy from user buffer memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, stream_.userBuffer[0], outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } - else { + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { + for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData, - &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { + for ( unsigned int j=0; j<streamChannels; j++ ) { + *out++ = in[j*inOffset]; + } + out += outJump; + in += inJump; + } + channelsLeft -= streamChannels; } } } @@ -1470,29 +1543,93 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, } } - AudioDeviceID inputDevice = handle->id[1]; + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - if ( stream_.doConvertBuffer[1] ) { + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; - if ( stream_.deviceInterleaved[1] ) - stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData; - else { - stream_.deviceBuffer = (char *) handle->deviceBuffer; + if ( stream_.deviceInterleaved[1] == false ) { // mono mode UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; - for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) { - memcpy( &stream_.deviceBuffer[i*bufferBytes], + for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { + memcpy( (void *)&outBuffer[i*stream_.bufferSize], inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes ); } } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } - } - else { - memcpy( stream_.userBuffer[1], - inBufferList->mBuffers[handle->iStream[1]].mData, - inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { + for ( unsigned int j=0; j<streamChannels; j++ ) { + out[j*outOffset] = *in++; + } + out += outJump; + in += inJump; + } + channelsLeft -= streamChannels; + } + } + + if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer + convertBuffer( stream_.userBuffer[1], + stream_.deviceBuffer, + stream_.convertInfo[1] ); + } } } @@ -1505,7 +1642,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, const char* RtApiCore :: getErrorCode( OSStatus code ) { - switch( code ) { + switch( code ) { case kAudioHardwareNotRunningError: return "kAudioHardwareNotRunningError"; @@ -1542,10 +1679,10 @@ const char* RtApiCore :: getErrorCode( OSStatus code ) default: return "CoreAudio unknown error"; - } + } } -//******************** End of __MACOSX_CORE__ *********************// + //******************** End of __MACOSX_CORE__ *********************// #endif #if defined(__UNIX_JACK__) @@ -1597,9 +1734,15 @@ struct JackHandle { :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } }; +void jackSilentError( const char * ) {}; + RtApiJack :: RtApiJack() { // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif } RtApiJack :: ~RtApiJack() @@ -1610,7 +1753,9 @@ RtApiJack :: ~RtApiJack() unsigned int RtApiJack :: getDeviceCount( void ) { // See if we can become a jack client. - jack_client_t *client = jack_client_new( "RtApiJackCount" ); + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); if ( client == 0 ) return 0; const char **ports; @@ -1619,7 +1764,7 @@ unsigned int RtApiJack :: getDeviceCount( void ) ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nChannels ]; iColon = port.find(":"); @@ -1643,7 +1788,9 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) RtAudio::DeviceInfo info; info.probed = false; - jack_client_t *client = jack_client_new( "RtApiJackInfo" ); + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); if ( client == 0 ) { errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; error( RtError::WARNING ); @@ -1656,7 +1803,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); @@ -1771,10 +1918,12 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Look for jack server and try to become a client (only do once per stream). jack_client_t *client = 0; if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; if ( options && !options->streamName.empty() ) - client = jack_client_new( options->streamName.c_str() ); + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); else - client = jack_client_new( "RtApiJack" ); + client = jack_client_open( "RtApiJack", jackoptions, status ); if ( client == 0 ) { errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; error( RtError::WARNING ); @@ -1792,7 +1941,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); @@ -2131,6 +2280,11 @@ void RtApiJack :: stopStream( void ) MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + JackHandle *handle = (JackHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -2161,6 +2315,21 @@ void RtApiJack :: abortStream( void ) stopStream(); } +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the jack_deactivate() +// function will return. +extern "C" void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + + pthread_exit( NULL ); +} + bool RtApiJack :: callbackEvent( unsigned long nframes ) { if ( stream_.state == STREAM_STOPPED ) return SUCCESS; @@ -2180,15 +2349,23 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - pthread_cond_signal( &handle->condition ); + if ( handle->internalDrain == true ) { + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + } else - stopStream(); + pthread_cond_signal( &handle->condition ); return SUCCESS; } MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } + // Invoke user callback first, to get fresh output data. if ( handle->drainCounter == 0 ) { RtAudioCallback callback = (RtAudioCallback) info->callback; @@ -2206,7 +2383,8 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) stream_.bufferSize, streamTime, status, info->userData ); if ( handle->drainCounter == 2 ) { MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); return SUCCESS; } else if ( handle->drainCounter == 1 ) @@ -2270,7 +2448,7 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) RtApi::tickStreamTime(); return SUCCESS; } -//******************** End of __UNIX_JACK__ *********************// + //******************** End of __UNIX_JACK__ *********************// #endif #if defined(__WINDOWS_ASIO__) // ASIO API on Windows @@ -2625,11 +2803,28 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; else if ( granularity == -1 ) { // Make sure bufferSize is a power of two. - double power = std::log10( (double) *bufferSize ) / log10( 2.0 ); - *bufferSize = (int) pow( 2.0, floor(power+0.5) ); + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else *bufferSize = preferSize; } else if ( granularity != 0 ) { // Set to an even multiple of granularity, rounding up. @@ -2897,6 +3092,11 @@ void RtApiAsio :: stopStream() MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { @@ -2932,10 +3132,10 @@ void RtApiAsio :: abortStream() // The following lines were commented-out because some behavior was // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completed + // continuing sound, even when the device buffers are completely // disposed. So now, calling abort is the same as calling stop. - //AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - //handle->drainCounter = 1; + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 1; stopStream(); } @@ -3188,15 +3388,15 @@ static const char* getAsioErrorString( ASIOError result ) }; static Messages m[] = - { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) if ( m[i].value == result ) return m[i].message; @@ -3220,7 +3420,7 @@ static const char* getAsioErrorString( ASIOError result ) #include <assert.h> #if defined(__MINGW32__) -// missing from latest mingw winapi + // missing from latest mingw winapi #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ @@ -3235,7 +3435,7 @@ static const char* getAsioErrorString( ASIOError result ) static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) { - if (laterPointer > earlierPointer) + if ( laterPointer > earlierPointer ) return laterPointer - earlierPointer; else return laterPointer - earlierPointer + bufferSize; @@ -3684,7 +3884,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned bufferBytes *= 2; // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); if ( FAILED( result ) ) { @@ -3850,6 +4050,11 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned // Update wave format structure and buffer information. waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) + bufferBytes *= 2; // Setup the secondary DS buffer description. dsBufferSize = bufferBytes; @@ -3871,6 +4076,20 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned return FAILURE; } + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + bufferBytes = dscbcaps.dwBufferBytes; + // Lock the capture buffer LPVOID audioPtr; DWORD dataLen; @@ -4111,9 +4330,9 @@ void RtApiDs :: startStream() timeBeginPeriod( 1 ); /* - memset( &statistics, 0, sizeof( statistics ) ); - statistics.sampleRate = stream_.sampleRate; - statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; + memset( &statistics, 0, sizeof( statistics ) ); + statistics.sampleRate = stream_.sampleRate; + statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; */ buffersRolling = false; @@ -4170,6 +4389,11 @@ void RtApiDs :: stopStream() MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + HRESULT result = 0; LPVOID audioPtr; DWORD dataLen; @@ -4187,7 +4411,7 @@ void RtApiDs :: stopStream() LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; result = buffer->Stop(); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4196,7 +4420,7 @@ void RtApiDs :: stopStream() // we won't have old data playing. result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4207,7 +4431,7 @@ void RtApiDs :: stopStream() // Unlock the DS buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4223,7 +4447,7 @@ void RtApiDs :: stopStream() result = buffer->Stop(); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4232,7 +4456,7 @@ void RtApiDs :: stopStream() // we won't have old data playing. result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4243,7 +4467,7 @@ void RtApiDs :: stopStream() // Unlock the DS buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; errorText_ = errorStream_.str(); goto unlock; } @@ -4256,6 +4480,7 @@ void RtApiDs :: stopStream() timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); } @@ -4301,6 +4526,12 @@ void RtApiDs :: callbackEvent() MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { @@ -4345,7 +4576,7 @@ void RtApiDs :: callbackEvent() long bufferBytes; if ( stream_.mode == DUPLEX && !buffersRolling ) { - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); // It takes a while for the devices to get rolling. As a result, // there's no guarantee that the capture and write device pointers @@ -4396,7 +4627,7 @@ void RtApiDs :: callbackEvent() Sleep( 1 ); } - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); buffersRolling = true; handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); @@ -4656,17 +4887,17 @@ void RtApiDs :: callbackEvent() } #ifdef GENERATE_DEBUG_LOG if ( currentDebugLogEntry < debugLog.size() ) - { - TTickRecord &r = debugLog[currentDebugLogEntry++]; - r.currentReadPointer = currentReadPos; - r.safeReadPointer = safeReadPos; - r.currentWritePointer = currentWritePos; - r.safeWritePointer = safeWritePos; - r.readTime = readTime; - r.writeTime = writeTime; - r.nextReadPointer = handles[1].bufferPointer; - r.nextWritePointer = handles[0].bufferPointer; - } + { + TTickRecord &r = debugLog[currentDebugLogEntry++]; + r.currentReadPointer = currentReadPos; + r.safeReadPointer = safeReadPos; + r.currentWritePointer = currentWritePos; + r.safeWritePointer = safeWritePos; + r.readTime = readTime; + r.writeTime = writeTime; + r.nextReadPointer = handles[1].bufferPointer; + r.nextWritePointer = handles[0].bufferPointer; + } #endif unlock: @@ -4762,7 +4993,7 @@ static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, static char* getErrorString( int code ) { - switch ( code ) { + switch ( code ) { case DSERR_ALLOCATED: return "Already allocated"; @@ -4811,7 +5042,7 @@ static char* getErrorString( int code ) default: return "DirectSound unknown error"; - } + } } //******************** End of __WINDOWS_DS__ *********************// #endif @@ -4822,12 +5053,13 @@ static char* getErrorString( int code ) #include <alsa/asoundlib.h> #include <unistd.h> -// A structure to hold various information related to the ALSA API -// implementation. + // A structure to hold various information related to the ALSA API + // implementation. struct AlsaHandle { snd_pcm_t *handles[2]; bool synchronized; bool xrun[2]; + pthread_cond_t runnable; AlsaHandle() :synchronized(false) { xrun[0] = false; xrun[1] = false; } @@ -4863,17 +5095,17 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) errorText_ = errorStream_.str(); error( RtError::WARNING ); goto nextcard; - } - subdevice = -1; - while( 1 ) { + } + subdevice = -1; + while( 1 ) { result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { + if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); break; } - if ( subdevice < 0 ) + if ( subdevice < 0 ) break; nDevices++; } @@ -4906,17 +5138,17 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) errorText_ = errorStream_.str(); error( RtError::WARNING ); goto nextcard; - } - subdevice = -1; - while( 1 ) { + } + subdevice = -1; + while( 1 ) { result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { + if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); break; } - if ( subdevice < 0 ) break; + if ( subdevice < 0 ) break; if ( nDevices == device ) { sprintf( name, "hw:%d,%d", card, subdevice ); goto foundDevice; @@ -4954,8 +5186,8 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) int openMode = SND_PCM_ASYNC; snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); snd_pcm_t *phandle; snd_pcm_hw_params_t *params; snd_pcm_hw_params_alloca( ¶ms ); @@ -5184,12 +5416,12 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; - } - subdevice = -1; - while( 1 ) { + } + subdevice = -1; + while( 1 ) { result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; + if ( result < 0 ) break; + if ( subdevice < 0 ) break; if ( nDevices == device ) { sprintf( name, "hw:%d,%d", card, subdevice ); snd_ctl_close( chandle ); @@ -5414,19 +5646,10 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // Set the buffer number, which in ALSA is referred to as the "period". - int dir; + int totalSize, dir; unsigned int periods = 0; if ( options ) periods = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if ( periods < 2 ) periods = 2; - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + totalSize = *bufferSize * periods; // Set the buffer (or period) size. snd_pcm_uframes_t periodSize = *bufferSize; @@ -5439,6 +5662,18 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } *bufferSize = periodSize; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + else periods = totalSize / *bufferSize; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // If attempting to setup a duplex stream, the bufferSize parameter // MUST be the same in both directions! if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { @@ -5468,9 +5703,19 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne snd_pcm_sw_params_alloca( &sw_params ); snd_pcm_sw_params_current( phandle, sw_params ); snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, 0x7fffffff ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, INT_MAX ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + result = snd_pcm_sw_params( phandle, sw_params ); if ( result < 0 ) { snd_pcm_close( phandle ); @@ -5504,6 +5749,12 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; goto error; } + + if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) apiInfo; apiInfo->handles[0] = 0; apiInfo->handles[1] = 0; @@ -5572,13 +5823,28 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.callbackInfo.object = (void *) this; // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". pthread_attr_t attr; pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - pthread_attr_setschedpolicy( &attr, SCHED_RR ); + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif @@ -5597,6 +5863,7 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne error: if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; @@ -5626,10 +5893,14 @@ void RtApiAlsa :: closeStream() return; } + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &apiInfo->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; if ( stream_.state == STREAM_RUNNING ) { stream_.state = STREAM_STOPPED; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) @@ -5639,6 +5910,7 @@ void RtApiAlsa :: closeStream() } if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; @@ -5707,6 +5979,8 @@ void RtApiAlsa :: startStream() unlock: MUTEX_UNLOCK( &stream_.mutex ); + pthread_cond_signal( &apiInfo->runnable ); + if ( result >= 0 ) return; error( RtError::SYSTEM_ERROR ); } @@ -5720,11 +5994,13 @@ void RtApiAlsa :: stopStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + int result = 0; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; @@ -5750,6 +6026,7 @@ void RtApiAlsa :: stopStream() } unlock: + stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; @@ -5765,11 +6042,13 @@ void RtApiAlsa :: abortStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + int result = 0; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; @@ -5792,18 +6071,24 @@ void RtApiAlsa :: abortStream() } unlock: + stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result >= 0 ) return; error( RtError::SYSTEM_ERROR ); } void RtApiAlsa :: callbackEvent() { + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { - if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds - return; + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { @@ -5813,7 +6098,6 @@ void RtApiAlsa :: callbackEvent() } int doStopStream = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; @@ -5826,7 +6110,12 @@ void RtApiAlsa :: callbackEvent() apiInfo->xrun[1] = false; } doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } MUTEX_LOCK( &stream_.mutex ); @@ -5867,7 +6156,7 @@ void RtApiAlsa :: callbackEvent() } if ( result < (int) stream_.bufferSize ) { - // Either an error or underrun occured. + // Either an error or overrun occured. if ( result == -EPIPE ) { snd_pcm_state_t state = snd_pcm_state( handle[1] ); if ( state == SND_PCM_STATE_XRUN ) { @@ -5888,7 +6177,7 @@ void RtApiAlsa :: callbackEvent() errorText_ = errorStream_.str(); } error( RtError::WARNING ); - goto unlock; + goto tryOutput; } // Do byte swapping if necessary. @@ -5904,6 +6193,8 @@ void RtApiAlsa :: callbackEvent() if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } + tryOutput: + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Setup parameters and do buffer conversion if necessary. @@ -5969,7 +6260,6 @@ void RtApiAlsa :: callbackEvent() RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); - else if ( doStopStream == 2 ) this->abortStream(); } extern "C" void *alsaCallbackHandler( void *ptr ) @@ -5978,15 +6268,6 @@ extern "C" void *alsaCallbackHandler( void *ptr ) RtApiAlsa *object = (RtApiAlsa *) info->object; bool *isRunning = &info->isRunning; -#ifdef SCHED_RR - // Set a higher scheduler priority (P.J. Leonard) - struct sched_param param; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - param.sched_priority = min + ( max - min ) / 2; // Is this the best number? - sched_setscheduler( 0, SCHED_RR, ¶m ); -#endif - while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); @@ -6017,6 +6298,7 @@ struct OssHandle { int id[2]; // device ids bool xrun[2]; bool triggered; + pthread_cond_t runnable; OssHandle() :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } @@ -6162,7 +6444,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, + unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { @@ -6259,14 +6541,14 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned // For duplex operation, specifically set this mode (this doesn't seem to work). /* - if ( flags | O_RDWR ) { + if ( flags | O_RDWR ) { result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } } - } */ // Check the device channel support. @@ -6474,6 +6756,11 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned goto error; } + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; } else { @@ -6537,7 +6824,19 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - pthread_attr_setschedpolicy( &attr, SCHED_RR ); + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif @@ -6556,6 +6855,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned error: if ( handle ) { + pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; @@ -6585,10 +6885,14 @@ void RtApiOss :: closeStream() return; } + OssHandle *handle = (OssHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); - OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.state == STREAM_RUNNING ) { if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); @@ -6598,6 +6902,7 @@ void RtApiOss :: closeStream() } if ( handle ) { + pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; @@ -6637,6 +6942,9 @@ void RtApiOss :: startStream() // when fed samples. MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); } void RtApiOss :: stopStream() @@ -6648,11 +6956,14 @@ void RtApiOss :: stopStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + int result = 0; OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -6701,9 +7012,9 @@ void RtApiOss :: stopStream() } unlock: + stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result != -1 ) return; error( RtError::SYSTEM_ERROR ); } @@ -6717,11 +7028,14 @@ void RtApiOss :: abortStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + int result = 0; OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -6744,18 +7058,24 @@ void RtApiOss :: abortStream() } unlock: + stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result != -1 ) return; error( RtError::SYSTEM_ERROR ); } void RtApiOss :: callbackEvent() { + OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { - if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds - return; + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { @@ -6769,7 +7089,6 @@ void RtApiOss :: callbackEvent() RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; @@ -6780,6 +7099,10 @@ void RtApiOss :: callbackEvent() } doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } MUTEX_LOCK( &stream_.mutex ); @@ -6828,7 +7151,7 @@ void RtApiOss :: callbackEvent() handle->xrun[0] = true; errorText_ = "RtApiOss::callbackEvent: audio write error."; error( RtError::WARNING ); - goto unlock; + // Continue on to input section. } } @@ -6872,7 +7195,6 @@ void RtApiOss :: callbackEvent() RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); - else if ( doStopStream == 2 ) this->abortStream(); } extern "C" void *ossCallbackHandler( void *ptr ) @@ -6881,13 +7203,6 @@ extern "C" void *ossCallbackHandler( void *ptr ) RtApiOss *object = (RtApiOss *) info->object; bool *isRunning = &info->isRunning; -#ifdef SCHED_RR - // Set a higher scheduler priority (P.J. Leonard) - struct sched_param param; - param.sched_priority = 39; // Is this the best number? - sched_setscheduler( 0, SCHED_RR, ¶m ); -#endif - while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); @@ -7077,10 +7392,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info if (info.inFormat == RTAUDIO_SINT8) { signed char *in = (signed char *)inBuffer; - scale = 1.0 / 128.0; + scale = 1.0 / 127.5; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7089,10 +7405,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT16) { Int16 *in = (Int16 *)inBuffer; - scale = 1.0 / 32768.0; + scale = 1.0 / 32767.5; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7101,10 +7418,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT24) { Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 8388608.0; + scale = 1.0 / 8388607.5; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff); + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7113,10 +7431,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT32) { Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 2147483648.0; + scale = 1.0 / 2147483647.5; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7151,10 +7470,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info if (info.inFormat == RTAUDIO_SINT8) { signed char *in = (signed char *)inBuffer; - scale = 1.0 / 128.0; + scale = (Float32) ( 1.0 / 127.5 ); for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7163,10 +7483,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT16) { Int16 *in = (Int16 *)inBuffer; - scale = 1.0 / 32768.0; + scale = (Float32) ( 1.0 / 32767.5 ); for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7175,10 +7496,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT24) { Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 8388608.0; + scale = (Float32) ( 1.0 / 8388607.5 ); for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff); + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7187,10 +7509,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } else if (info.inFormat == RTAUDIO_SINT32) { Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 2147483648.0; + scale = (Float32) ( 1.0 / 2147483647.5 ); for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; + out[info.outOffset[j]] += 0.5; out[info.outOffset[j]] *= scale; } in += info.inJump; @@ -7269,7 +7592,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0); + out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7279,7 +7602,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float64 *in = (Float64 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0); + out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7336,7 +7659,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388608.0); + out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7346,7 +7669,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float64 *in = (Float64 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0); + out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7401,7 +7724,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.0); + out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7411,7 +7734,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float64 *in = (Float64 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.0); + out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7465,7 +7788,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.0); + out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7475,7 +7798,7 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info Float64 *in = (Float64 *)inBuffer; for (unsigned int i=0; i<stream_.bufferSize; i++) { for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.0); + out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); } in += info.inJump; out += info.outJump; @@ -7484,6 +7807,10 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } } + //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } + //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } + //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) { register char val; @@ -7516,8 +7843,8 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat *(ptr) = *(ptr+1); *(ptr+1) = val; - // Increment 4 bytes. - ptr += 4; + // Increment 3 more bytes. + ptr += 3; } } else if ( format == RTAUDIO_FLOAT64 ) { @@ -7545,18 +7872,18 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat *(ptr) = *(ptr+1); *(ptr+1) = val; - // Increment 8 bytes. - ptr += 8; + // Increment 5 more bytes. + ptr += 5; } } } -// Indentation settings for Vim and Emacs -// -// Local Variables: -// c-basic-offset: 2 -// indent-tabs-mode: nil -// End: -// -// vim: et sts=2 sw=2 + // Indentation settings for Vim and Emacs + // + // Local Variables: + // c-basic-offset: 2 + // indent-tabs-mode: nil + // End: + // + // vim: et sts=2 sw=2 |