diff options
Diffstat (limited to 'pd/portaudio_v18/pa_mac_core/pa_mac_core.c')
| -rw-r--r-- | pd/portaudio_v18/pa_mac_core/pa_mac_core.c | 339 |
1 files changed, 213 insertions, 126 deletions
diff --git a/pd/portaudio_v18/pa_mac_core/pa_mac_core.c b/pd/portaudio_v18/pa_mac_core/pa_mac_core.c index 5bf24cb6..e0a31374 100644 --- a/pd/portaudio_v18/pa_mac_core/pa_mac_core.c +++ b/pd/portaudio_v18/pa_mac_core/pa_mac_core.c @@ -1,5 +1,5 @@ /* - * $Id: pa_mac_core.c,v 1.8.4.8 2003/03/07 01:34:18 philburk Exp $ + * $Id: pa_mac_core.c,v 1.8.4.12 2003/04/16 19:06:01 philburk Exp $ * pa_mac_core.c * Implementation of PortAudio for Mac OS X Core Audio * @@ -97,6 +97,17 @@ See code related to "streamInterleavingBuffer". 03.06.2003 - Phil Burk and Ryan Francesconi - fixed numChannels query for MOTU828. Handle fact that MOTU828 gives you 8 channels even when you ask for 2! + 04.06.2003 - Phil Burk - Combine Dominic Mazzoni's technique of using Configuration to query maxChannels + with old technique of scanning for mormat. + Increase channel scan by 1 to handle mono USB microphones. + Do not merge or split channels in AudioConverter to handle 2+2 channels + of Quattro which has a format of 2 channels. + 04.07.2003 - Phil Burk - use AudioGetCurrentHostTime instead of getrusage() which can lock threads. + 04.10.2003 - Phil Burk - fixed pointer bug with input deinterleaving loop. + Detect and ignore NULL inputData and outputData in CodeAudio callback. + Overlap creation and deletion of AudioConverters to prevent thread death when device rate changes. + 04.16.2003 - Phil Burk - Fixed input channel scrambling when numChannels != 2^N. Caused by alignment + error when filling RingBuffer with 2^N zero bytes. */ #include <CoreServices/CoreServices.h> @@ -107,12 +118,16 @@ #include <AudioUnit/AudioUnit.h> #include <AudioToolbox/DefaultAudioOutput.h> #include <AudioToolbox/AudioConverter.h> +#include <CoreAudio/HostTime.h> #include "portaudio.h" #include "pa_host.h" #include "pa_trace.h" #include "ringbuffer.h" +/************************************************* Configuration ********/ +#define PA_ENABLE_LOAD_MEASUREMENT (1) + /************************************************* Constants ********/ #define SET_DEVICE_BUFFER_SIZE (1) @@ -173,8 +188,8 @@ typedef struct PaHostSoundControl char *ringBufferData; Boolean formatListenerCalled; /* For measuring CPU utilization. */ - struct rusage entryRusage; - double inverseMicrosPerHostBuffer; /* 1/Microseconds of real-time audio per user buffer. */ + UInt64 entryTime; + double inverseHostTicksPerBuffer; /* 1/Ticks of real-time audio per user buffer. */ } PaHostSoundControl; /************************************************************** @@ -289,15 +304,7 @@ static void Pa_StartUsageCalculation( internalPortAudioStream *past ) PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; if( pahsc == NULL ) return; /* Query user CPU timer for usage analysis and to prevent overuse of CPU. */ - getrusage( RUSAGE_SELF, &pahsc->entryRusage ); -} - -static long SubtractTime_AminusB( struct timeval *timeA, struct timeval *timeB ) -{ - long secs = timeA->tv_sec - timeB->tv_sec; - long usecs = secs * 1000000; - usecs += (timeA->tv_usec - timeB->tv_usec); - return usecs; + pahsc->entryTime = AudioGetCurrentHostTime(); } /****************************************************************************** @@ -306,9 +313,9 @@ static long SubtractTime_AminusB( struct timeval *timeA, struct timeval *timeB ) */ static void Pa_EndUsageCalculation( internalPortAudioStream *past ) { - struct rusage currentRusage; - long usecsElapsed; - double newUsage; + UInt64 exitTime; + UInt64 ticksElapsed; + double newUsage; #define LOWPASS_COEFFICIENT_0 (0.95) #define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) @@ -316,16 +323,15 @@ static void Pa_EndUsageCalculation( internalPortAudioStream *past ) PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; if( pahsc == NULL ) return; - if( getrusage( RUSAGE_SELF, ¤tRusage ) == 0 ) - { - usecsElapsed = SubtractTime_AminusB( ¤tRusage.ru_utime, &pahsc->entryRusage.ru_utime ); - - /* Use inverse because it is faster than the divide. */ - newUsage = usecsElapsed * pahsc->inverseMicrosPerHostBuffer; + exitTime = AudioGetCurrentHostTime(); + + ticksElapsed = exitTime - pahsc->entryTime; - past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + + /* Use inverse because it is faster than the divide. */ + newUsage = ticksElapsed * pahsc->inverseHostTicksPerBuffer; + /* Low pass filter result. */ + past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + (LOWPASS_COEFFICIENT_1 * newUsage); - } } /****************************************** END CPU UTILIZATION *******/ @@ -523,10 +529,61 @@ static int PaOSX_ScanDevices( Boolean isInput ) } /************************************************************************* -** Determine the maximum number of channels a device will support. +** Determine the maximum number of channels based on the configuration. ** @return maxChannels or negative error. */ -static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) +static int PaOSX_GetMaxChannels_Config( AudioDeviceID devID, Boolean isInput ) +{ + OSStatus err; + UInt32 outSize; + Boolean outWritable; + AudioBufferList *list; + int numChannels; + int i; + + // Determine maximum number of channels supported. + // dmazzoni: new method + + outSize = 0; + err = AudioDeviceGetPropertyInfo(devID, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &outSize, &outWritable); + if ( err != noErr ) + { + PRINT_ERR("PaOSX_GetMaxChannels_Config: Could not get stream configuration info", err); + sSavedHostError = err; + return paHostError; + } + + list = (AudioBufferList *)PaHost_AllocateFastMemory( outSize ); + err = AudioDeviceGetProperty(devID, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &outSize, list); + if ( err != noErr ) + { + PRINT_ERR("PaOSX_GetMaxChannels_Config: Could not get stream configuration", err); + sSavedHostError = err; + return paHostError; + } + + numChannels = 0; + for( i=0; i<list->mNumberBuffers; i++ ) + { + int bufChannels = list->mBuffers[i].mNumberChannels; + DBUG(("PaOSX_GetMaxChannels_Config: buffer %d has %d channels.\n", i, bufChannels )); + numChannels += bufChannels; + } + + PaHost_FreeFastMemory( list, outSize ); + + return numChannels; +} + +/************************************************************************* +** Determine the maximum number of channels a device will support based on scanning the format. +** @return maxChannels or negative error. +*/ +static int PaOSX_GetMaxChannels_Format( AudioDeviceID devID, Boolean isInput ) { OSStatus err; UInt32 outSize; @@ -540,13 +597,14 @@ static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) // For example, some 8 channel devices return 2 when given 256 as input. gotMax = false; maxChannels = 0; + numChannels = 0; while( !gotMax ) { memset( &formatDesc, 0, sizeof(formatDesc)); - numChannels = maxChannels + 2; - DBUG(("PaOSX_GetMaxChannels: try numChannels = %d = %d + 2\n", - numChannels, maxChannels )); + numChannels = numChannels + 1; + DBUG(("PaOSX_GetMaxChannels: try numChannels = %d = %d + 1\n", + numChannels, numChannels )); formatDesc.mChannelsPerFrame = numChannels; outSize = sizeof(formatDesc); @@ -557,7 +615,10 @@ static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) err, formatDesc.mChannelsPerFrame )); if( err != noErr ) { - gotMax = true; + if (numChannels > (maxChannels + 4)) // Try several possibilities above current max + { + gotMax = true; + } } else { @@ -568,7 +629,10 @@ static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) } else if(formatDesc.mChannelsPerFrame < numChannels) { - gotMax = true; + if (numChannels > (maxChannels + 4)) // Try several possibilities above current max + { + gotMax = true; + } } else { @@ -579,6 +643,24 @@ static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) return maxChannels; } + + +/************************************************************************* +** Determine the maximum number of channels a device will support. +** It is not clear at this point which the better technique so +** we do both and use the biggest result. +** +** @return maxChannels or negative error. +*/ +static int PaOSX_GetMaxChannels( AudioDeviceID devID, Boolean isInput ) +{ + int maxChannelsFormat; + int maxChannelsConfig; + maxChannelsFormat = PaOSX_GetMaxChannels_Format( devID, isInput ); + maxChannelsConfig = PaOSX_GetMaxChannels_Config( devID, isInput ); + return (maxChannelsFormat > maxChannelsConfig) ? maxChannelsFormat : maxChannelsConfig; +} + /************************************************************************* ** Try to fill in the device info for this device. ** Return 1 if a good device that PA can use. @@ -733,12 +815,22 @@ static OSStatus PaOSX_LoadAndProcess( internalPortAudioStream *past, inputBuffer = pahsc->input.converterBuffer; } + /* Measure CPU load. */ +#if PA_ENABLE_LOAD_MEASUREMENT + Pa_StartUsageCalculation( past ); +#endif + /* Fill part of audio converter buffer by converting input to user format, * calling user callback, then converting output to native format. */ if( PaConvert_Process( past, inputBuffer, outputBuffer )) { past->past_StopSoon = 1; } + +#if PA_ENABLE_LOAD_MEASUREMENT + Pa_EndUsageCalculation( past ); +#endif + } return err; } @@ -777,11 +869,11 @@ static OSStatus PaOSX_WriteInputRingBuffer( internalPortAudioStream *past, char *inputNativeBufferfPtr = NULL; PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; - /* Do we need to interleave the buffers first? */ + /* Do we need to deinterleave the buffers first? */ if( past->past_NumInputChannels != inInputData->mBuffers[0].mNumberChannels ) { - - numFramesInInputBuffer = inInputData->mBuffers[0].mDataByteSize / (sizeof(float) * inInputData->mBuffers[0].mNumberChannels); + numFramesInInputBuffer = inInputData->mBuffers[0].mDataByteSize / + (sizeof(float) * inInputData->mBuffers[0].mNumberChannels); numBytes = numFramesInInputBuffer * sizeof(float) * past->past_NumInputChannels; @@ -813,11 +905,14 @@ static OSStatus PaOSX_WriteInputRingBuffer( internalPortAudioStream *past, for( j=0; j<numChannelsUsedInThisBuffer; j++ ) { int k; + float *dest = &pahsc->input.streamInterleavingBuffer[ currentInterleavedChannelIndex ]; + float *src = &((float *)inInputData->mBuffers[i].mData)[ j ]; /* Move one channel from CoreAudio buffer to interleaved buffer. */ for( k=0; k<numFramesInInputBuffer; k++ ) { - pahsc->input.streamInterleavingBuffer[ k*numInterleavedChannels + currentInterleavedChannelIndex ] = - ((float *)inInputData->mBuffers[i].mData)[ k*numBufChannels + j ]; + *dest = *src; + src += numBufChannels; + dest += numInterleavedChannels; } currentInterleavedChannelIndex++; } @@ -830,7 +925,7 @@ static OSStatus PaOSX_WriteInputRingBuffer( internalPortAudioStream *past, else { inputNativeBufferfPtr = (char*)inInputData->mBuffers[0].mData; - numBytes += inInputData->mBuffers[0].mDataByteSize; + numBytes = inInputData->mBuffers[0].mDataByteSize; } writeRoom = RingBuffer_GetWriteAvailable( &pahsc->ringBuffer ); @@ -838,7 +933,7 @@ static OSStatus PaOSX_WriteInputRingBuffer( internalPortAudioStream *past, if( numBytes <= writeRoom ) { RingBuffer_Write( &pahsc->ringBuffer, inputNativeBufferfPtr, numBytes ); - DBUGBACK(("PaOSX_WriteInputRingBuffer: wrote %ld bytes to FIFO.\n", inInputData->mBuffers[0].mDataByteSize)); + DBUGBACK(("PaOSX_WriteInputRingBuffer: wrote %ld bytes to FIFO.\n", numBytes)); } // FIXME else drop samples on floor, remember overflow??? return noErr; @@ -854,6 +949,12 @@ static OSStatus PaOSX_HandleInput( internalPortAudioStream *past, OSStatus err = noErr; PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( inInputData == NULL ) + { + DBUG(("PaOSX_HandleInput: inInputData == NULL\n")); + return noErr; + } + if( inInputData->mNumberBuffers > 0 ) { /* Write to FIFO here if we are only using this callback. */ @@ -894,9 +995,16 @@ static OSStatus PaOSX_HandleOutput( internalPortAudioStream *past, Boolean deinterleavingNeeded; int numFramesInOutputBuffer; + if( outOutputData == NULL ) + { + DBUG(("PaOSX_HandleOutput: outOutputData == NULL\n")); + return noErr; + } + deinterleavingNeeded = past->past_NumOutputChannels != outOutputData->mBuffers[0].mNumberChannels; - numFramesInOutputBuffer = outOutputData->mBuffers[0].mDataByteSize / (sizeof(float) * outOutputData->mBuffers[0].mNumberChannels); + numFramesInOutputBuffer = outOutputData->mBuffers[0].mDataByteSize / + (sizeof(float) * outOutputData->mBuffers[0].mNumberChannels); if( pahsc->mode != PA_MODE_INPUT_ONLY ) { @@ -931,7 +1039,7 @@ static OSStatus PaOSX_HandleOutput( internalPortAudioStream *past, outputNativeBufferfPtr = (void*)outOutputData->mBuffers[0].mData; } - /* Pull code from PA user through converter. */ + /* Pull data from PA user through converter. */ err = AudioConverterFillBuffer( pahsc->output.converter, PaOSX_OutputConverterCallbackProc, @@ -962,11 +1070,14 @@ static OSStatus PaOSX_HandleOutput( internalPortAudioStream *past, for( j=0; j<numChannelsUsedInThisBuffer; j++ ) { int k; + float *dest = &((float *)outOutputData->mBuffers[i].mData)[ j ]; + float *src = &pahsc->output.streamInterleavingBuffer[ currentInterleavedChannelIndex ]; /* Move one channel from interleaved buffer to CoreAudio buffer. */ for( k=0; k<numFramesInOutputBuffer; k++ ) { - ((float *)outOutputData->mBuffers[i].mData)[ k*numBufChannels + j ] = - pahsc->output.streamInterleavingBuffer[ k*numInterleavedChannels + currentInterleavedChannelIndex ]; + *dest = *src; + dest += numBufChannels; + src += numInterleavedChannels; } currentInterleavedChannelIndex++; } @@ -999,7 +1110,7 @@ static OSStatus PaOSX_CoreAudioInputCallback (AudioDeviceID inDevice, const Aud pahsc = (PaHostSoundControl *) past->past_DeviceData; /* If there is a FIFO for input then write to it. */ - if( pahsc->ringBufferData != NULL ) + if( (pahsc->ringBufferData != NULL) && (inInputData != NULL) ) { err = PaOSX_WriteInputRingBuffer( past, inInputData ); if( err != noErr ) goto error; @@ -1051,8 +1162,6 @@ static OSStatus PaOSX_CoreAudioIOCallback (AudioDeviceID inDevice, const AudioT past->past_FrameCount = inInputTime->mSampleTime; } - /* Measure CPU load. */ - Pa_StartUsageCalculation( past ); past->past_NumCallbacks += 1; /* Process full input buffer. */ @@ -1062,8 +1171,6 @@ static OSStatus PaOSX_CoreAudioIOCallback (AudioDeviceID inDevice, const AudioT /* Fill up empty output buffers. */ err = PaOSX_HandleOutput( past, outOutputData ); if( err != 0 ) goto error; - - Pa_EndUsageCalculation( past ); } if( err != 0 ) DBUG(("PaOSX_CoreAudioIOCallback: returns %ld.\n", err )); @@ -1198,7 +1305,7 @@ static void PaOSX_FixVolumeScalars( AudioDeviceID devID, Boolean isInput, kAudioDevicePropertyMute, &dataSize, &uidata32 ); if( err == noErr ) { - DBUG(("uidata32 for channel %d = %ld\n", iChannel, uidata32)); + DBUG(("mute for channel %d = %ld\n", iChannel, uidata32)); if( uidata32 == 1 ) // muted? { dataSize = sizeof( uidata32 ); @@ -1304,24 +1411,16 @@ static OSStatus PAOSX_DevicePropertyListener (AudioDeviceID inDevice, UInt32 dataSize; OSStatus err = noErr; AudioStreamBasicDescription userStreamFormat, hardwareStreamFormat; - Boolean updateInverseMicros; - Boolean updateConverter; + PaHostInOut *hostInOut; + AudioStreamBasicDescription *destFormatPtr, *srcFormatPtr; past = (internalPortAudioStream *) inClientData; pahsc = (PaHostSoundControl *) past->past_DeviceData; DBUG(("PAOSX_DevicePropertyListener: called with propertyID = 0x%0X\n", (unsigned int) inPropertyID )); - updateInverseMicros = (inDevice == pahsc->primaryDeviceID) && - ((inPropertyID == kAudioDevicePropertyStreamFormat) || - (inPropertyID == kAudioDevicePropertyBufferFrameSize)); - - updateConverter = (inPropertyID == kAudioDevicePropertyStreamFormat); - - // Sample rate needed for both. - if( updateConverter || updateInverseMicros ) + if(inPropertyID == kAudioDevicePropertyStreamFormat) { - /* Get target device format */ dataSize = sizeof(hardwareStreamFormat); err = AudioDeviceGetProperty(inDevice, 0, isInput, @@ -1332,12 +1431,7 @@ static OSStatus PAOSX_DevicePropertyListener (AudioDeviceID inDevice, sSavedHostError = err; goto error; } - } - - if( updateConverter ) - { - DBUG(("PAOSX_DevicePropertyListener: HW rate = %f\n", hardwareStreamFormat.mSampleRate )); - DBUG(("PAOSX_DevicePropertyListener: user rate = %f\n", past->past_SampleRate )); + DBUG(("PAOSX_DevicePropertyListener: HW mChannelsPerFrame = %d\n", (int)hardwareStreamFormat.mChannelsPerFrame )); /* Set source user format. */ @@ -1349,69 +1443,48 @@ static OSStatus PAOSX_DevicePropertyListener (AudioDeviceID inDevice, userStreamFormat.mBytesPerFrame = userStreamFormat.mChannelsPerFrame * sizeof(float); userStreamFormat.mBytesPerPacket = userStreamFormat.mBytesPerFrame * userStreamFormat.mFramesPerPacket; - /* Don't use AudioConverter for merging channels. */ - if( hardwareStreamFormat.mChannelsPerFrame > userStreamFormat.mChannelsPerFrame ) - { - hardwareStreamFormat.mChannelsPerFrame = userStreamFormat.mChannelsPerFrame; - hardwareStreamFormat.mBytesPerFrame = userStreamFormat.mBytesPerFrame; - hardwareStreamFormat.mBytesPerPacket = userStreamFormat.mBytesPerPacket; - } - + /* Don't use AudioConverter for merging or splitting channels. */ + hardwareStreamFormat.mChannelsPerFrame = userStreamFormat.mChannelsPerFrame; + hardwareStreamFormat.mBytesPerFrame = userStreamFormat.mBytesPerFrame; + hardwareStreamFormat.mBytesPerPacket = userStreamFormat.mBytesPerPacket; + if( isInput ) { - if( pahsc->input.converter != NULL ) - { - verify_noerr(AudioConverterDispose (pahsc->input.converter)); - } - - // Convert from hardware format to user format. - err = AudioConverterNew ( - &hardwareStreamFormat, - &userStreamFormat, - &pahsc->input.converter ); - if( err != noErr ) - { - PRINT_ERR("Could not create input format converter", err); - sSavedHostError = err; - goto error; - } + hostInOut = &pahsc->input; + srcFormatPtr = &hardwareStreamFormat; + destFormatPtr = &userStreamFormat; } else { - if( pahsc->output.converter != NULL ) - { - verify_noerr(AudioConverterDispose (pahsc->output.converter)); - } - - // Convert from user format to hardware format. - err = AudioConverterNew ( - &userStreamFormat, - &hardwareStreamFormat, - &pahsc->output.converter ); - if( err != noErr ) - { - PRINT_ERR("Could not create output format converter", err); - sSavedHostError = err; - goto error; - } + hostInOut = &pahsc->output; + srcFormatPtr = &userStreamFormat; + destFormatPtr = &hardwareStreamFormat; } - } - - if( updateInverseMicros ) - { - // Update coefficient used to calculate CPU Load based on sampleRate and bufferSize. - UInt32 ioBufferSize; - dataSize = sizeof(ioBufferSize); - err = AudioDeviceGetProperty( inDevice, 0, isInput, - kAudioDevicePropertyBufferFrameSize, &dataSize, - &ioBufferSize); - if( err == noErr ) + DBUG(("PAOSX_DevicePropertyListener: source rate = %f\n", srcFormatPtr->mSampleRate )); + DBUG(("PAOSX_DevicePropertyListener: dest rate = %f\n", destFormatPtr->mSampleRate )); + + // Don't delete old converter until we create new one so we don't pull + // the rug out from under other audio threads. + AudioConverterRef oldConverter = hostInOut->converter; + + // Make converter to change sample rate. + err = AudioConverterNew ( + srcFormatPtr, + destFormatPtr, + &hostInOut->converter ); + if( err != noErr ) { - pahsc->inverseMicrosPerHostBuffer = hardwareStreamFormat.mSampleRate / - (1000000.0 * ioBufferSize); + PRINT_ERR("Could not create format converter", err); + sSavedHostError = err; + goto error; + } + + if( oldConverter != NULL ) + { + verify_noerr( AudioConverterDispose( oldConverter ) ); } } - + error: pahsc->formatListenerCalled = true; return err; @@ -1430,6 +1503,7 @@ static PaError PaOSX_CreateInputRingBuffer( internalPortAudioStream *past ) UInt32 framesPerHostBuffer; UInt32 bytesForDevice; UInt32 bytesForUser; + UInt32 bytesPerFrame; AudioStreamBasicDescription formatDesc; dataSize = sizeof(formatDesc); @@ -1437,7 +1511,7 @@ static PaError PaOSX_CreateInputRingBuffer( internalPortAudioStream *past ) kAudioDevicePropertyStreamFormat, &dataSize, &formatDesc); if( err != noErr ) { - PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get I/O buffer size.\n", err); + PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get input format.\n", err); sSavedHostError = err; return paHostError; } @@ -1452,15 +1526,15 @@ static PaError PaOSX_CreateInputRingBuffer( internalPortAudioStream *past ) &framesPerHostBuffer); if( err != noErr ) { - PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get I/O buffer size.\n", err); + PRINT_ERR("PaOSX_CreateInputRingBuffer: Could not get input buffer size.\n", err); sSavedHostError = err; return paHostError; } - bytesForDevice = framesPerHostBuffer * formatDesc.mChannelsPerFrame * sizeof(Float32) * 2; + bytesPerFrame = past->past_NumInputChannels * sizeof(Float32); - bytesForUser = past->past_FramesPerUserBuffer * past->past_NumInputChannels * - sizeof(Float32) * 3 * sampleRateRatio; + bytesForDevice = framesPerHostBuffer * bytesPerFrame * 2; + bytesForUser = past->past_FramesPerUserBuffer * bytesPerFrame * 3 * sampleRateRatio; // Ring buffer should be large enough to consume audio input from device, // and to deliver a complete user buffer. @@ -1475,8 +1549,13 @@ static PaError PaOSX_CreateInputRingBuffer( internalPortAudioStream *past ) return paInsufficientMemory; } RingBuffer_Init( &pahsc->ringBuffer, numBytes, pahsc->ringBufferData ); - // make it look full at beginning - RingBuffer_AdvanceWriteIndex( &pahsc->ringBuffer, numBytes ); + // Make it look almost full at beginning. We must advance by an integral number of frames + // so that the channels don't get scrambled when numChannels is not a power of 2. + { + int numZeroFrames = numBytes / bytesPerFrame; + int numZeroBytes = numZeroFrames * bytesPerFrame; + RingBuffer_AdvanceWriteIndex( &pahsc->ringBuffer, numZeroBytes ); + } return paNoError; } @@ -1750,7 +1829,7 @@ PaError PaHost_OpenStream( internalPortAudioStream *past ) DBUG(("outputDeviceID = %ld\n", pahsc->output.audioDeviceID )); DBUG(("inputDeviceID = %ld\n", pahsc->input.audioDeviceID )); DBUG(("primaryDeviceID = %ld\n", pahsc->primaryDeviceID )); - + /* ------------------ OUTPUT */ if( useOutput ) { @@ -1764,6 +1843,14 @@ PaError PaHost_OpenStream( internalPortAudioStream *past ) result = PaOSX_OpenInputDevice( past ); if( result < 0 ) goto error; } + +#if PA_ENABLE_LOAD_MEASUREMENT + pahsc->inverseHostTicksPerBuffer = past->past_SampleRate / + (AudioGetHostClockFrequency() * past->past_FramesPerUserBuffer); + DBUG(("inverseHostTicksPerBuffer based on buffer size of %d frames.\n", past->past_FramesPerUserBuffer )); +#else + PRINT(("WARNING - Pa_GetCPULoad() mesaurement disabled in pa_mac_core.c.\n")); +#endif return result; |