diff options
Diffstat (limited to 'pd/portaudio/src/hostapi/coreaudio/pa_mac_core.c')
| -rw-r--r-- | pd/portaudio/src/hostapi/coreaudio/pa_mac_core.c | 2238 |
1 files changed, 2238 insertions, 0 deletions
diff --git a/pd/portaudio/src/hostapi/coreaudio/pa_mac_core.c b/pd/portaudio/src/hostapi/coreaudio/pa_mac_core.c new file mode 100644 index 00000000..7c887bd6 --- /dev/null +++ b/pd/portaudio/src/hostapi/coreaudio/pa_mac_core.c @@ -0,0 +1,2238 @@ +/* + * Implementation of the PortAudio API for Apple AUHAL + * + * PortAudio Portable Real-Time Audio Library + * Latest Version at: http://www.portaudio.com + * + * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code. + * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation) + * + * Dominic's code was based on code by Phil Burk, Darren Gibbs, + * Gord Peters, Stephane Letz, and Greg Pfiel. + * + * The following people also deserve acknowledgements: + * + * Olivier Tristan for feedback and testing + * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O + * interface. + * + * + * Based on the Open Source API proposed by Ross Bencina + * Copyright (c) 1999-2002 Ross Bencina, Phil Burk + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + */ + +/* + * The text above constitutes the entire PortAudio license; however, + * the PortAudio community also makes the following non-binding requests: + * + * Any person wishing to distribute modifications to the Software is + * requested to send the modifications to the original developer so that + * they can be incorporated into the canonical version. It is also + * requested that these non-binding requests be included along with the + * license above. + */ + +/** + @file pa_mac_core + @ingroup hostapi_src + @author Bjorn Roche + @brief AUHAL implementation of PortAudio +*/ + +/* FIXME: not all error conditions call PaUtil_SetLastHostErrorInfo() + * PaMacCore_SetError() will do this. + */ + +#include "pa_mac_core_internal.h" + +#include <string.h> /* strlen(), memcmp() etc. */ + +#include "pa_mac_core.h" +#include "pa_mac_core_utilities.h" +#include "pa_mac_core_blocking.h" + + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +/* prototypes for functions declared in this file */ + +PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index ); + +/* + * Function declared in pa_mac_core.h. Sets up a PaMacCoreStreamInfoStruct + * with the requested flags and initializes channel map. + */ +void PaMacCore_SetupStreamInfo( PaMacCoreStreamInfo *data, const unsigned long flags ) +{ + bzero( data, sizeof( PaMacCoreStreamInfo ) ); + data->size = sizeof( PaMacCoreStreamInfo ); + data->hostApiType = paCoreAudio; + data->version = 0x01; + data->flags = flags; + data->channelMap = NULL; + data->channelMapSize = 0; +} + +/* + * Function declared in pa_mac_core.h. Adds channel mapping to a PaMacCoreStreamInfoStruct + */ +void PaMacCore_SetupChannelMap( PaMacCoreStreamInfo *data, const SInt32 * const channelMap, const unsigned long channelMapSize ) +{ + data->channelMap = channelMap; + data->channelMapSize = channelMapSize; +} +static char *channelName = NULL; +static int channelNameSize = 0; +static bool ensureChannelNameSize( int size ) +{ + if( size >= channelNameSize ) { + free( channelName ); + channelName = (char *) malloc( ( channelNameSize = size ) + 1 ); + if( !channelName ) { + channelNameSize = 0; + return false; + } + } + return true; +} +/* + * Function declared in pa_mac_core.h. retrives channel names. + */ +const char *PaMacCore_GetChannelName( int device, int channelIndex, bool input ) +{ + struct PaUtilHostApiRepresentation *hostApi; + PaError err; + OSStatus error; + err = PaUtil_GetHostApiRepresentation( &hostApi, paCoreAudio ); + assert(err == paNoError); + PaMacAUHAL *macCoreHostApi = (PaMacAUHAL*)hostApi; + AudioDeviceID hostApiDevice = macCoreHostApi->devIds[device]; + + UInt32 size = 0; + + error = AudioDeviceGetPropertyInfo( hostApiDevice, + channelIndex + 1, + input, + kAudioDevicePropertyChannelName, + &size, + NULL ); + if( error ) { + //try the CFString + CFStringRef name; + bool isDeviceName = false; + size = sizeof( name ); + error = AudioDeviceGetProperty( hostApiDevice, + channelIndex + 1, + input, + kAudioDevicePropertyChannelNameCFString, + &size, + &name ); + if( error ) { //as a last-ditch effort, get the device name. Later we'll append the channel number. + size = sizeof( name ); + error = AudioDeviceGetProperty( hostApiDevice, + channelIndex + 1, + input, + kAudioDevicePropertyDeviceNameCFString, + &size, + &name ); + if( error ) + return NULL; + isDeviceName = true; + } + if( isDeviceName ) { + name = CFStringCreateWithFormat( NULL, NULL, CFSTR( "%@: %d"), name, channelIndex + 1 ); + } + + CFIndex length = CFStringGetLength(name); + while( ensureChannelNameSize( length * sizeof(UniChar) + 1 ) ) { + if( CFStringGetCString( name, channelName, channelNameSize, kCFStringEncodingUTF8 ) ) { + if( isDeviceName ) + CFRelease( name ); + return channelName; + } + if( length == 0 ) + ++length; + length *= 2; + } + if( isDeviceName ) + CFRelease( name ); + return NULL; + } + + //continue with C string: + if( !ensureChannelNameSize( size ) ) + return NULL; + + error = AudioDeviceGetProperty( hostApiDevice, + channelIndex + 1, + input, + kAudioDevicePropertyChannelName, + &size, + channelName ); + + if( error ) { + ERR( error ); + return NULL; + } + return channelName; +} + + + + + +AudioDeviceID PaMacCore_GetStreamInputDevice( PaStream* s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + VVDBUG(("PaMacCore_GetStreamInputHandle()\n")); + + return ( stream->inputDevice ); +} + +AudioDeviceID PaMacCore_GetStreamOutputDevice( PaStream* s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + VVDBUG(("PaMacCore_GetStreamOutputHandle()\n")); + + return ( stream->outputDevice ); +} + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#define RING_BUFFER_ADVANCE_DENOMINATOR (4) + +static void Terminate( struct PaUtilHostApiRepresentation *hostApi ); +static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi, + const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate ); +static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi, + PaStream** s, + const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate, + unsigned long framesPerBuffer, + PaStreamFlags streamFlags, + PaStreamCallback *streamCallback, + void *userData ); +static PaError CloseStream( PaStream* stream ); +static PaError StartStream( PaStream *stream ); +static PaError StopStream( PaStream *stream ); +static PaError AbortStream( PaStream *stream ); +static PaError IsStreamStopped( PaStream *s ); +static PaError IsStreamActive( PaStream *stream ); +static PaTime GetStreamTime( PaStream *stream ); +static void setStreamStartTime( PaStream *stream ); +static OSStatus AudioIOProc( void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData ); +static double GetStreamCpuLoad( PaStream* stream ); + +static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi, + PaDeviceInfo *deviceInfo, + AudioDeviceID macCoreDeviceId, + int isInput); + +static PaError OpenAndSetupOneAudioUnit( + const PaStreamParameters *inStreamParams, + const PaStreamParameters *outStreamParams, + const unsigned long requestedFramesPerBuffer, + unsigned long *actualInputFramesPerBuffer, + unsigned long *actualOutputFramesPerBuffer, + const PaMacAUHAL *auhalHostApi, + AudioUnit *audioUnit, + AudioConverterRef *srConverter, + AudioDeviceID *audioDevice, + const double sampleRate, + void *refCon ); + +/* for setting errors. */ +#define PA_AUHAL_SET_LAST_HOST_ERROR( errorCode, errorText ) \ + PaUtil_SetLastHostErrorInfo( paInDevelopment, errorCode, errorText ) + +/*currently, this is only used in initialization, but it might be modified + to be used when the list of devices changes.*/ +static PaError gatherDeviceInfo(PaMacAUHAL *auhalHostApi) +{ + UInt32 size; + UInt32 propsize; + VVDBUG(("gatherDeviceInfo()\n")); + /* -- free any previous allocations -- */ + if( auhalHostApi->devIds ) + PaUtil_GroupFreeMemory(auhalHostApi->allocations, auhalHostApi->devIds); + auhalHostApi->devIds = NULL; + + /* -- figure out how many devices there are -- */ + AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, + &propsize, + NULL ); + auhalHostApi->devCount = propsize / sizeof( AudioDeviceID ); + + VDBUG( ( "Found %ld device(s).\n", auhalHostApi->devCount ) ); + + /* -- copy the device IDs -- */ + auhalHostApi->devIds = (AudioDeviceID *)PaUtil_GroupAllocateMemory( + auhalHostApi->allocations, + propsize ); + if( !auhalHostApi->devIds ) + return paInsufficientMemory; + AudioHardwareGetProperty( kAudioHardwarePropertyDevices, + &propsize, + auhalHostApi->devIds ); +#ifdef MAC_CORE_VERBOSE_DEBUG + { + int i; + for( i=0; i<auhalHostApi->devCount; ++i ) + printf( "Device %d\t: %ld\n", i, auhalHostApi->devIds[i] ); + } +#endif + + size = sizeof(AudioDeviceID); + auhalHostApi->defaultIn = kAudioDeviceUnknown; + auhalHostApi->defaultOut = kAudioDeviceUnknown; + + /* determine the default device. */ + /* I am not sure how these calls to AudioHardwareGetProperty() + could fail, but in case they do, we use the first available + device as the default. */ + if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, + &size, + &auhalHostApi->defaultIn) ) { + int i; + auhalHostApi->defaultIn = kAudioDeviceUnknown; + VDBUG(("Failed to get default input device from OS.")); + VDBUG((" I will substitute the first available input Device.")); + for( i=0; i<auhalHostApi->devCount; ++i ) { + PaDeviceInfo devInfo; + if( 0 != GetChannelInfo( auhalHostApi, &devInfo, + auhalHostApi->devIds[i], TRUE ) ) + if( devInfo.maxInputChannels ) { + auhalHostApi->defaultIn = auhalHostApi->devIds[i]; + break; + } + } + } + if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, + &size, + &auhalHostApi->defaultOut) ) { + int i; + auhalHostApi->defaultIn = kAudioDeviceUnknown; + VDBUG(("Failed to get default output device from OS.")); + VDBUG((" I will substitute the first available output Device.")); + for( i=0; i<auhalHostApi->devCount; ++i ) { + PaDeviceInfo devInfo; + if( 0 != GetChannelInfo( auhalHostApi, &devInfo, + auhalHostApi->devIds[i], FALSE ) ) + if( devInfo.maxOutputChannels ) { + auhalHostApi->defaultOut = auhalHostApi->devIds[i]; + break; + } + } + } + + VDBUG( ( "Default in : %ld\n", auhalHostApi->defaultIn ) ); + VDBUG( ( "Default out: %ld\n", auhalHostApi->defaultOut ) ); + + return paNoError; +} + +static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi, + PaDeviceInfo *deviceInfo, + AudioDeviceID macCoreDeviceId, + int isInput) +{ + UInt32 propSize; + PaError err = paNoError; + UInt32 i; + int numChannels = 0; + AudioBufferList *buflist = NULL; + UInt32 frameLatency; + + VVDBUG(("GetChannelInfo()\n")); + + /* Get the number of channels from the stream configuration. + Fail if we can't get this. */ + + err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL)); + if (err) + return err; + + buflist = PaUtil_AllocateMemory(propSize); + if( !buflist ) + return paInsufficientMemory; + err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist)); + if (err) + goto error; + + for (i = 0; i < buflist->mNumberBuffers; ++i) + numChannels += buflist->mBuffers[i].mNumberChannels; + + if (isInput) + deviceInfo->maxInputChannels = numChannels; + else + deviceInfo->maxOutputChannels = numChannels; + + if (numChannels > 0) /* do not try to retrieve the latency if there is no channels. */ + { + /* Get the latency. Don't fail if we can't get this. */ + /* default to something reasonable */ + deviceInfo->defaultLowInputLatency = .01; + deviceInfo->defaultHighInputLatency = .10; + deviceInfo->defaultLowOutputLatency = .01; + deviceInfo->defaultHighOutputLatency = .10; + propSize = sizeof(UInt32); + err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency)); + if (!err) + { + /** FEEDBACK: + * This code was arrived at by trial and error, and some extentive, but not exhaustive + * testing. Sebastien Beaulieu <seb@plogue.com> has suggested using + * kAudioDevicePropertyLatency + kAudioDevicePropertySafetyOffset + buffer size instead. + * At the time this code was written, many users were reporting dropouts with audio + * programs that probably used this formula. This was probably + * around 10.4.4, and the problem is probably fixed now. So perhaps + * his formula should be reviewed and used. + * */ + double secondLatency = frameLatency / deviceInfo->defaultSampleRate; + if (isInput) + { + deviceInfo->defaultLowInputLatency = 3 * secondLatency; + deviceInfo->defaultHighInputLatency = 3 * 10 * secondLatency; + } + else + { + deviceInfo->defaultLowOutputLatency = 3 * secondLatency; + deviceInfo->defaultHighOutputLatency = 3 * 10 * secondLatency; + } + } + } + PaUtil_FreeMemory( buflist ); + return paNoError; + error: + PaUtil_FreeMemory( buflist ); + return err; +} + +static PaError InitializeDeviceInfo( PaMacAUHAL *auhalHostApi, + PaDeviceInfo *deviceInfo, + AudioDeviceID macCoreDeviceId, + PaHostApiIndex hostApiIndex ) +{ + Float64 sampleRate; + char *name; + PaError err = paNoError; + UInt32 propSize; + + VVDBUG(("InitializeDeviceInfo(): macCoreDeviceId=%ld\n", macCoreDeviceId)); + + memset(deviceInfo, 0, sizeof(deviceInfo)); + + deviceInfo->structVersion = 2; + deviceInfo->hostApi = hostApiIndex; + + /* Get the device name. Fail if we can't get it. */ + err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL)); + if (err) + return err; + + name = PaUtil_GroupAllocateMemory(auhalHostApi->allocations,propSize); + if ( !name ) + return paInsufficientMemory; + err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name)); + if (err) + return err; + deviceInfo->name = name; + + /* Try to get the default sample rate. Don't fail if we can't get this. */ + propSize = sizeof(Float64); + err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate)); + if (err) + deviceInfo->defaultSampleRate = 0.0; + else + deviceInfo->defaultSampleRate = sampleRate; + + /* Get the maximum number of input and output channels. Fail if we can't get this. */ + + err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 1); + if (err) + return err; + + err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 0); + if (err) + return err; + + return paNoError; +} + +PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex ) +{ + PaError result = paNoError; + int i; + PaMacAUHAL *auhalHostApi; + PaDeviceInfo *deviceInfoArray; + + VVDBUG(("PaMacCore_Initialize(): hostApiIndex=%d\n", hostApiIndex)); + + auhalHostApi = (PaMacAUHAL*)PaUtil_AllocateMemory( sizeof(PaMacAUHAL) ); + if( !auhalHostApi ) + { + result = paInsufficientMemory; + goto error; + } + + auhalHostApi->allocations = PaUtil_CreateAllocationGroup(); + if( !auhalHostApi->allocations ) + { + result = paInsufficientMemory; + goto error; + } + + auhalHostApi->devIds = NULL; + auhalHostApi->devCount = 0; + + /* get the info we need about the devices */ + result = gatherDeviceInfo( auhalHostApi ); + if( result != paNoError ) + goto error; + + *hostApi = &auhalHostApi->inheritedHostApiRep; + (*hostApi)->info.structVersion = 1; + (*hostApi)->info.type = paCoreAudio; + (*hostApi)->info.name = "Core Audio"; + + (*hostApi)->info.defaultInputDevice = paNoDevice; + (*hostApi)->info.defaultOutputDevice = paNoDevice; + + (*hostApi)->info.deviceCount = 0; + + if( auhalHostApi->devCount > 0 ) + { + (*hostApi)->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory( + auhalHostApi->allocations, sizeof(PaDeviceInfo*) * auhalHostApi->devCount); + if( !(*hostApi)->deviceInfos ) + { + result = paInsufficientMemory; + goto error; + } + + /* allocate all device info structs in a contiguous block */ + deviceInfoArray = (PaDeviceInfo*)PaUtil_GroupAllocateMemory( + auhalHostApi->allocations, sizeof(PaDeviceInfo) * auhalHostApi->devCount ); + if( !deviceInfoArray ) + { + result = paInsufficientMemory; + goto error; + } + + for( i=0; i < auhalHostApi->devCount; ++i ) + { + int err; + err = InitializeDeviceInfo( auhalHostApi, &deviceInfoArray[i], + auhalHostApi->devIds[i], + hostApiIndex ); + if (err == paNoError) + { /* copy some info and set the defaults */ + (*hostApi)->deviceInfos[(*hostApi)->info.deviceCount] = &deviceInfoArray[i]; + if (auhalHostApi->devIds[i] == auhalHostApi->defaultIn) + (*hostApi)->info.defaultInputDevice = (*hostApi)->info.deviceCount; + if (auhalHostApi->devIds[i] == auhalHostApi->defaultOut) + (*hostApi)->info.defaultOutputDevice = (*hostApi)->info.deviceCount; + (*hostApi)->info.deviceCount++; + } + else + { /* there was an error. we need to shift the devices down, so we ignore this one */ + int j; + auhalHostApi->devCount--; + for( j=i; j<auhalHostApi->devCount; ++j ) + auhalHostApi->devIds[j] = auhalHostApi->devIds[j+1]; + i--; + } + } + } + + (*hostApi)->Terminate = Terminate; + (*hostApi)->OpenStream = OpenStream; + (*hostApi)->IsFormatSupported = IsFormatSupported; + + PaUtil_InitializeStreamInterface( &auhalHostApi->callbackStreamInterface, + CloseStream, StartStream, + StopStream, AbortStream, IsStreamStopped, + IsStreamActive, + GetStreamTime, GetStreamCpuLoad, + PaUtil_DummyRead, PaUtil_DummyWrite, + PaUtil_DummyGetReadAvailable, + PaUtil_DummyGetWriteAvailable ); + + PaUtil_InitializeStreamInterface( &auhalHostApi->blockingStreamInterface, + CloseStream, StartStream, + StopStream, AbortStream, IsStreamStopped, + IsStreamActive, + GetStreamTime, PaUtil_DummyGetCpuLoad, + ReadStream, WriteStream, + GetStreamReadAvailable, + GetStreamWriteAvailable ); + + return result; + +error: + if( auhalHostApi ) + { + if( auhalHostApi->allocations ) + { + PaUtil_FreeAllAllocations( auhalHostApi->allocations ); + PaUtil_DestroyAllocationGroup( auhalHostApi->allocations ); + } + + PaUtil_FreeMemory( auhalHostApi ); + } + return result; +} + + +static void Terminate( struct PaUtilHostApiRepresentation *hostApi ) +{ + PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi; + + VVDBUG(("Terminate()\n")); + + /* + IMPLEMENT ME: + - clean up any resources not handled by the allocation group + TODO: Double check that everything is handled by alloc group + */ + + if( auhalHostApi->allocations ) + { + PaUtil_FreeAllAllocations( auhalHostApi->allocations ); + PaUtil_DestroyAllocationGroup( auhalHostApi->allocations ); + } + + PaUtil_FreeMemory( auhalHostApi ); +} + + +static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi, + const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate ) +{ + int inputChannelCount, outputChannelCount; + PaSampleFormat inputSampleFormat, outputSampleFormat; + + VVDBUG(("IsFormatSupported(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld sampleRate=%g\n", + inputParameters ? inputParameters->channelCount : -1, + inputParameters ? inputParameters->sampleFormat : -1, + outputParameters ? outputParameters->channelCount : -1, + outputParameters ? outputParameters->sampleFormat : -1, + (float) sampleRate )); + + /** These first checks are standard PA checks. We do some fancier checks + later. */ + if( inputParameters ) + { + inputChannelCount = inputParameters->channelCount; + inputSampleFormat = inputParameters->sampleFormat; + + /* all standard sample formats are supported by the buffer adapter, + this implementation doesn't support any custom sample formats */ + if( inputSampleFormat & paCustomFormat ) + return paSampleFormatNotSupported; + + /* unless alternate device specification is supported, reject the use of + paUseHostApiSpecificDeviceSpecification */ + + if( inputParameters->device == paUseHostApiSpecificDeviceSpecification ) + return paInvalidDevice; + + /* check that input device can support inputChannelCount */ + if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels ) + return paInvalidChannelCount; + } + else + { + inputChannelCount = 0; + } + + if( outputParameters ) + { + outputChannelCount = outputParameters->channelCount; + outputSampleFormat = outputParameters->sampleFormat; + + /* all standard sample formats are supported by the buffer adapter, + this implementation doesn't support any custom sample formats */ + if( outputSampleFormat & paCustomFormat ) + return paSampleFormatNotSupported; + + /* unless alternate device specification is supported, reject the use of + paUseHostApiSpecificDeviceSpecification */ + + if( outputParameters->device == paUseHostApiSpecificDeviceSpecification ) + return paInvalidDevice; + + /* check that output device can support outputChannelCount */ + if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels ) + return paInvalidChannelCount; + + } + else + { + outputChannelCount = 0; + } + + /* FEEDBACK */ + /* I think the only way to check a given format SR combo is */ + /* to try opening it. This could be disruptive, is that Okay? */ + /* The alternative is to just read off available sample rates, */ + /* but this will not work %100 of the time (eg, a device that */ + /* supports N output at one rate but only N/2 at a higher rate.)*/ + + /* The following code opens the device with the requested parameters to + see if it works. */ + { + PaError err; + PaStream *s; + err = OpenStream( hostApi, &s, inputParameters, outputParameters, + sampleRate, 1024, 0, (PaStreamCallback *)1, NULL ); + if( err != paNoError && err != paInvalidSampleRate ) + DBUG( ( "OpenStream @ %g returned: %d: %s\n", + (float) sampleRate, err, Pa_GetErrorText( err ) ) ); + if( err ) + return err; + err = CloseStream( s ); + if( err ) { + /* FEEDBACK: is this more serious? should we assert? */ + DBUG( ( "WARNING: could not close Stream. %d: %s\n", + err, Pa_GetErrorText( err ) ) ); + } + } + + return paFormatIsSupported; +} + +static PaError OpenAndSetupOneAudioUnit( + const PaStreamParameters *inStreamParams, + const PaStreamParameters *outStreamParams, + const unsigned long requestedFramesPerBuffer, + unsigned long *actualInputFramesPerBuffer, + unsigned long *actualOutputFramesPerBuffer, + const PaMacAUHAL *auhalHostApi, + AudioUnit *audioUnit, + AudioConverterRef *srConverter, + AudioDeviceID *audioDevice, + const double sampleRate, + void *refCon ) +{ + ComponentDescription desc; + Component comp; + /*An Apple TN suggests using CAStreamBasicDescription, but that is C++*/ + AudioStreamBasicDescription desiredFormat; + OSErr result = noErr; + PaError paResult = paNoError; + int line = 0; + UInt32 callbackKey; + AURenderCallbackStruct rcbs; + unsigned long macInputStreamFlags = paMacCorePlayNice; + unsigned long macOutputStreamFlags = paMacCorePlayNice; + SInt32 const *inChannelMap = NULL; + SInt32 const *outChannelMap = NULL; + unsigned long inChannelMapSize = 0; + unsigned long outChannelMapSize = 0; + + VVDBUG(("OpenAndSetupOneAudioUnit(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld, requestedFramesPerBuffer=%ld\n", + inStreamParams ? inStreamParams->channelCount : -1, + inStreamParams ? inStreamParams->sampleFormat : -1, + outStreamParams ? outStreamParams->channelCount : -1, + outStreamParams ? outStreamParams->sampleFormat : -1, + requestedFramesPerBuffer )); + + /* -- handle the degenerate case -- */ + if( !inStreamParams && !outStreamParams ) { + *audioUnit = NULL; + *audioDevice = kAudioDeviceUnknown; + return paNoError; + } + + /* -- get the user's api specific info, if they set any -- */ + if( inStreamParams && inStreamParams->hostApiSpecificStreamInfo ) + { + macInputStreamFlags= + ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo) + ->flags; + inChannelMap = ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo) + ->channelMap; + inChannelMapSize = ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo) + ->channelMapSize; + } + if( outStreamParams && outStreamParams->hostApiSpecificStreamInfo ) + { + macOutputStreamFlags= + ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo) + ->flags; + outChannelMap = ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo) + ->channelMap; + outChannelMapSize = ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo) + ->channelMapSize; + } + /* Override user's flags here, if desired for testing. */ + + /* + * The HAL AU is a Mac OS style "component". + * the first few steps deal with that. + * Later steps work on a combination of Mac OS + * components and the slightly lower level + * HAL. + */ + + /* -- describe the output type AudioUnit -- */ + /* Note: for the default AudioUnit, we could use the + * componentSubType value kAudioUnitSubType_DefaultOutput; + * but I don't think that's relevant here. + */ + desc.componentType = kAudioUnitType_Output; + desc.componentSubType = kAudioUnitSubType_HALOutput; + desc.componentManufacturer = kAudioUnitManufacturer_Apple; + desc.componentFlags = 0; + desc.componentFlagsMask = 0; + /* -- find the component -- */ + comp = FindNextComponent( NULL, &desc ); + if( !comp ) + { + DBUG( ( "AUHAL component not found." ) ); + *audioUnit = NULL; + *audioDevice = kAudioDeviceUnknown; + return paUnanticipatedHostError; + } + /* -- open it -- */ + result = OpenAComponent( comp, audioUnit ); + if( result ) + { + DBUG( ( "Failed to open AUHAL component." ) ); + *audioUnit = NULL; + *audioDevice = kAudioDeviceUnknown; + return ERR( result ); + } + /* -- prepare a little error handling logic / hackery -- */ +#define ERR_WRAP(mac_err) do { result = mac_err ; line = __LINE__ ; if ( result != noErr ) goto error ; } while(0) + + /* -- if there is input, we have to explicitly enable input -- */ + if( inStreamParams ) + { + UInt32 enableIO = 1; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Input, + INPUT_ELEMENT, + &enableIO, + sizeof(enableIO) ) ); + } + /* -- if there is no output, we must explicitly disable output -- */ + if( !outStreamParams ) + { + UInt32 enableIO = 0; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, + OUTPUT_ELEMENT, + &enableIO, + sizeof(enableIO) ) ); + } + + /* -- set the devices -- */ + /* make sure input and output are the same device if we are doing input and + output. */ + if( inStreamParams && outStreamParams ) + { + assert( outStreamParams->device == inStreamParams->device ); + } + if( inStreamParams ) + { + *audioDevice = auhalHostApi->devIds[inStreamParams->device] ; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, + INPUT_ELEMENT, + audioDevice, + sizeof(AudioDeviceID) ) ); + } + if( outStreamParams ) + { + *audioDevice = auhalHostApi->devIds[outStreamParams->device] ; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, + OUTPUT_ELEMENT, + audioDevice, + sizeof(AudioDeviceID) ) ); + } + + /* -- set format -- */ + bzero( &desiredFormat, sizeof(desiredFormat) ); + desiredFormat.mFormatID = kAudioFormatLinearPCM ; + desiredFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked; + desiredFormat.mFramesPerPacket = 1; + desiredFormat.mBitsPerChannel = sizeof( float ) * 8; + + result = 0; + /* set device format first, but only touch the device if the user asked */ + if( inStreamParams ) { + /*The callback never calls back if we don't set the FPB */ + /*This seems wierd, because I would think setting anything on the device + would be disruptive.*/ + paResult = setBestFramesPerBuffer( *audioDevice, FALSE, + requestedFramesPerBuffer, + actualInputFramesPerBuffer ); + if( paResult ) goto error; + if( macInputStreamFlags & paMacCoreChangeDeviceParameters ) { + bool requireExact; + requireExact=macInputStreamFlags & paMacCoreFailIfConversionRequired; + paResult = setBestSampleRateForDevice( *audioDevice, FALSE, + requireExact, sampleRate ); + if( paResult ) goto error; + } + if( actualInputFramesPerBuffer && actualOutputFramesPerBuffer ) + *actualOutputFramesPerBuffer = *actualInputFramesPerBuffer ; + } + if( outStreamParams && !inStreamParams ) { + /*The callback never calls back if we don't set the FPB */ + /*This seems wierd, because I would think setting anything on the device + would be disruptive.*/ + paResult = setBestFramesPerBuffer( *audioDevice, TRUE, + requestedFramesPerBuffer, + actualOutputFramesPerBuffer ); + if( paResult ) goto error; + if( macOutputStreamFlags & paMacCoreChangeDeviceParameters ) { + bool requireExact; + requireExact=macOutputStreamFlags & paMacCoreFailIfConversionRequired; + paResult = setBestSampleRateForDevice( *audioDevice, TRUE, + requireExact, sampleRate ); + if( paResult ) goto error; + } + } + + /* -- set the quality of the output converter -- */ + if( outStreamParams ) { + UInt32 value = kAudioConverterQuality_Max; + switch( macOutputStreamFlags & 0x0700 ) { + case 0x0100: /*paMacCore_ConversionQualityMin:*/ + value=kRenderQuality_Min; + break; + case 0x0200: /*paMacCore_ConversionQualityLow:*/ + value=kRenderQuality_Low; + break; + case 0x0300: /*paMacCore_ConversionQualityMedium:*/ + value=kRenderQuality_Medium; + break; + case 0x0400: /*paMacCore_ConversionQualityHigh:*/ + value=kRenderQuality_High; + break; + } + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioUnitProperty_RenderQuality, + kAudioUnitScope_Global, + OUTPUT_ELEMENT, + &value, + sizeof(value) ) ); + } + /* now set the format on the Audio Units. */ + if( outStreamParams ) + { + desiredFormat.mSampleRate =sampleRate; + desiredFormat.mBytesPerPacket=sizeof(float)*outStreamParams->channelCount; + desiredFormat.mBytesPerFrame =sizeof(float)*outStreamParams->channelCount; + desiredFormat.mChannelsPerFrame = outStreamParams->channelCount; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + OUTPUT_ELEMENT, + &desiredFormat, + sizeof(AudioStreamBasicDescription) ) ); + } + if( inStreamParams ) + { + AudioStreamBasicDescription sourceFormat; + UInt32 size = sizeof( AudioStreamBasicDescription ); + + /* keep the sample rate of the device, or we confuse AUHAL */ + ERR_WRAP( AudioUnitGetProperty( *audioUnit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, + INPUT_ELEMENT, + &sourceFormat, + &size ) ); + desiredFormat.mSampleRate = sourceFormat.mSampleRate; + desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount; + desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount; + desiredFormat.mChannelsPerFrame = inStreamParams->channelCount; + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + INPUT_ELEMENT, + &desiredFormat, + sizeof(AudioStreamBasicDescription) ) ); + } + /* set the maximumFramesPerSlice */ + /* not doing this causes real problems + (eg. the callback might not be called). The idea of setting both this + and the frames per buffer on the device is that we'll be most likely + to actually get the frame size we requested in the callback with the + minimum latency. */ + if( outStreamParams ) { + UInt32 size = sizeof( *actualOutputFramesPerBuffer ); + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Input, + OUTPUT_ELEMENT, + actualOutputFramesPerBuffer, + sizeof(unsigned long) ) ); + ERR_WRAP( AudioUnitGetProperty( *audioUnit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Global, + OUTPUT_ELEMENT, + actualOutputFramesPerBuffer, + &size ) ); + } + if( inStreamParams ) { + /*UInt32 size = sizeof( *actualInputFramesPerBuffer );*/ + ERR_WRAP( AudioUnitSetProperty( *audioUnit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Output, + INPUT_ELEMENT, + actualInputFramesPerBuffer, + sizeof(unsigned long) ) ); +/* Don't know why this causes problems + ERR_WRAP( AudioUnitGetProperty( *audioUnit, + kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Global, //Output, + INPUT_ELEMENT, + actualInputFramesPerBuffer, + &size ) ); +*/ + } + + /* -- if we have input, we may need to setup an SR converter -- */ + /* even if we got the sample rate we asked for, we need to do + the conversion in case another program changes the underlying SR. */ + /* FIXME: I think we need to monitor stream and change the converter if the incoming format changes. */ + if( inStreamParams ) { + AudioStreamBasicDescription desiredFormat; + AudioStreamBasicDescription sourceFormat; + UInt32 sourceSize = sizeof( sourceFormat ); + bzero( &desiredFormat, sizeof(desiredFormat) ); + desiredFormat.mSampleRate = sampleRate; + desiredFormat.mFormatID = kAudioFormatLinearPCM ; + desiredFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked; + desiredFormat.mFramesPerPacket = 1; + desiredFormat.mBitsPerChannel = sizeof( float ) * 8; + desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount; + desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount; + desiredFormat.mChannelsPerFrame = inStreamParams->channelCount; + + /* get the source format */ + ERR_WRAP( AudioUnitGetProperty( + *audioUnit, + kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, + INPUT_ELEMENT, + &sourceFormat, + &sourceSize ) ); + + if( desiredFormat.mSampleRate != sourceFormat.mSampleRate ) + { + UInt32 value = kAudioConverterQuality_Max; + switch( macInputStreamFlags & 0x0700 ) { + case 0x0100: /*paMacCore_ConversionQualityMin:*/ + value=kAudioConverterQuality_Min; + break; + case 0x0200: /*paMacCore_ConversionQualityLow:*/ + value=kAudioConverterQuality_Low; + break; + case 0x0300: /*paMacCore_ConversionQualityMedium:*/ + value=kAudioConverterQuality_Medium; + break; + case 0x0400: /*paMacCore_ConversionQualityHigh:*/ + value=kAudioConverterQuality_High; + break; + } + VDBUG(( "Creating sample rate converter for input" + " to convert from %g to %g\n", + (float)sourceFormat.mSampleRate, + (float)desiredFormat.mSampleRate ) ); + /* create our converter */ + ERR_WRAP( AudioConverterNew( + &sourceFormat, + &desiredFormat, + srConverter ) ); + /* Set quality */ + ERR_WRAP( AudioConverterSetProperty( + *srConverter, + kAudioConverterSampleRateConverterQuality, + sizeof( value ), + &value ) ); + } + } + /* -- set IOProc (callback) -- */ + callbackKey = outStreamParams ? kAudioUnitProperty_SetRenderCallback + : kAudioOutputUnitProperty_SetInputCallback ; + rcbs.inputProc = AudioIOProc; + rcbs.inputProcRefCon = refCon; + ERR_WRAP( AudioUnitSetProperty( + *audioUnit, + callbackKey, + kAudioUnitScope_Output, + outStreamParams ? OUTPUT_ELEMENT : INPUT_ELEMENT, + &rcbs, + sizeof(rcbs)) ); + + if( inStreamParams && outStreamParams && *srConverter ) + ERR_WRAP( AudioUnitSetProperty( + *audioUnit, + kAudioOutputUnitProperty_SetInputCallback, + kAudioUnitScope_Output, + INPUT_ELEMENT, + &rcbs, + sizeof(rcbs)) ); + + /* channel mapping. */ + if(inChannelMap) + { + UInt32 mapSize = inChannelMapSize *sizeof(SInt32); + + //for each channel of desired input, map the channel from + //the device's output channel. + ERR_WRAP( AudioUnitSetProperty(*audioUnit, + kAudioOutputUnitProperty_ChannelMap, + kAudioUnitScope_Output, + INPUT_ELEMENT, + inChannelMap, + mapSize)); + } + if(outChannelMap) + { + UInt32 mapSize = outChannelMapSize *sizeof(SInt32); + + //for each channel of desired output, map the channel from + //the device's output channel. + ERR_WRAP(AudioUnitSetProperty(*audioUnit, + kAudioOutputUnitProperty_ChannelMap, + kAudioUnitScope_Output, + OUTPUT_ELEMENT, + outChannelMap, + mapSize)); + } + /* initialize the audio unit */ + ERR_WRAP( AudioUnitInitialize(*audioUnit) ); + + if( inStreamParams && outStreamParams ) + VDBUG( ("Opened device %ld for input and output.\n", *audioDevice ) ); + else if( inStreamParams ) + VDBUG( ("Opened device %ld for input.\n", *audioDevice ) ); + else if( outStreamParams ) + VDBUG( ("Opened device %ld for output.\n", *audioDevice ) ); + return paNoError; +#undef ERR_WRAP + + error: + CloseComponent( *audioUnit ); + *audioUnit = NULL; + if( result ) + return PaMacCore_SetError( result, line, 1 ); + return paResult; +} + +/* see pa_hostapi.h for a list of validity guarantees made about OpenStream parameters */ +static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi, + PaStream** s, + const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate, + unsigned long framesPerBuffer, + PaStreamFlags streamFlags, + PaStreamCallback *streamCallback, + void *userData ) +{ + PaError result = paNoError; + PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi; + PaMacCoreStream *stream = 0; + int inputChannelCount, outputChannelCount; + PaSampleFormat inputSampleFormat, outputSampleFormat; + PaSampleFormat hostInputSampleFormat, hostOutputSampleFormat; + VVDBUG(("OpenStream(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld SR=%g, FPB=%ld\n", + inputParameters ? inputParameters->channelCount : -1, + inputParameters ? inputParameters->sampleFormat : -1, + outputParameters ? outputParameters->channelCount : -1, + outputParameters ? outputParameters->sampleFormat : -1, + (float) sampleRate, + framesPerBuffer )); + VDBUG( ("Opening Stream.\n") ); + + /*These first few bits of code are from paSkeleton with few modifications.*/ + if( inputParameters ) + { + inputChannelCount = inputParameters->channelCount; + inputSampleFormat = inputParameters->sampleFormat; + + /* unless alternate device specification is supported, reject the use of + paUseHostApiSpecificDeviceSpecification */ + + if( inputParameters->device == paUseHostApiSpecificDeviceSpecification ) + return paInvalidDevice; + + /* check that input device can support inputChannelCount */ + if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels ) + return paInvalidChannelCount; + + /* Host supports interleaved float32 */ + hostInputSampleFormat = paFloat32; + } + else + { + inputChannelCount = 0; + inputSampleFormat = hostInputSampleFormat = paFloat32; /* Surpress 'uninitialised var' warnings. */ + } + + if( outputParameters ) + { + outputChannelCount = outputParameters->channelCount; + outputSampleFormat = outputParameters->sampleFormat; + + /* unless alternate device specification is supported, reject the use of + paUseHostApiSpecificDeviceSpecification */ + + if( outputParameters->device == paUseHostApiSpecificDeviceSpecification ) + return paInvalidDevice; + + /* check that output device can support inputChannelCount */ + if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels ) + return paInvalidChannelCount; + + /* Host supports interleaved float32 */ + hostOutputSampleFormat = paFloat32; + } + else + { + outputChannelCount = 0; + outputSampleFormat = hostOutputSampleFormat = paFloat32; /* Surpress 'uninitialized var' warnings. */ + } + + /* validate platform specific flags */ + if( (streamFlags & paPlatformSpecificFlags) != 0 ) + return paInvalidFlag; /* unexpected platform specific flag */ + + stream = (PaMacCoreStream*)PaUtil_AllocateMemory( sizeof(PaMacCoreStream) ); + if( !stream ) + { + result = paInsufficientMemory; + goto error; + } + + /* If we fail after this point, we my be left in a bad state, with + some data structures setup and others not. So, first thing we + do is initialize everything so that if we fail, we know what hasn't + been touched. + */ + + stream->inputAudioBufferList.mBuffers[0].mData = NULL; + stream->inputRingBuffer.buffer = NULL; + bzero( &stream->blio, sizeof( PaMacBlio ) ); +/* + stream->blio.inputRingBuffer.buffer = NULL; + stream->blio.outputRingBuffer.buffer = NULL; + stream->blio.inputSampleFormat = inputParameters?inputParameters->sampleFormat:0; + stream->blio.inputSampleSize = computeSampleSizeFromFormat(stream->blio.inputSampleFormat); + stream->blio.outputSampleFormat=outputParameters?outputParameters->sampleFormat:0; + stream->blio.outputSampleSize = computeSampleSizeFromFormat(stream->blio.outputSampleFormat); +*/ + stream->inputSRConverter = NULL; + stream->inputUnit = NULL; + stream->outputUnit = NULL; + stream->inputFramesPerBuffer = 0; + stream->outputFramesPerBuffer = 0; + stream->bufferProcessorIsInitialized = FALSE; + + /* assert( streamCallback ) ; */ /* only callback mode is implemented */ + if( streamCallback ) + { + PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation, + &auhalHostApi->callbackStreamInterface, + streamCallback, userData ); + } + else + { + PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation, + &auhalHostApi->blockingStreamInterface, + BlioCallback, &stream->blio ); + } + + PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate ); + + /* -- handle paFramesPerBufferUnspecified -- */ + if( framesPerBuffer == paFramesPerBufferUnspecified ) { + long requested = 64; + if( inputParameters ) + requested = MAX( requested, inputParameters->suggestedLatency * sampleRate / 2 ); + if( outputParameters ) + requested = MAX( requested, outputParameters->suggestedLatency *sampleRate / 2 ); + VDBUG( ("Block Size unspecified. Based on Latency, the user wants a Block Size near: %ld.\n", + requested ) ); + if( requested <= 64 ) { + /*requested a realtively low latency. make sure this is in range of devices */ + /*try to get the device's min natural buffer size and use that (but no smaller than 64).*/ + AudioValueRange audioRange; + size_t size = sizeof( audioRange ); + if( inputParameters ) { + WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device], + 0, + false, + kAudioDevicePropertyBufferFrameSizeRange, + &size, &audioRange ) ); + if( result ) + requested = MAX( requested, audioRange.mMinimum ); + } + if( outputParameters ) { + WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device], + 0, + false, + kAudioDevicePropertyBufferFrameSizeRange, + &size, &audioRange ) ); + if( result ) + requested = MAX( requested, audioRange.mMinimum ); + } + } else { + /* requested a realtively high latency. make sure this is in range of devices */ + /*try to get the device's max natural buffer size and use that (but no larger than 1024).*/ + AudioValueRange audioRange; + size_t size = sizeof( audioRange ); + requested = MIN( requested, 1024 ); + if( inputParameters ) { + WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device], + 0, + false, + kAudioDevicePropertyBufferFrameSizeRange, + &size, &audioRange ) ); + if( result ) + requested = MIN( requested, audioRange.mMaximum ); + } + if( outputParameters ) { + WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device], + 0, + false, + kAudioDevicePropertyBufferFrameSizeRange, + &size, &audioRange ) ); + if( result ) + requested = MIN( requested, audioRange.mMaximum ); + } + } + /* -- double check ranges -- */ + if( requested > 1024 ) requested = 1024; + if( requested < 64 ) requested = 64; + VDBUG(("After querying hardware, setting block size to %ld.\n", requested)); + framesPerBuffer = requested; + } + + /* -- Now we actually open and setup streams. -- */ + if( inputParameters && outputParameters && outputParameters->device == inputParameters->device ) + { /* full duplex. One device. */ + result = OpenAndSetupOneAudioUnit( inputParameters, + outputParameters, + framesPerBuffer, + &(stream->inputFramesPerBuffer), + &(stream->outputFramesPerBuffer), + auhalHostApi, + &(stream->inputUnit), + &(stream->inputSRConverter), + &(stream->inputDevice), + sampleRate, + stream ); + stream->outputUnit = stream->inputUnit; + stream->outputDevice = stream->inputDevice; + if( result != paNoError ) + goto error; + } + else + { /* full duplex, different devices OR simplex */ + result = OpenAndSetupOneAudioUnit( NULL, + outputParameters, + framesPerBuffer, + NULL, + &(stream->outputFramesPerBuffer), + auhalHostApi, + &(stream->outputUnit), + NULL, + &(stream->outputDevice), + sampleRate, + stream ); + if( result != paNoError ) + goto error; + result = OpenAndSetupOneAudioUnit( inputParameters, + NULL, + framesPerBuffer, + &(stream->inputFramesPerBuffer), + NULL, + auhalHostApi, + &(stream->inputUnit), + &(stream->inputSRConverter), + &(stream->inputDevice), + sampleRate, + stream ); + if( result != paNoError ) + goto error; + } + + if( stream->inputUnit ) { + const size_t szfl = sizeof(float); + /* setup the AudioBufferList used for input */ + bzero( &stream->inputAudioBufferList, sizeof( AudioBufferList ) ); + stream->inputAudioBufferList.mNumberBuffers = 1; + stream->inputAudioBufferList.mBuffers[0].mNumberChannels + = inputChannelCount; + stream->inputAudioBufferList.mBuffers[0].mDataByteSize + = stream->inputFramesPerBuffer*inputChannelCount*szfl; + stream->inputAudioBufferList.mBuffers[0].mData + = (float *) calloc( + stream->inputFramesPerBuffer*inputChannelCount, + szfl ); + if( !stream->inputAudioBufferList.mBuffers[0].mData ) + { + result = paInsufficientMemory; + goto error; + } + + /* + * If input and output devs are different or we are doing SR conversion, + * we also need a + * ring buffer to store inpt data while waiting for output + * data. + */ + if( (stream->outputUnit && stream->inputUnit != stream->outputUnit) + || stream->inputSRConverter ) + { + /* May want the ringSize ot initial position in + ring buffer to depend somewhat on sample rate change */ + + void *data; + long ringSize; + + ringSize = computeRingBufferSize( inputParameters, + outputParameters, + stream->inputFramesPerBuffer, + stream->outputFramesPerBuffer, + sampleRate ); + /*ringSize <<= 4; *//*16x bigger, for testing */ + + + /*now, we need to allocate memory for the ring buffer*/ + data = calloc( ringSize, szfl ); + if( !data ) + { + result = paInsufficientMemory; + goto error; + } + + /* now we can initialize the ring buffer */ + PaUtil_InitializeRingBuffer( &stream->inputRingBuffer, + ringSize*szfl, data ) ; + /* advance the read point a little, so we are reading from the + middle of the buffer */ + if( stream->outputUnit ) + PaUtil_AdvanceRingBufferWriteIndex( &stream->inputRingBuffer, ringSize*szfl / RING_BUFFER_ADVANCE_DENOMINATOR ); + } + } + + /* -- initialize Blio Buffer Processors -- */ + if( !streamCallback ) + { + long ringSize; + + ringSize = computeRingBufferSize( inputParameters, + outputParameters, + stream->inputFramesPerBuffer, + stream->outputFramesPerBuffer, + sampleRate ); + result = initializeBlioRingBuffers( &stream->blio, + inputParameters?inputParameters->sampleFormat:0 , + outputParameters?outputParameters->sampleFormat:0 , + MAX(stream->inputFramesPerBuffer,stream->outputFramesPerBuffer), + ringSize, + inputParameters?inputChannelCount:0 , + outputParameters?outputChannelCount:0 ) ; + if( result != paNoError ) + goto error; + } + + /* -- initialize Buffer Processor -- */ + { + unsigned long maxHostFrames = stream->inputFramesPerBuffer; + if( stream->outputFramesPerBuffer > maxHostFrames ) + maxHostFrames = stream->outputFramesPerBuffer; + result = PaUtil_InitializeBufferProcessor( &stream->bufferProcessor, + inputChannelCount, inputSampleFormat, + hostInputSampleFormat, + outputChannelCount, outputSampleFormat, + hostOutputSampleFormat, + sampleRate, + streamFlags, + framesPerBuffer, + /* If sample rate conversion takes place, the buffer size + will not be known. */ + maxHostFrames, + stream->inputSRConverter + ? paUtilUnknownHostBufferSize + : paUtilBoundedHostBufferSize, + streamCallback ? streamCallback : BlioCallback, + streamCallback ? userData : &stream->blio ); + if( result != paNoError ) + goto error; + } + stream->bufferProcessorIsInitialized = TRUE; + + /* + IMPLEMENT ME: initialise the following fields with estimated or actual + values. + I think this is okay the way it is br 12/1/05 + maybe need to change input latency estimate if IO devs differ + */ + stream->streamRepresentation.streamInfo.inputLatency = + PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor)/sampleRate; + stream->streamRepresentation.streamInfo.outputLatency = + PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor)/sampleRate; + stream->streamRepresentation.streamInfo.sampleRate = sampleRate; + + stream->sampleRate = sampleRate; + stream->outDeviceSampleRate = 0; + if( stream->outputUnit ) { + Float64 rate; + UInt32 size = sizeof( rate ); + result = ERR( AudioDeviceGetProperty( stream->outputDevice, + 0, + FALSE, + kAudioDevicePropertyNominalSampleRate, + &size, &rate ) ); + if( result ) + goto error; + stream->outDeviceSampleRate = rate; + } + stream->inDeviceSampleRate = 0; + if( stream->inputUnit ) { + Float64 rate; + UInt32 size = sizeof( rate ); + result = ERR( AudioDeviceGetProperty( stream->inputDevice, + 0, + TRUE, + kAudioDevicePropertyNominalSampleRate, + &size, &rate ) ); + if( result ) + goto error; + stream->inDeviceSampleRate = rate; + } + stream->userInChan = inputChannelCount; + stream->userOutChan = outputChannelCount; + + stream->isTimeSet = FALSE; + stream->state = STOPPED; + stream->xrunFlags = 0; + + *s = (PaStream*)stream; + + return result; + +error: + CloseStream( stream ); + return result; +} + +PaTime GetStreamTime( PaStream *s ) +{ + /* FIXME: I am not at all sure this timing info stuff is right. + patest_sine_time reports negative latencies, which is wierd.*/ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + AudioTimeStamp timeStamp; + + VVDBUG(("GetStreamTime()\n")); + + if ( !stream->isTimeSet ) + return (PaTime)0; + + if ( stream->outputDevice ) { + AudioDeviceGetCurrentTime( stream->outputDevice, &timeStamp); + return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->outDeviceSampleRate; + } else if ( stream->inputDevice ) { + AudioDeviceGetCurrentTime( stream->inputDevice, &timeStamp); + return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->inDeviceSampleRate; + } else { + return (PaTime)0; + } +} + +static void setStreamStartTime( PaStream *stream ) +{ + /* FIXME: I am not at all sure this timing info stuff is right. + patest_sine_time reports negative latencies, which is wierd.*/ + PaMacCoreStream *s = (PaMacCoreStream *) stream; + VVDBUG(("setStreamStartTime()\n")); + if( s->outputDevice ) + AudioDeviceGetCurrentTime( s->outputDevice, &s->startTime); + else if( s->inputDevice ) + AudioDeviceGetCurrentTime( s->inputDevice, &s->startTime); + else + bzero( &s->startTime, sizeof( s->startTime ) ); + + //FIXME: we need a memory barier here + + s->isTimeSet = TRUE; +} + + +static PaTime TimeStampToSecs(PaMacCoreStream *stream, const AudioTimeStamp* timeStamp) +{ + VVDBUG(("TimeStampToSecs()\n")); + //printf( "ATS: %lu, %g, %g\n", timeStamp->mFlags, timeStamp->mSampleTime, timeStamp->mRateScalar ); + if (timeStamp->mFlags & kAudioTimeStampSampleTimeValid) + return (timeStamp->mSampleTime / stream->sampleRate); + else + return 0; +} + +#define RING_BUFFER_EMPTY (1000) + +static OSStatus ringBufferIOProc( AudioConverterRef inAudioConverter, + UInt32*ioDataSize, + void** outData, + void*inUserData ) +{ + void *dummyData; + long dummySize; + PaUtilRingBuffer *rb = (PaUtilRingBuffer *) inUserData; + + VVDBUG(("ringBufferIOProc()\n")); + + assert( sizeof( UInt32 ) == sizeof( long ) ); + if( PaUtil_GetRingBufferReadAvailable( rb ) == 0 ) { + *outData = NULL; + *ioDataSize = 0; + return RING_BUFFER_EMPTY; + } + PaUtil_GetRingBufferReadRegions( rb, *ioDataSize, + outData, (long *)ioDataSize, + &dummyData, &dummySize ); + + assert( *ioDataSize ); + PaUtil_AdvanceRingBufferReadIndex( rb, *ioDataSize ); + + return noErr; +} + +/* + * Called by the AudioUnit API to process audio from the sound card. + * This is where the magic happens. + */ +/* FEEDBACK: there is a lot of redundant code here because of how all the cases differ. This makes it hard to maintain, so if there are suggestinos for cleaning it up, I'm all ears. */ +static OSStatus AudioIOProc( void *inRefCon, + AudioUnitRenderActionFlags *ioActionFlags, + const AudioTimeStamp *inTimeStamp, + UInt32 inBusNumber, + UInt32 inNumberFrames, + AudioBufferList *ioData ) +{ + unsigned long framesProcessed = 0; + PaStreamCallbackTimeInfo timeInfo = {0,0,0}; + PaMacCoreStream *stream = (PaMacCoreStream*)inRefCon; + const bool isRender = inBusNumber == OUTPUT_ELEMENT; + int callbackResult = paContinue ; + + VVDBUG(("AudioIOProc()\n")); + + PaUtil_BeginCpuLoadMeasurement( &stream->cpuLoadMeasurer ); + + /* -----------------------------------------------------------------*\ + This output may be useful for debugging, + But printing durring the callback is a bad enough idea that + this is not enabled by enableing the usual debugging calls. + \* -----------------------------------------------------------------*/ + /* + static int renderCount = 0; + static int inputCount = 0; + printf( "------------------- starting reder/input\n" ); + if( isRender ) + printf("Render callback (%d):\t", ++renderCount); + else + printf("Input callback (%d):\t", ++inputCount); + printf( "Call totals: %d (input), %d (render)\n", inputCount, renderCount ); + + printf( "--- inBusNumber: %lu\n", inBusNumber ); + printf( "--- inNumberFrames: %lu\n", inNumberFrames ); + printf( "--- %x ioData\n", (unsigned) ioData ); + if( ioData ) + { + int i=0; + printf( "--- ioData.mNumBuffers %lu: \n", ioData->mNumberBuffers ); + for( i=0; i<ioData->mNumberBuffers; ++i ) + printf( "--- ioData buffer %d size: %lu.\n", i, ioData->mBuffers[i].mDataByteSize ); + } + ----------------------------------------------------------------- */ + + if( !stream->isTimeSet ) + setStreamStartTime( stream ); + + if( isRender ) { + AudioTimeStamp currentTime; + timeInfo.outputBufferDacTime = TimeStampToSecs(stream, inTimeStamp); + AudioDeviceGetCurrentTime(stream->outputDevice, ¤tTime); + timeInfo.currentTime = TimeStampToSecs(stream, ¤tTime); + } + if( isRender && stream->inputUnit == stream->outputUnit ) + timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp); + if( !isRender ) { + AudioTimeStamp currentTime; + timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp); + AudioDeviceGetCurrentTime(stream->inputDevice, ¤tTime); + timeInfo.currentTime = TimeStampToSecs(stream, ¤tTime); + } + + //printf( "---%g, %g, %g\n", timeInfo.inputBufferAdcTime, timeInfo.currentTime, timeInfo.outputBufferDacTime ); + + if( isRender && stream->inputUnit == stream->outputUnit + && !stream->inputSRConverter ) + { + /* --------- Full Duplex, One Device, no SR Conversion ------- + * + * This is the lowest latency case, and also the simplest. + * Input data and output data are available at the same time. + * we do not use the input SR converter or the input ring buffer. + * + */ + OSErr err = 0; + unsigned long frames; + + /* -- start processing -- */ + PaUtil_BeginBufferProcessing( &(stream->bufferProcessor), + &timeInfo, + stream->xrunFlags ); + stream->xrunFlags = 0; + + /* -- compute frames. do some checks -- */ + assert( ioData->mNumberBuffers == 1 ); + assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan ); + frames = ioData->mBuffers[0].mDataByteSize; + frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels; + /* -- copy and process input data -- */ + err= AudioUnitRender(stream->inputUnit, + ioActionFlags, + inTimeStamp, + INPUT_ELEMENT, + inNumberFrames, + &stream->inputAudioBufferList ); + /* FEEDBACK: I'm not sure what to do when this call fails */ + assert( !err ); + + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + stream->inputAudioBufferList.mBuffers[0].mData, + stream->inputAudioBufferList.mBuffers[0].mNumberChannels); + /* -- Copy and process output data -- */ + PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor), + 0, + ioData->mBuffers[0].mData, + ioData->mBuffers[0].mNumberChannels); + /* -- complete processing -- */ + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + } + else if( isRender ) + { + /* -------- Output Side of Full Duplex (Separate Devices or SR Conversion) + * -- OR Simplex Output + * + * This case handles output data as in the full duplex case, + * and, if there is input data, reads it off the ring buffer + * and into the PA buffer processor. If sample rate conversion + * is required on input, that is done here as well. + */ + unsigned long frames; + + /* Sometimes, when stopping a duplex stream we get erroneous + xrun flags, so if this is our last run, clear the flags. */ + int xrunFlags = stream->xrunFlags; +/* + if( xrunFlags & paInputUnderflow ) + printf( "input underflow.\n" ); + if( xrunFlags & paInputOverflow ) + printf( "input overflow.\n" ); +*/ + if( stream->state == STOPPING || stream->state == CALLBACK_STOPPED ) + xrunFlags = 0; + + /* -- start processing -- */ + PaUtil_BeginBufferProcessing( &(stream->bufferProcessor), + &timeInfo, + xrunFlags ); + stream->xrunFlags = 0; /* FEEDBACK: we only send flags to Buf Proc once */ + + /* -- Copy and process output data -- */ + assert( ioData->mNumberBuffers == 1 ); + frames = ioData->mBuffers[0].mDataByteSize; + frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels; + assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan ); + PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor), + 0, + ioData->mBuffers[0].mData, + ioData->mBuffers[0].mNumberChannels); + + /* -- copy and process input data, and complete processing -- */ + if( stream->inputUnit ) { + const int flsz = sizeof( float ); + /* Here, we read the data out of the ring buffer, through the + audio converter. */ + int inChan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels; + if( stream->inputSRConverter ) + { + OSStatus err; + UInt32 size; + float data[ inChan * frames ]; + size = sizeof( data ); + err = AudioConverterFillBuffer( + stream->inputSRConverter, + ringBufferIOProc, + &stream->inputRingBuffer, + &size, + (void *)&data ); + if( err == RING_BUFFER_EMPTY ) + { /*the ring buffer callback underflowed */ + err = 0; + bzero( ((char *)data) + size, sizeof(data)-size ); + stream->xrunFlags |= paInputUnderflow; + } + ERR( err ); + assert( !err ); + + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data, + inChan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + } + else + { + /* Without the AudioConverter is actually a bit more complex + because we have to do a little buffer processing that the + AudioConverter would otherwise handle for us. */ + void *data1, *data2; + long size1, size2; + PaUtil_GetRingBufferReadRegions( &stream->inputRingBuffer, + inChan*frames*flsz, + &data1, &size1, + &data2, &size2 ); + if( size1 / ( flsz * inChan ) == frames ) { + /* simplest case: all in first buffer */ + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data1, + inChan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1 ); + } else if( ( size1 + size2 ) / ( flsz * inChan ) < frames ) { + /*we underflowed. take what data we can, zero the rest.*/ + float data[frames*inChan]; + if( size1 ) + memcpy( data, data1, size1 ); + if( size2 ) + memcpy( data+size1, data2, size2 ); + bzero( data+size1+size2, frames*flsz*inChan - size1 - size2 ); + + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames ); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data, + inChan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + PaUtil_AdvanceRingBufferReadIndex( &stream->inputRingBuffer, + size1+size2 ); + /* flag underflow */ + stream->xrunFlags |= paInputUnderflow; + } else { + /*we got all the data, but split between buffers*/ + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), + size1 / ( flsz * inChan ) ); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data1, + inChan ); + PaUtil_Set2ndInputFrameCount( &(stream->bufferProcessor), + size2 / ( flsz * inChan ) ); + PaUtil_Set2ndInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data2, + inChan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1+size2 ); + } + } + } else { + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + } + + } + else + { + /* ------------------ Input + * + * First, we read off the audio data and put it in the ring buffer. + * if this is an input-only stream, we need to process it more, + * otherwise, we let the output case deal with it. + */ + OSErr err = 0; + int chan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels ; + /* FIXME: looping here may not actually be necessary, but it was something I tried in testing. */ + do { + err= AudioUnitRender(stream->inputUnit, + ioActionFlags, + inTimeStamp, + INPUT_ELEMENT, + inNumberFrames, + &stream->inputAudioBufferList ); + if( err == -10874 ) + inNumberFrames /= 2; + } while( err == -10874 && inNumberFrames > 1 ); + /* FEEDBACK: I'm not sure what to do when this call fails */ + ERR( err ); + assert( !err ); + if( stream->inputSRConverter || stream->outputUnit ) + { + /* If this is duplex or we use a converter, put the data + into the ring buffer. */ + long bytesIn, bytesOut; + bytesIn = sizeof( float ) * inNumberFrames * chan; + bytesOut = PaUtil_WriteRingBuffer( &stream->inputRingBuffer, + stream->inputAudioBufferList.mBuffers[0].mData, + bytesIn ); + if( bytesIn != bytesOut ) + stream->xrunFlags |= paInputOverflow ; + } + else + { + /* for simplex input w/o SR conversion, + just pop the data into the buffer processor.*/ + PaUtil_BeginBufferProcessing( &(stream->bufferProcessor), + &timeInfo, + stream->xrunFlags ); + stream->xrunFlags = 0; + + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), inNumberFrames); + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + stream->inputAudioBufferList.mBuffers[0].mData, + chan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + } + if( !stream->outputUnit && stream->inputSRConverter ) + { + /* ------------------ Simplex Input w/ SR Conversion + * + * if this is a simplex input stream, we need to read off the buffer, + * do our sample rate conversion and pass the results to the buffer + * processor. + * The logic here is complicated somewhat by the fact that we don't + * know how much data is available, so we loop on reasonably sized + * chunks, and let the BufferProcessor deal with the rest. + * + */ + /*This might be too big or small depending on SR conversion*/ + float data[ chan * inNumberFrames ]; + OSStatus err; + do + { /*Run the buffer processor until we are out of data*/ + UInt32 size; + long f; + + size = sizeof( data ); + err = AudioConverterFillBuffer( + stream->inputSRConverter, + ringBufferIOProc, + &stream->inputRingBuffer, + &size, + (void *)data ); + if( err != RING_BUFFER_EMPTY ) + ERR( err ); + assert( err == 0 || err == RING_BUFFER_EMPTY ); + + f = size / ( chan * sizeof(float) ); + PaUtil_SetInputFrameCount( &(stream->bufferProcessor), f ); + if( f ) + { + PaUtil_BeginBufferProcessing( &(stream->bufferProcessor), + &timeInfo, + stream->xrunFlags ); + stream->xrunFlags = 0; + + PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor), + 0, + data, + chan ); + framesProcessed = + PaUtil_EndBufferProcessing( &(stream->bufferProcessor), + &callbackResult ); + } + } while( callbackResult == paContinue && !err ); + } + } + + switch( callbackResult ) + { + case paContinue: break; + case paComplete: + case paAbort: + stream->isTimeSet = FALSE; + stream->state = CALLBACK_STOPPED ; + if( stream->outputUnit ) + AudioOutputUnitStop(stream->outputUnit); + if( stream->inputUnit ) + AudioOutputUnitStop(stream->inputUnit); + break; + } + + PaUtil_EndCpuLoadMeasurement( &stream->cpuLoadMeasurer, framesProcessed ); + return noErr; +} + + +/* + When CloseStream() is called, the multi-api layer ensures that + the stream has already been stopped or aborted. +*/ +static PaError CloseStream( PaStream* s ) +{ + /* This may be called from a failed OpenStream. + Therefore, each piece of info is treated seperately. */ + PaError result = paNoError; + PaMacCoreStream *stream = (PaMacCoreStream*)s; + + VVDBUG(("CloseStream()\n")); + VDBUG( ( "Closing stream.\n" ) ); + + if( stream ) { + if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) { + AudioUnitUninitialize( stream->outputUnit ); + CloseComponent( stream->outputUnit ); + } + stream->outputUnit = NULL; + if( stream->inputUnit ) + { + AudioUnitUninitialize( stream->inputUnit ); + CloseComponent( stream->inputUnit ); + stream->inputUnit = NULL; + } + if( stream->inputRingBuffer.buffer ) + free( (void *) stream->inputRingBuffer.buffer ); + stream->inputRingBuffer.buffer = NULL; + /*TODO: is there more that needs to be done on error + from AudioConverterDispose?*/ + if( stream->inputSRConverter ) + ERR( AudioConverterDispose( stream->inputSRConverter ) ); + stream->inputSRConverter = NULL; + if( stream->inputAudioBufferList.mBuffers[0].mData ) + free( stream->inputAudioBufferList.mBuffers[0].mData ); + stream->inputAudioBufferList.mBuffers[0].mData = NULL; + + result = destroyBlioRingBuffers( &stream->blio ); + if( result ) + return result; + if( stream->bufferProcessorIsInitialized ) + PaUtil_TerminateBufferProcessor( &stream->bufferProcessor ); + PaUtil_TerminateStreamRepresentation( &stream->streamRepresentation ); + PaUtil_FreeMemory( stream ); + } + + return result; +} + + +static PaError StartStream( PaStream *s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + OSErr result = noErr; + VVDBUG(("StartStream()\n")); + VDBUG( ( "Starting stream.\n" ) ); + +#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0) + + /*FIXME: maybe want to do this on close/abort for faster start? */ + PaUtil_ResetBufferProcessor( &stream->bufferProcessor ); + if( stream->inputSRConverter ) + ERR_WRAP( AudioConverterReset( stream->inputSRConverter ) ); + + /* -- start -- */ + stream->state = ACTIVE; + if( stream->inputUnit ) { + ERR_WRAP( AudioOutputUnitStart(stream->inputUnit) ); + } + if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) { + ERR_WRAP( AudioOutputUnitStart(stream->outputUnit) ); + } + + //setStreamStartTime( stream ); + //stream->isTimeSet = TRUE; + + return paNoError; +#undef ERR_WRAP +} + +// it's not clear from appl's docs that this really waits +// until all data is flushed. +static ComponentResult BlockWhileAudioUnitIsRunning( AudioUnit audioUnit, AudioUnitElement element ) +{ + Boolean isRunning = 1; + while( isRunning ) { + UInt32 s = sizeof( isRunning ); + ComponentResult err = AudioUnitGetProperty( audioUnit, kAudioOutputUnitProperty_IsRunning, kAudioUnitScope_Global, element, &isRunning, &s ); + if( err ) + return err; + Pa_Sleep( 100 ); + } + return noErr; +} + +static PaError StopStream( PaStream *s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + OSErr result = noErr; + PaError paErr; + VVDBUG(("StopStream()\n")); + + VDBUG( ("Waiting for BLIO.\n") ); + waitUntilBlioWriteBufferIsFlushed( &stream->blio ); + VDBUG( ( "Stopping stream.\n" ) ); + + stream->isTimeSet = FALSE; + stream->state = STOPPING; + +#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0) + /* -- stop and reset -- */ + if( stream->inputUnit == stream->outputUnit && stream->inputUnit ) + { + ERR_WRAP( AudioOutputUnitStop(stream->inputUnit) ); + ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,0) ); + ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,1) ); + ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 1) ); + ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 0) ); + } + else + { + if( stream->inputUnit ) + { + ERR_WRAP(AudioOutputUnitStop(stream->inputUnit) ); + ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,1) ); + ERR_WRAP(AudioUnitReset(stream->inputUnit,kAudioUnitScope_Global,1)); + } + if( stream->outputUnit ) + { + ERR_WRAP(AudioOutputUnitStop(stream->outputUnit)); + ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->outputUnit,0) ); + ERR_WRAP(AudioUnitReset(stream->outputUnit,kAudioUnitScope_Global,0)); + } + } + if( stream->inputRingBuffer.buffer ) { + PaUtil_FlushRingBuffer( &stream->inputRingBuffer ); + bzero( (void *)stream->inputRingBuffer.buffer, + stream->inputRingBuffer.bufferSize ); + /* advance the write point a little, so we are reading from the + middle of the buffer. We'll need extra at the end because + testing has shown that this helps. */ + if( stream->outputUnit ) + PaUtil_AdvanceRingBufferWriteIndex( &stream->inputRingBuffer, + stream->inputRingBuffer.bufferSize + / RING_BUFFER_ADVANCE_DENOMINATOR ); + } + + stream->xrunFlags = 0; + stream->state = STOPPED; + + paErr = resetBlioRingBuffers( &stream->blio ); + if( paErr ) + return paErr; + +/* + //stream->isTimeSet = FALSE; +*/ + + VDBUG( ( "Stream Stopped.\n" ) ); + return paNoError; +#undef ERR_WRAP +} + +static PaError AbortStream( PaStream *s ) +{ + VVDBUG(("AbortStream()->StopStream()\n")); + VDBUG( ( "Aborting stream.\n" ) ); + /* We have nothing faster than StopStream. */ + return StopStream(s); +} + + +static PaError IsStreamStopped( PaStream *s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + VVDBUG(("IsStreamStopped()\n")); + + return stream->state == STOPPED ? 1 : 0; +} + + +static PaError IsStreamActive( PaStream *s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + VVDBUG(("IsStreamActive()\n")); + return ( stream->state == ACTIVE || stream->state == STOPPING ); +} + + +static double GetStreamCpuLoad( PaStream* s ) +{ + PaMacCoreStream *stream = (PaMacCoreStream*)s; + VVDBUG(("GetStreamCpuLoad()\n")); + + return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer ); +} |