diff options
Diffstat (limited to 'pd/portaudio/pa_mac_core/pa_mac_core.c')
| -rw-r--r-- | pd/portaudio/pa_mac_core/pa_mac_core.c | 2105 |
1 files changed, 0 insertions, 2105 deletions
diff --git a/pd/portaudio/pa_mac_core/pa_mac_core.c b/pd/portaudio/pa_mac_core/pa_mac_core.c deleted file mode 100644 index 16eb0824..00000000 --- a/pd/portaudio/pa_mac_core/pa_mac_core.c +++ /dev/null @@ -1,2105 +0,0 @@ -/* - * This is the AUHAL implementation of portaudio. Hopefully this will - * one day replace pa_mac_core. - * - * 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. - * - * Bjorn Roche and XO Audio LLC reserve no rights to this code. - * The maintainers of PortAudio may redistribute and modify the code and - * licenses as they deam appropriate. - * - * 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. - * - * 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. - * - * 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. - */ - -/** - @file pa_mac_core - @author Bjorn Roche - @brief AUHAL implementation of PortAudio -*/ - -#include <string.h> /* strlen(), memcmp() etc. */ - -#include <AudioUnit/AudioUnit.h> -#include <AudioToolbox/AudioToolbox.h> - - -#include "pa_util.h" -#include "pa_allocation.h" -#include "pa_hostapi.h" -#include "pa_stream.h" -#include "pa_cpuload.h" -#include "pa_process.h" -#include "../pablio/ringbuffer.h" -#include "pa_mac_core.h" - -#ifndef MIN -#define MIN(a, b) (((a)<(b))?(a):(b)) -#endif - -#ifndef MAX -#define MAX(a, b) (((a)<(b))?(b):(a)) -#endif - -/* prototypes for functions declared in this file */ - -#ifdef __cplusplus -extern "C" -{ -#endif /* __cplusplus */ - -PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index ); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - -#define ERR(mac_error) PaMacCore_SetError(mac_error, __LINE__, 1 ) -#define WARNING(mac_error) PaMacCore_SetError(mac_error, __LINE__, 0 ) - -/* Help keep track of AUHAL element numbers */ -#define INPUT_ELEMENT (1) -#define OUTPUT_ELEMENT (0) - -/* Normal level of debugging: fine for most apps that don't mind the occational warning being printf'ed */ -/* - */ -#define MAC_CORE_DEBUG -#ifdef MAC_CORE_DEBUG -# define DBUG(MSG) do { printf("||PaMacCore (AUHAL)|| "); printf MSG ; fflush(stdout); } while(0) -#else -# define DBUG(MSG) -#endif - -/* Verbose Debugging: useful for developement */ -/* -#define MAC_CORE_VERBOSE_DEBUG - */ -#ifdef MAC_CORE_VERBOSE_DEBUG -# define VDBUG(MSG) do { printf("||PaMacCore (v )|| "); printf MSG ; fflush(stdout); } while(0) -#else -# define VDBUG(MSG) -#endif - -/* Very Verbose Debugging: Traces every call. */ -/* -#define MAC_CORE_VERY_VERBOSE_DEBUG - */ -#ifdef MAC_CORE_VERY_VERBOSE_DEBUG -# define VVDBUG(MSG) do { printf("||PaMacCore (vv)|| "); printf MSG ; fflush(stdout); } while(0) -#else -# define VVDBUG(MSG) -#endif - -#define RING_BUFFER_ADVANCE_DENOMINATOR (4) - -/* Some utilities that sort of belong here, but were getting too unweildy */ -#include "pa_mac_core_utilities.c" -/* Special purpose ring buffer just for pa_mac_core input processing. */ -/* #include "pa_mac_core_input_ring_buffer.c" */ -#include "../pablio/ringbuffer.c" - - -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 ReadStream( PaStream* stream, void *buffer, unsigned long frames ); -static PaError WriteStream( PaStream* stream, const void *buffer, unsigned long frames ); -static signed long GetStreamReadAvailable( PaStream* stream ); -static signed long GetStreamWriteAvailable( PaStream* stream ); -/* PaMacAUHAL - host api datastructure specific to this implementation */ -typedef struct -{ - PaUtilHostApiRepresentation inheritedHostApiRep; - PaUtilStreamInterface callbackStreamInterface; - PaUtilStreamInterface blockingStreamInterface; - - PaUtilAllocationGroup *allocations; - - /* implementation specific data goes here */ - long devCount; - AudioDeviceID *devIds; /*array of all audio devices*/ - AudioDeviceID defaultIn; - AudioDeviceID defaultOut; -} -PaMacAUHAL; - -/* stream data structure specifically for this implementation */ -typedef struct PaMacCoreStream -{ - PaUtilStreamRepresentation streamRepresentation; - PaUtilCpuLoadMeasurer cpuLoadMeasurer; - PaUtilBufferProcessor bufferProcessor; - - /* implementation specific data goes here */ - bool bufferProcessorIsInitialized; - AudioUnit inputUnit; - AudioUnit outputUnit; - AudioDeviceID inputDevice; - AudioDeviceID outputDevice; - size_t userInChan; - size_t userOutChan; - size_t inputFramesPerBuffer; - size_t outputFramesPerBuffer; - /* We use this ring buffer when input and out devs are different. */ - RingBuffer inputRingBuffer; - /* We may need to do SR conversion on input. */ - AudioConverterRef inputSRConverter; - /* We need to preallocate an inputBuffer for reading data. */ - AudioBufferList inputAudioBufferList; - AudioTimeStamp startTime; - //volatile bool isTimeSet; - volatile PaStreamCallbackFlags xrunFlags; - volatile enum { - STOPPED = 0, /* playback is completely stopped, - and the user has called StopStream(). */ - CALLBACK_STOPPED = 1, /* callback has requested stop, - but user has not yet called StopStream(). */ - STOPPING = 2, /* The stream is in the process of closing. - This state is just used internally; - externally it is indistinguishable from - ACTIVE.*/ - ACTIVE = 3 /* The stream is active and running. */ - } state; - double sampleRate; -} -PaMacCoreStream; - -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; - /* FEEDBACK: these calls could fail, in which case default in and out will - be unknown devices or could be undefined. Do I need to be more - rigorous here? */ - AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, - &size, - &auhalHostApi->defaultIn); - AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, - &size, - &auhalHostApi->defaultOut); - 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; - 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); - err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist)); - if (err) - return err; - - 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 = .01; - deviceInfo->defaultLowOutputLatency = .01; - deviceInfo->defaultHighOutputLatency = .01; - propSize = sizeof(UInt32); - err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency)); - if (!err) - { - double secondLatency = frameLatency / deviceInfo->defaultSampleRate; - if (isInput) - { - deviceInfo->defaultLowInputLatency = secondLatency; - deviceInfo->defaultHighInputLatency = secondLatency; - } - else - { - deviceInfo->defaultLowOutputLatency = secondLatency; - deviceInfo->defaultHighOutputLatency = secondLatency; - } - } - } - return paNoError; -} - -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; - UInt32 callbackKey; - AURenderCallbackStruct rcbs; - unsigned long macInputStreamFlags = paMacCorePlayNice; - unsigned long macOutputStreamFlags = paMacCorePlayNice; - - 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; - if( outStreamParams && outStreamParams->hostApiSpecificStreamInfo ) - macOutputStreamFlags= - ((paMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo) - ->flags; - /* 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; - 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; - 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 & paMacCore_ChangeDeviceParameters ) { - bool requireExact; - requireExact=macInputStreamFlags&paMacCore_FailIfConversionRequired; - 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 & paMacCore_ChangeDeviceParameters ) { - bool requireExact; - requireExact=macOutputStreamFlags&paMacCore_FailIfConversionRequired; - 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)) ); - - /*IMPLEMENTME: may need to worry about channel mapping.*/ - - /* 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 = paInt16; /* 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; - 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, streamCallback, userData ); - } - - 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 */ - /* Calculate an appropriate ring buffer size. It must be at least - 3x framesPerBuffer and 2x suggested latency and it must be a - power of 2. FEEDBACK: too liberal/conservative/another way?*/ - double latency; - int index, i; - void *data; - long ringSize; - if( !outputParameters ) - latency = inputParameters->suggestedLatency; - else - latency = MAX( inputParameters->suggestedLatency, - outputParameters->suggestedLatency ); - ringSize = latency * sampleRate * 2 * inputChannelCount; - VDBUG( ( "suggested latency: %d\n", (int) (latency*sampleRate) ) ); - if( ringSize < stream->inputFramesPerBuffer * 3 ) - ringSize = stream->inputFramesPerBuffer * 3 * inputChannelCount; - if( outputParameters && ringSize < stream->outputFramesPerBuffer * 3 ) - ringSize = stream->outputFramesPerBuffer * 3 * inputChannelCount; - VDBUG(("inFramesPerBuffer:%d\n",(int)stream->inputFramesPerBuffer)); - if( outputParameters ) - VDBUG(("outFramesPerBuffer:%d\n", - (int)stream->outputFramesPerBuffer)); - VDBUG(("Ringbuffer size (1): %d\n", (int)ringSize )); - - /* round up to the next power of 2 */ - index = -1; - for( i=0; i<sizeof(long)*8; ++i ) - if( ringSize >> i & 0x01 ) - index = i; - assert( index > 0 ); - if( ringSize <= ( 0x01 << index ) ) - ringSize = 0x01 << index ; - else - ringSize = 0x01 << ( index + 1 ); - - /*ringSize <<= 4; *//*16x bigger, for testing */ - - VDBUG(( "Final Ringbuffer size (2): %d\n", (int)ringSize )); - - /*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 */ - assert( 0 == - RingBuffer_Init( &stream->inputRingBuffer, - ringSize*szfl, data ) ); - /* advance the read point a little, so we are reading from the - middle of the buffer */ - if( stream->outputUnit ) - RingBuffer_AdvanceWriteIndex( &stream->inputRingBuffer, ringSize*szfl / RING_BUFFER_ADVANCE_DENOMINATOR ); - } - } - - /* -- 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, userData ); - 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); - stream->streamRepresentation.streamInfo.outputLatency = - PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor); - stream->streamRepresentation.streamInfo.sampleRate = sampleRate; - - stream->sampleRate = sampleRate; - stream->userInChan = inputChannelCount; - stream->userOutChan = outputChannelCount; - - //stream->isTimeSet = FALSE; - stream->state = STOPPED; - stream->xrunFlags = 0; - - *s = (PaStream*)stream; - - setStreamStartTime( 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); - else if ( stream->inputDevice ) - AudioDeviceGetCurrentTime( stream->inputDevice, &timeStamp); - else - return (PaTime)0; - - return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->sampleRate; -} - -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.*/ - VVDBUG(("setStreamStartTime()\n")); - PaMacCoreStream *s = (PaMacCoreStream *) stream; - if( s->inputDevice ) - AudioDeviceGetCurrentTime( s->inputDevice, &s->startTime); - else - AudioDeviceGetCurrentTime( s->outputDevice, &s->startTime); -} - - -static PaTime TimeStampToSecs(PaMacCoreStream *stream, const AudioTimeStamp* timeStamp) -{ - VVDBUG(("TimeStampToSecs()\n")); - 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; - RingBuffer *rb = (RingBuffer *) inUserData; - - VVDBUG(("ringBufferIOProc()\n")); - - assert( sizeof( UInt32 ) == sizeof( long ) ); - if( RingBuffer_GetReadAvailable( rb ) == 0 ) { - *outData = NULL; - *ioDataSize = 0; - return RING_BUFFER_EMPTY; - } - RingBuffer_GetReadRegions( rb, *ioDataSize, - outData, (long *)ioDataSize, - &dummyData, &dummySize ); - - assert( *ioDataSize ); - RingBuffer_AdvanceReadIndex( 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 ); - - //if( !stream->isTimeSet ) - // setStreamStartTime( stream ); - //stream->isTimeSet = TRUE; - - - /* -----------------------------------------------------------------*\ - 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( 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); - } - - - 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; - RingBuffer_GetReadRegions( &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 ); - RingBuffer_AdvanceReadIndex(&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 ); - RingBuffer_AdvanceReadIndex( &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 ); - RingBuffer_AdvanceReadIndex(&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 = RingBuffer_Write( &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->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( 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; - - 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) ); - } - - return paNoError; -#undef ERR_WRAP -} - - -static PaError StopStream( PaStream *s ) -{ - PaMacCoreStream *stream = (PaMacCoreStream*)s; - OSErr result = noErr; - VVDBUG(("StopStream()\n")); - VDBUG( ( "Stopping stream.\n" ) ); - - 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( 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(AudioUnitReset(stream->inputUnit,kAudioUnitScope_Global,1)); - } - if( stream->outputUnit ) - { - ERR_WRAP(AudioOutputUnitStop(stream->outputUnit)); - ERR_WRAP(AudioUnitReset(stream->outputUnit,kAudioUnitScope_Global,0)); - } - } - if( stream->inputRingBuffer.buffer ) { - RingBuffer_Flush( &stream->inputRingBuffer ); - bzero(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 ) - RingBuffer_AdvanceWriteIndex( &stream->inputRingBuffer, - stream->inputRingBuffer.bufferSize - / RING_BUFFER_ADVANCE_DENOMINATOR ); - } - - //stream->isTimeSet = FALSE; - stream->xrunFlags = 0; - stream->state = STOPPED; - - 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 ); -} - - -/* - As separate stream interfaces are used for blocking and callback - streams, the following functions can be guaranteed to only be called - for blocking streams. IMPLEMENTME: no blocking interface yet! -*/ - -static PaError ReadStream( PaStream* s, - void *buffer, - unsigned long frames ) -{ - PaMacCoreStream *stream = (PaMacCoreStream*)s; - VVDBUG(("ReadStream()\n")); - - /* suppress unused variable warnings */ - (void) buffer; - (void) frames; - (void) stream; - - /* IMPLEMENT ME, see portaudio.h for required behavior*/ - - return paNoError; -} - - -static PaError WriteStream( PaStream* s, - const void *buffer, - unsigned long frames ) -{ - PaMacCoreStream *stream = (PaMacCoreStream*)s; - VVDBUG(("WriteStream()\n")); - - /* suppress unused variable warnings */ - (void) buffer; - (void) frames; - (void) stream; - - /* IMPLEMENT ME, see portaudio.h for required behavior*/ - - return paNoError; -} - - -static signed long GetStreamReadAvailable( PaStream* s ) -{ - PaMacCoreStream *stream = (PaMacCoreStream*)s; - VVDBUG(("GetStreamReadAvailable()\n")); - - /* suppress unused variable warnings */ - (void) stream; - - /* IMPLEMENT ME, see portaudio.h for required behavior*/ - - return 0; -} - - -static signed long GetStreamWriteAvailable( PaStream* s ) -{ - PaMacCoreStream *stream = (PaMacCoreStream*)s; - VVDBUG(("GetStreamWriteAvailable()\n")); - - /* suppress unused variable warnings */ - (void) stream; - - /* IMPLEMENT ME, see portaudio.h for required behavior*/ - - return 0; -} |