[156f39]: portaudio / src / hostapi / coreaudio / pa_mac_core.c Maximize Restore History

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pa_mac_core.c    2791 lines (2471 with data), 116.0 kB

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/*
* 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 <libkern/OSAtomic.h>
#include "pa_mac_core.h"
#include "pa_mac_core_utilities.h"
#include "pa_mac_core_blocking.h"
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
/* This is a reasonable size for a small buffer based on experience. */
#define PA_MAC_SMALL_BUFFER_SIZE (64)
/* 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);
if( err != paNoError )
return NULL;
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;
}
PaError PaMacCore_GetBufferSizeRange( PaDeviceIndex device,
long *minBufferSizeFrames, long *maxBufferSizeFrames )
{
PaError result;
PaUtilHostApiRepresentation *hostApi;
result = PaUtil_GetHostApiRepresentation( &hostApi, paCoreAudio );
if( result == paNoError )
{
PaDeviceIndex hostApiDeviceIndex;
result = PaUtil_DeviceIndexToHostApiDeviceIndex( &hostApiDeviceIndex, device, hostApi );
if( result == paNoError )
{
PaMacAUHAL *macCoreHostApi = (PaMacAUHAL*)hostApi;
AudioDeviceID macCoreDeviceId = macCoreHostApi->devIds[hostApiDeviceIndex];
AudioValueRange audioRange;
UInt32 propSize = sizeof( audioRange );
// return the size range for the output scope unless we only have inputs
Boolean isInput = 0;
if( macCoreHostApi->inheritedHostApiRep.deviceInfos[hostApiDeviceIndex]->maxOutputChannels == 0 )
isInput = 1;
result = WARNING(AudioDeviceGetProperty( macCoreDeviceId, 0, isInput, kAudioDevicePropertyBufferFrameSizeRange, &propSize, &audioRange ) );
*minBufferSizeFrames = audioRange.mMinimum;
*maxBufferSizeFrames = audioRange.mMaximum;
}
}
return result;
}
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 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 PaMacCoreStream *stream,
const PaStreamParameters *inStreamParams,
const PaStreamParameters *outStreamParams,
const UInt32 requestedFramesPerBuffer,
UInt32 *actualInputFramesPerBuffer,
UInt32 *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 )
/*
* Callback called when starting or stopping a stream.
*/
static void startStopCallback(
void * inRefCon,
AudioUnit ci,
AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement )
{
PaMacCoreStream *stream = (PaMacCoreStream *) inRefCon;
UInt32 isRunning;
UInt32 size = sizeof( isRunning );
OSStatus err;
err = AudioUnitGetProperty( ci, kAudioOutputUnitProperty_IsRunning, inScope, inElement, &isRunning, &size );
assert( !err );
if( err )
isRunning = false; //it's very unclear what to do in case of error here. There's no real way to notify the user, and crashing seems unreasonable.
if( isRunning )
return; //We are only interested in when we are stopping
// -- if we are using 2 I/O units, we only need one notification!
if( stream->inputUnit && stream->outputUnit && stream->inputUnit != stream->outputUnit && ci == stream->inputUnit )
return;
PaStreamFinishedCallback *sfc = stream->streamRepresentation.streamFinishedCallback;
if( stream->state == STOPPING )
stream->state = STOPPED ;
if( sfc )
sfc( stream->streamRepresentation.userData );
}
/*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;
}
/* =================================================================================================== */
/**
* @internal
* @brief Clip the desired size against the allowed IO buffer size range for the device.
*/
static PaError ClipToDeviceBufferSize( AudioDeviceID macCoreDeviceId,
int isInput, UInt32 desiredSize, UInt32 *allowedSize )
{
UInt32 resultSize = desiredSize;
AudioValueRange audioRange;
UInt32 propSize = sizeof( audioRange );
PaError err = WARNING(AudioDeviceGetProperty( macCoreDeviceId, 0, isInput, kAudioDevicePropertyBufferFrameSizeRange, &propSize, &audioRange ) );
resultSize = MAX( resultSize, audioRange.mMinimum );
resultSize = MIN( resultSize, audioRange.mMaximum );
*allowedSize = resultSize;
return err;
}
/* =================================================================================================== */
#if 0
static void DumpDeviceProperties( AudioDeviceID macCoreDeviceId,
int isInput )
{
PaError err;
int i;
UInt32 propSize;
UInt32 deviceLatency;
UInt32 streamLatency;
UInt32 bufferFrames;
UInt32 safetyOffset;
AudioStreamID streamIDs[128];
printf("\n======= latency query : macCoreDeviceId = %d, isInput %d =======\n", (int)macCoreDeviceId, isInput );
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyBufferFrameSize, &propSize, &bufferFrames));
printf("kAudioDevicePropertyBufferFrameSize: err = %d, propSize = %d, value = %d\n", err, propSize, bufferFrames );
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertySafetyOffset, &propSize, &safetyOffset));
printf("kAudioDevicePropertySafetyOffset: err = %d, propSize = %d, value = %d\n", err, propSize, safetyOffset );
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &deviceLatency));
printf("kAudioDevicePropertyLatency: err = %d, propSize = %d, value = %d\n", err, propSize, deviceLatency );
AudioValueRange audioRange;
propSize = sizeof( audioRange );
err = WARNING(AudioDeviceGetProperty( macCoreDeviceId, 0, isInput, kAudioDevicePropertyBufferFrameSizeRange, &propSize, &audioRange ) );
printf("kAudioDevicePropertyBufferFrameSizeRange: err = %d, propSize = %u, minimum = %g\n", err, propSize, audioRange.mMinimum);
printf("kAudioDevicePropertyBufferFrameSizeRange: err = %d, propSize = %u, maximum = %g\n", err, propSize, audioRange.mMaximum );
/* Get the streams from the device and query their latency. */
propSize = sizeof(streamIDs);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreams, &propSize, &streamIDs[0]));
int numStreams = propSize / sizeof(AudioStreamID);
for( i=0; i<numStreams; i++ )
{
printf("Stream #%d = %d---------------------- \n", i, streamIDs[i] );
propSize = sizeof(UInt32);
err = WARNING(AudioStreamGetProperty(streamIDs[i], 0, kAudioStreamPropertyLatency, &propSize, &streamLatency));
printf(" kAudioStreamPropertyLatency: err = %d, propSize = %d, value = %d\n", err, propSize, streamLatency );
}
}
#endif
/* =================================================================================================== */
/**
* @internal
* Calculate the fixed latency from the system and the device.
* Sum of kAudioStreamPropertyLatency +
* kAudioDevicePropertySafetyOffset +
* kAudioDevicePropertyLatency
*
* Some useful info from Jeff Moore on latency.
* http://osdir.com/ml/coreaudio-api/2010-01/msg00046.html
* http://osdir.com/ml/coreaudio-api/2009-07/msg00140.html
*/
static PaError CalculateFixedDeviceLatency( AudioDeviceID macCoreDeviceId, int isInput, UInt32 *fixedLatencyPtr )
{
PaError err;
UInt32 propSize;
UInt32 deviceLatency;
UInt32 streamLatency;
UInt32 safetyOffset;
AudioStreamID streamIDs[1];
// To get stream latency we have to get a streamID from the device.
// We are only going to look at the first stream so only fetch one stream.
propSize = sizeof(streamIDs);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreams, &propSize, &streamIDs[0]));
if( err != paNoError ) goto error;
if( propSize == sizeof(AudioStreamID) )
{
propSize = sizeof(UInt32);
err = WARNING(AudioStreamGetProperty(streamIDs[0], 0, kAudioStreamPropertyLatency, &propSize, &streamLatency));
}
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertySafetyOffset, &propSize, &safetyOffset));
if( err != paNoError ) goto error;
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &deviceLatency));
if( err != paNoError ) goto error;
*fixedLatencyPtr = deviceLatency + streamLatency + safetyOffset;
return err;
error:
return err;
}
/* =================================================================================================== */
static PaError CalculateDefaultDeviceLatencies( AudioDeviceID macCoreDeviceId,
int isInput, UInt32 *lowLatencyFramesPtr,
UInt32 *highLatencyFramesPtr )
{
UInt32 propSize;
UInt32 bufferFrames = 0;
UInt32 fixedLatency = 0;
UInt32 clippedMinBufferSize = 0;
//DumpDeviceProperties( macCoreDeviceId, isInput );
PaError err = CalculateFixedDeviceLatency( macCoreDeviceId, isInput, &fixedLatency );
if( err != paNoError ) goto error;
// For low latency use a small fixed size buffer clipped to the device range.
err = ClipToDeviceBufferSize( macCoreDeviceId, isInput, PA_MAC_SMALL_BUFFER_SIZE, &clippedMinBufferSize );
if( err != paNoError ) goto error;
// For high latency use the default device buffer size.
propSize = sizeof(UInt32);
err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyBufferFrameSize, &propSize, &bufferFrames));
if( err != paNoError ) goto error;
*lowLatencyFramesPtr = fixedLatency + clippedMinBufferSize;
*highLatencyFramesPtr = fixedLatency + bufferFrames;
return err;
error:
return err;
}
/* =================================================================================================== */
static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi,
PaDeviceInfo *deviceInfo,
AudioDeviceID macCoreDeviceId,
int isInput)
{
UInt32 propSize;
PaError err = paNoError;
UInt32 i;
int numChannels = 0;
AudioBufferList *buflist = NULL;
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 are 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;
UInt32 lowLatencyFrames = 0;
UInt32 highLatencyFrames = 0;
err = CalculateDefaultDeviceLatencies( macCoreDeviceId, isInput, &lowLatencyFrames, &highLatencyFrames );
if( err == 0 )
{
double lowLatencySeconds = lowLatencyFrames / deviceInfo->defaultSampleRate;
double highLatencySeconds = highLatencyFrames / deviceInfo->defaultSampleRate;
if (isInput)
{
deviceInfo->defaultLowInputLatency = lowLatencySeconds;
deviceInfo->defaultHighInputLatency = highLatencySeconds;
}
else
{
deviceInfo->defaultLowOutputLatency = lowLatencySeconds;
deviceInfo->defaultHighOutputLatency = highLatencySeconds;
}
}
}
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 = NULL;
PaDeviceInfo *deviceInfoArray;
int unixErr;
VVDBUG(("PaMacCore_Initialize(): hostApiIndex=%d\n", hostApiIndex));
SInt32 major;
SInt32 minor;
Gestalt(gestaltSystemVersionMajor, &major);
Gestalt(gestaltSystemVersionMinor, &minor);
// Starting with 10.6 systems, the HAL notification thread is created internally
if (major == 10 && minor >= 6) {
CFRunLoopRef theRunLoop = NULL;
AudioObjectPropertyAddress theAddress = { kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
OSStatus osErr = AudioObjectSetPropertyData (kAudioObjectSystemObject, &theAddress, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
if (osErr != noErr) {
goto error;
}
}
unixErr = initializeXRunListenerList();
if( 0 != unixErr ) {
return UNIX_ERR(unixErr);
}
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 )
{
int unixErr;
PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi;
VVDBUG(("Terminate()\n"));
unixErr = destroyXRunListenerList();
if( 0 != unixErr )
UNIX_ERR(unixErr);
/*
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 void InitializeDeviceProperties( PaMacCoreDeviceProperties *deviceProperties )
{
memset( deviceProperties, 0, sizeof(PaMacCoreDeviceProperties) );
deviceProperties->sampleRate = 1.0; // Better than random. Overwritten by actual values later on.
deviceProperties->samplePeriod = 1.0 / deviceProperties->sampleRate;
}
static Float64 CalculateSoftwareLatencyFromProperties( PaMacCoreStream *stream, PaMacCoreDeviceProperties *deviceProperties )
{
UInt32 latencyFrames = deviceProperties->bufferFrameSize + deviceProperties->deviceLatency + deviceProperties->safetyOffset;
return latencyFrames * deviceProperties->samplePeriod; // same as dividing by sampleRate but faster
}
static Float64 CalculateHardwareLatencyFromProperties( PaMacCoreStream *stream, PaMacCoreDeviceProperties *deviceProperties )
{
return deviceProperties->deviceLatency * deviceProperties->samplePeriod; // same as dividing by sampleRate but faster
}
/* Calculate values used to convert Apple timestamps into PA timestamps
* from the device properties. The final results of this calculation
* will be used in the audio callback function.
*/
static void UpdateTimeStampOffsets( PaMacCoreStream *stream )
{
Float64 inputSoftwareLatency = 0.0;
Float64 inputHardwareLatency = 0.0;
Float64 outputSoftwareLatency = 0.0;
Float64 outputHardwareLatency = 0.0;
if( stream->inputUnit != NULL )
{
inputSoftwareLatency = CalculateSoftwareLatencyFromProperties( stream, &stream->inputProperties );
inputHardwareLatency = CalculateHardwareLatencyFromProperties( stream, &stream->inputProperties );
}
if( stream->outputUnit != NULL )
{
outputSoftwareLatency = CalculateSoftwareLatencyFromProperties( stream, &stream->outputProperties );
outputHardwareLatency = CalculateHardwareLatencyFromProperties( stream, &stream->outputProperties );
}
/* We only need a mutex around setting these variables as a group. */
pthread_mutex_lock( &stream->timingInformationMutex );
stream->timestampOffsetCombined = inputSoftwareLatency + outputSoftwareLatency;
stream->timestampOffsetInputDevice = inputHardwareLatency;
stream->timestampOffsetOutputDevice = outputHardwareLatency;
pthread_mutex_unlock( &stream->timingInformationMutex );
}
/* ================================================================================= */
/* can be used to update from nominal or actual sample rate */
static OSStatus UpdateSampleRateFromDeviceProperty( PaMacCoreStream *stream, AudioDeviceID deviceID, Boolean isInput, AudioDevicePropertyID sampleRatePropertyID )
{
PaMacCoreDeviceProperties * deviceProperties = isInput ? &stream->inputProperties : &stream->outputProperties;
Float64 sampleRate = 0.0;
UInt32 propSize = sizeof(Float64);
OSStatus osErr = AudioDeviceGetProperty( deviceID, 0, isInput, sampleRatePropertyID, &propSize, &sampleRate);
if( (osErr == noErr) && (sampleRate > 1000.0) ) /* avoid divide by zero if there's an error */
{
deviceProperties->sampleRate = sampleRate;
deviceProperties->samplePeriod = 1.0 / sampleRate;
}
return osErr;
}
static OSStatus AudioDevicePropertyActualSampleRateListenerProc( AudioDeviceID inDevice, UInt32 inChannel, Boolean isInput, AudioDevicePropertyID inPropertyID, void *inClientData )
{
PaMacCoreStream *stream = (PaMacCoreStream*)inClientData;
// Make sure the callback is operating on a stream that is still valid!
assert( stream->streamRepresentation.magic == PA_STREAM_MAGIC );
OSStatus osErr = UpdateSampleRateFromDeviceProperty( stream, inDevice, isInput, kAudioDevicePropertyActualSampleRate );
if( osErr == noErr )
{
UpdateTimeStampOffsets( stream );
}
return osErr;
}
/* ================================================================================= */
static OSStatus QueryUInt32DeviceProperty( AudioDeviceID deviceID, Boolean isInput, AudioDevicePropertyID propertyID, UInt32 *outValue )
{
UInt32 propertyValue = 0;
UInt32 propertySize = sizeof(UInt32);
OSStatus osErr = AudioDeviceGetProperty( deviceID, 0, isInput, propertyID, &propertySize, &propertyValue);
if( osErr == noErr )
{
*outValue = propertyValue;
}
return osErr;
}
static OSStatus AudioDevicePropertyGenericListenerProc( AudioDeviceID inDevice, UInt32 inChannel, Boolean isInput, AudioDevicePropertyID inPropertyID, void *inClientData )
{
OSStatus osErr = noErr;
PaMacCoreStream *stream = (PaMacCoreStream*)inClientData;
// Make sure the callback is operating on a stream that is still valid!
assert( stream->streamRepresentation.magic == PA_STREAM_MAGIC );
PaMacCoreDeviceProperties *deviceProperties = isInput ? &stream->inputProperties : &stream->outputProperties;
UInt32 *valuePtr = NULL;
switch( inPropertyID )
{
case kAudioDevicePropertySafetyOffset:
valuePtr = &deviceProperties->safetyOffset;
break;
case kAudioDevicePropertyLatency:
valuePtr = &deviceProperties->deviceLatency;
break;
case kAudioDevicePropertyBufferFrameSize:
valuePtr = &deviceProperties->bufferFrameSize;
break;
}
if( valuePtr != NULL )
{
osErr = QueryUInt32DeviceProperty( inDevice, isInput, inPropertyID, valuePtr );
if( osErr == noErr )
{
UpdateTimeStampOffsets( stream );
}
}
return osErr;
}
/* ================================================================================= */
/*
* Setup listeners in case device properties change during the run. */
static OSStatus SetupDevicePropertyListeners( PaMacCoreStream *stream, AudioDeviceID deviceID, Boolean isInput )
{
OSStatus osErr = noErr;
PaMacCoreDeviceProperties *deviceProperties = isInput ? &stream->inputProperties : &stream->outputProperties;
if( (osErr = QueryUInt32DeviceProperty( deviceID, isInput,
kAudioDevicePropertyLatency, &deviceProperties->deviceLatency )) != noErr ) return osErr;
if( (osErr = QueryUInt32DeviceProperty( deviceID, isInput,
kAudioDevicePropertyBufferFrameSize, &deviceProperties->bufferFrameSize )) != noErr ) return osErr;
if( (osErr = QueryUInt32DeviceProperty( deviceID, isInput,
kAudioDevicePropertySafetyOffset, &deviceProperties->safetyOffset )) != noErr ) return osErr;
AudioDeviceAddPropertyListener( deviceID, 0, isInput, kAudioDevicePropertyActualSampleRate,
AudioDevicePropertyActualSampleRateListenerProc, stream );
AudioDeviceAddPropertyListener( deviceID, 0, isInput, kAudioStreamPropertyLatency,
AudioDevicePropertyGenericListenerProc, stream );
AudioDeviceAddPropertyListener( deviceID, 0, isInput, kAudioDevicePropertyBufferFrameSize,
AudioDevicePropertyGenericListenerProc, stream );
AudioDeviceAddPropertyListener( deviceID, 0, isInput, kAudioDevicePropertySafetyOffset,
AudioDevicePropertyGenericListenerProc, stream );
return osErr;
}
static void CleanupDevicePropertyListeners( PaMacCoreStream *stream, AudioDeviceID deviceID, Boolean isInput )
{
AudioDeviceRemovePropertyListener( deviceID, 0, isInput, kAudioDevicePropertyActualSampleRate,
AudioDevicePropertyActualSampleRateListenerProc );
AudioDeviceRemovePropertyListener( deviceID, 0, isInput, kAudioDevicePropertyLatency,
AudioDevicePropertyGenericListenerProc );
AudioDeviceRemovePropertyListener( deviceID, 0, isInput, kAudioDevicePropertyBufferFrameSize,
AudioDevicePropertyGenericListenerProc );
AudioDeviceRemovePropertyListener( deviceID, 0, isInput, kAudioDevicePropertySafetyOffset,
AudioDevicePropertyGenericListenerProc );
}
/* ================================================================================= */
static PaError OpenAndSetupOneAudioUnit(
const PaMacCoreStream *stream,
const PaStreamParameters *inStreamParams,
const PaStreamParameters *outStreamParams,
const UInt32 requestedFramesPerBuffer,
UInt32 *actualInputFramesPerBuffer,
UInt32 *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;
OSStatus 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 && outStreamParams != inStreamParams )
{
*audioDevice = auhalHostApi->devIds[outStreamParams->device] ;
ERR_WRAP( AudioUnitSetProperty( *audioUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global,
OUTPUT_ELEMENT,
audioDevice,
sizeof(AudioDeviceID) ) );
}
/* -- add listener for dropouts -- */
result = AudioDeviceAddPropertyListener( *audioDevice,
0,
outStreamParams ? false : true,
kAudioDeviceProcessorOverload,
xrunCallback,
addToXRunListenerList( (void *)stream ) ) ;
if( result == kAudioHardwareIllegalOperationError ) {
// -- already registered, we're good
} else {
// -- not already registered, just check for errors
ERR_WRAP( result );
}
/* -- listen for stream start and stop -- */
ERR_WRAP( AudioUnitAddPropertyListener( *audioUnit,
kAudioOutputUnitProperty_IsRunning,
startStopCallback,
(void *)stream ) );
/* -- 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(*actualOutputFramesPerBuffer) ) );
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(*actualInputFramesPerBuffer) ) );
/* 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;
}
/* =================================================================================================== */
static UInt32 CalculateOptimalBufferSize( PaMacAUHAL *auhalHostApi,
const PaStreamParameters *inputParameters,
const PaStreamParameters *outputParameters,
UInt32 fixedInputLatency,
UInt32 fixedOutputLatency,
double sampleRate,
UInt32 requestedFramesPerBuffer )
{
UInt32 resultBufferSizeFrames = 0;
// Use maximum of suggested input and output latencies.
if( inputParameters )
{
UInt32 suggestedLatencyFrames = inputParameters->suggestedLatency * sampleRate;
// Calculate a buffer size assuming we are double buffered.
SInt32 variableLatencyFrames = suggestedLatencyFrames - fixedInputLatency;
// Prevent negative latency.
variableLatencyFrames = MAX( variableLatencyFrames, 0 );
resultBufferSizeFrames = MAX( resultBufferSizeFrames, (UInt32) variableLatencyFrames );
}
if( outputParameters )
{
UInt32 suggestedLatencyFrames = outputParameters->suggestedLatency * sampleRate;
SInt32 variableLatencyFrames = suggestedLatencyFrames - fixedOutputLatency;
variableLatencyFrames = MAX( variableLatencyFrames, 0 );
resultBufferSizeFrames = MAX( resultBufferSizeFrames, (UInt32) variableLatencyFrames );
}
// can't have zero frames. code to round up to next user buffer requires non-zero
resultBufferSizeFrames = MAX( resultBufferSizeFrames, 1 );
if( requestedFramesPerBuffer != paFramesPerBufferUnspecified )
{
// make host buffer the next highest integer multiple of user frames per buffer
UInt32 n = (resultBufferSizeFrames + requestedFramesPerBuffer - 1) / requestedFramesPerBuffer;
resultBufferSizeFrames = n * requestedFramesPerBuffer;
// FIXME: really we should be searching for a multiple of requestedFramesPerBuffer
// that is >= suggested latency and also fits within device buffer min/max
}else{
VDBUG( ("Block Size unspecified. Based on Latency, the user wants a Block Size near: %ld.\n",
resultBufferSizeFrames ) );
}
// Clip to the capabilities of the device.
if( inputParameters )
{
ClipToDeviceBufferSize( auhalHostApi->devIds[inputParameters->device],
true, // In the old code isInput was false!
resultBufferSizeFrames, &resultBufferSizeFrames );
}
if( outputParameters )
{
ClipToDeviceBufferSize( auhalHostApi->devIds[outputParameters->device],
false, resultBufferSizeFrames, &resultBufferSizeFrames );
}
VDBUG(("After querying hardware, setting block size to %ld.\n", resultBufferSizeFrames));
return resultBufferSizeFrames;
}
/* =================================================================================================== */
/* 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 requestedFramesPerBuffer,
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;
UInt32 fixedInputLatency = 0;
UInt32 fixedOutputLatency = 0;
// Accumulate contributions to latency in these variables.
UInt32 inputLatencyFrames = 0;
UInt32 outputLatencyFrames = 0;
UInt32 suggestedLatencyFramesPerBuffer = requestedFramesPerBuffer;
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,
requestedFramesPerBuffer ));
VDBUG( ("Opening Stream.\n") );
/* These first few bits of code are from paSkeleton with few modifications. */
if( inputParameters )
{
inputChannelCount = inputParameters->channelCount;
inputSampleFormat = inputParameters->sampleFormat;
/* @todo Blocking read/write on Mac is not yet supported. */
if( !streamCallback && inputSampleFormat & paNonInterleaved )
{
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;
/* Host supports interleaved float32 */
hostInputSampleFormat = paFloat32;
}
else
{
inputChannelCount = 0;
inputSampleFormat = hostInputSampleFormat = paFloat32; /* Surpress 'uninitialised var' warnings. */
}
if( outputParameters )
{
outputChannelCount = outputParameters->channelCount;
outputSampleFormat = outputParameters->sampleFormat;
/* @todo Blocking read/write on Mac is not yet supported. */
if( !streamCallback && outputSampleFormat & paNonInterleaved )
{
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 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.
*/
bzero( stream, sizeof( PaMacCoreStream ) );
/*
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);
*/
/* 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 );
if( inputParameters )
{
CalculateFixedDeviceLatency( auhalHostApi->devIds[inputParameters->device], true, &fixedInputLatency );
inputLatencyFrames += fixedInputLatency;
}
if( outputParameters )
{
CalculateFixedDeviceLatency( auhalHostApi->devIds[outputParameters->device], false, &fixedOutputLatency );
outputLatencyFrames += fixedOutputLatency;
}
suggestedLatencyFramesPerBuffer = CalculateOptimalBufferSize( auhalHostApi, inputParameters, outputParameters,
fixedInputLatency, fixedOutputLatency,
sampleRate, requestedFramesPerBuffer );
if( requestedFramesPerBuffer == paFramesPerBufferUnspecified )
{
requestedFramesPerBuffer = suggestedLatencyFramesPerBuffer;
}
/* -- Now we actually open and setup streams. -- */
if( inputParameters && outputParameters && outputParameters->device == inputParameters->device )
{ /* full duplex. One device. */
UInt32 inputFramesPerBuffer = (UInt32) stream->inputFramesPerBuffer;
UInt32 outputFramesPerBuffer = (UInt32) stream->outputFramesPerBuffer;
result = OpenAndSetupOneAudioUnit( stream,
inputParameters,
outputParameters,
suggestedLatencyFramesPerBuffer,
&inputFramesPerBuffer,
&outputFramesPerBuffer,
auhalHostApi,
&(stream->inputUnit),
&(stream->inputSRConverter),
&(stream->inputDevice),
sampleRate,
stream );
stream->inputFramesPerBuffer = inputFramesPerBuffer;
stream->outputFramesPerBuffer = outputFramesPerBuffer;
stream->outputUnit = stream->inputUnit;
stream->outputDevice = stream->inputDevice;
if( result != paNoError )
goto error;
}
else
{ /* full duplex, different devices OR simplex */
UInt32 outputFramesPerBuffer = (UInt32) stream->outputFramesPerBuffer;
UInt32 inputFramesPerBuffer = (UInt32) stream->inputFramesPerBuffer;
result = OpenAndSetupOneAudioUnit( stream,
NULL,
outputParameters,
suggestedLatencyFramesPerBuffer,
NULL,
&outputFramesPerBuffer,
auhalHostApi,
&(stream->outputUnit),
NULL,
&(stream->outputDevice),
sampleRate,
stream );
if( result != paNoError )
goto error;
result = OpenAndSetupOneAudioUnit( stream,
inputParameters,
NULL,
suggestedLatencyFramesPerBuffer,
&inputFramesPerBuffer,
NULL,
auhalHostApi,
&(stream->inputUnit),
&(stream->inputSRConverter),
&(stream->inputDevice),
sampleRate,
stream );
if( result != paNoError )
goto error;
stream->inputFramesPerBuffer = inputFramesPerBuffer;
stream->outputFramesPerBuffer = outputFramesPerBuffer;
}
inputLatencyFrames += stream->inputFramesPerBuffer;
outputLatencyFrames += stream->outputFramesPerBuffer;
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*inputParameters->channelCount );
if( !data )
{
result = paInsufficientMemory;
goto error;
}
/* now we can initialize the ring buffer */
PaUtil_InitializeRingBuffer( &stream->inputRingBuffer, szfl*inputParameters->channelCount, ringSize, 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 / RING_BUFFER_ADVANCE_DENOMINATOR );
// Just adds to input latency between input device and PA full duplex callback.
inputLatencyFrames += ringSize;
}
}
/* -- 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;
inputLatencyFrames += ringSize;
outputLatencyFrames += ringSize;
}
/* -- 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,
requestedFramesPerBuffer,
/* 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;
// Calculate actual latency from the sum of individual latencies.
if( inputParameters )
{
inputLatencyFrames += PaUtil_GetBufferProcessorInputLatencyFrames(&stream->bufferProcessor);
stream->streamRepresentation.streamInfo.inputLatency = inputLatencyFrames / sampleRate;
}
else
{
stream->streamRepresentation.streamInfo.inputLatency = 0.0;
}
if( outputParameters )
{
outputLatencyFrames += PaUtil_GetBufferProcessorOutputLatencyFrames(&stream->bufferProcessor);
stream->streamRepresentation.streamInfo.outputLatency = outputLatencyFrames / sampleRate;
}
else
{
stream->streamRepresentation.streamInfo.outputLatency = 0.0;
}
stream->streamRepresentation.streamInfo.sampleRate = sampleRate;
stream->sampleRate = sampleRate;
stream->userInChan = inputChannelCount;
stream->userOutChan = outputChannelCount;
// Setup property listeners for timestamp and latency calculations.
pthread_mutex_init( &stream->timingInformationMutex, NULL );
stream->timingInformationMutexIsInitialized = 1;
InitializeDeviceProperties( &stream->inputProperties ); // zeros the struct. doesn't actually init it to useful values
InitializeDeviceProperties( &stream->outputProperties ); // zeros the struct. doesn't actually init it to useful values
if( stream->outputUnit )
{
Boolean isInput = FALSE;
// Start with the current values for the device properties.
// Init with nominal sample rate. Use actual sample rate where available
result = ERR( UpdateSampleRateFromDeviceProperty(
stream, stream->outputDevice, isInput, kAudioDevicePropertyNominalSampleRate ) );
if( result )
goto error; /* fail if we can't even get a nominal device sample rate */
UpdateSampleRateFromDeviceProperty( stream, stream->outputDevice, isInput, kAudioDevicePropertyActualSampleRate );
SetupDevicePropertyListeners( stream, stream->outputDevice, isInput );
}
if( stream->inputUnit )
{
Boolean isInput = TRUE;
// as above
result = ERR( UpdateSampleRateFromDeviceProperty(
stream, stream->inputDevice, isInput, kAudioDevicePropertyNominalSampleRate ) );
if( result )
goto error;
UpdateSampleRateFromDeviceProperty( stream, stream->inputDevice, isInput, kAudioDevicePropertyActualSampleRate );
SetupDevicePropertyListeners( stream, stream->inputDevice, isInput );
}
UpdateTimeStampOffsets( stream );
// Setup timestamp copies to be used by audio callback.
stream->timestampOffsetCombined_ioProcCopy = stream->timestampOffsetCombined;
stream->timestampOffsetInputDevice_ioProcCopy = stream->timestampOffsetInputDevice;
stream->timestampOffsetOutputDevice_ioProcCopy = stream->timestampOffsetOutputDevice;
stream->state = STOPPED;
stream->xrunFlags = 0;
*s = (PaStream*)stream;
return result;
error:
CloseStream( stream );
return result;
}
#define HOST_TIME_TO_PA_TIME( x ) ( AudioConvertHostTimeToNanos( (x) ) * 1.0E-09) /* convert to nanoseconds and then to seconds */
PaTime GetStreamTime( PaStream *s )
{
return HOST_TIME_TO_PA_TIME( AudioGetCurrentHostTime() );
}
#define RING_BUFFER_EMPTY (1000)
static OSStatus ringBufferIOProc( AudioConverterRef inAudioConverter,
UInt32*ioDataSize,
void** outData,
void*inUserData )
{
void *dummyData;
ring_buffer_size_t dummySize;
PaUtilRingBuffer *rb = (PaUtilRingBuffer *) inUserData;
VVDBUG(("ringBufferIOProc()\n"));
if( PaUtil_GetRingBufferReadAvailable( rb ) == 0 ) {
*outData = NULL;
*ioDataSize = 0;
return RING_BUFFER_EMPTY;
}
assert(sizeof(UInt32) == sizeof(ring_buffer_size_t));
assert( ( (*ioDataSize) / rb->elementSizeBytes ) * rb->elementSizeBytes == (*ioDataSize) ) ;
(*ioDataSize) /= rb->elementSizeBytes ;
PaUtil_GetRingBufferReadRegions( rb, *ioDataSize,
outData, (ring_buffer_size_t *)ioDataSize,
&dummyData, &dummySize );
assert( *ioDataSize );
PaUtil_AdvanceRingBufferReadIndex( rb, *ioDataSize );
(*ioDataSize) *= rb->elementSizeBytes ;
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 ;
double hostTimeStampInPaTime = HOST_TIME_TO_PA_TIME(inTimeStamp->mHostTime);
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 );
}
----------------------------------------------------------------- */
/* compute PaStreamCallbackTimeInfo */
if( pthread_mutex_trylock( &stream->timingInformationMutex ) == 0 ){
/* snapshot the ioproc copy of timing information */
stream->timestampOffsetCombined_ioProcCopy = stream->timestampOffsetCombined;
stream->timestampOffsetInputDevice_ioProcCopy = stream->timestampOffsetInputDevice;
stream->timestampOffsetOutputDevice_ioProcCopy = stream->timestampOffsetOutputDevice;
pthread_mutex_unlock( &stream->timingInformationMutex );
}
/* For timeInfo.currentTime we could calculate current time backwards from the HAL audio
output time to give a more accurate impression of the current timeslice but it doesn't
seem worth it at the moment since other PA host APIs don't do any better.
*/
timeInfo.currentTime = HOST_TIME_TO_PA_TIME( AudioGetCurrentHostTime() );
/*
For an input HAL AU, inTimeStamp is the time the samples are received from the hardware,
for an output HAL AU inTimeStamp is the time the samples are sent to the hardware.
PA expresses timestamps in terms of when the samples enter the ADC or leave the DAC
so we add or subtract kAudioDevicePropertyLatency below.
*/
/* FIXME: not sure what to do below if the host timestamps aren't valid (kAudioTimeStampHostTimeValid isn't set)
Could ask on CA mailing list if it is possible for it not to be set. If so, could probably grab a now timestamp
at the top and compute from there (modulo scheduling jitter) or ask on mailing list for other options. */
if( isRender )
{
if( stream->inputUnit ) /* full duplex */
{
if( stream->inputUnit == stream->outputUnit ) /* full duplex AUHAL IOProc */
{
// Ross and Phil agreed that the following calculation is correct based on an email from Jeff Moore:
// http://osdir.com/ml/coreaudio-api/2009-07/msg00140.html
// Basically the difference between the Apple output timestamp and the PA timestamp is kAudioDevicePropertyLatency.
timeInfo.inputBufferAdcTime = hostTimeStampInPaTime -
(stream->timestampOffsetCombined_ioProcCopy + stream->timestampOffsetInputDevice_ioProcCopy);
timeInfo.outputBufferDacTime = hostTimeStampInPaTime + stream->timestampOffsetOutputDevice_ioProcCopy;
}
else /* full duplex with ring-buffer from a separate input AUHAL ioproc */
{
/* FIXME: take the ring buffer latency into account */
timeInfo.inputBufferAdcTime = hostTimeStampInPaTime -
(stream->timestampOffsetCombined_ioProcCopy + stream->timestampOffsetInputDevice_ioProcCopy);
timeInfo.outputBufferDacTime = hostTimeStampInPaTime + stream->timestampOffsetOutputDevice_ioProcCopy;
}
}
else /* output only */
{
timeInfo.inputBufferAdcTime = 0;
timeInfo.outputBufferDacTime = hostTimeStampInPaTime + stream->timestampOffsetOutputDevice_ioProcCopy;
}
}
else /* input only */
{
timeInfo.inputBufferAdcTime = hostTimeStampInPaTime - stream->timestampOffsetInputDevice_ioProcCopy;
timeInfo.outputBufferDacTime = 0;
}
//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.
*
*/
OSStatus err = 0;
unsigned long frames;
long bytesPerFrame = sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
/* -- start processing -- */
PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
&timeInfo,
stream->xrunFlags );
stream->xrunFlags = 0; //FIXME: this flag also gets set outside by a callback, which calls the xrunCallback function. It should be in the same thread as the main audio callback, but the apple docs just use the word "usually" so it may be possible to loose an xrun notification, if that callback happens here.
/* -- compute frames. do some checks -- */
assert( ioData->mNumberBuffers == 1 );
assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
frames = ioData->mBuffers[0].mDataByteSize / bytesPerFrame;
/* -- 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. There's nothing in the PA API to
* do about failures in the callback system. */
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;
long bytesPerFrame = sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
/* 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 / bytesPerFrame;
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;
long bytesPerFrame = flsz * inChan;
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 );
/* The ring buffer can underflow normally when the stream is stopping.
* So only report an error if the stream is active. */
if( stream->state == ACTIVE )
{
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;
ring_buffer_size_t size1, size2;
ring_buffer_size_t framesReadable = PaUtil_GetRingBufferReadRegions( &stream->inputRingBuffer,
frames,
&data1, &size1,
&data2, &size2 );
if( size1 == 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( framesReadable < frames ) {
long sizeBytes1 = size1 * bytesPerFrame;
long sizeBytes2 = size2 * bytesPerFrame;
/*we underflowed. take what data we can, zero the rest.*/
unsigned char data[ frames * bytesPerFrame ];
if( size1 > 0 )
{
memcpy( data, data1, sizeBytes1 );
}
if( size2 > 0 )
{
memcpy( data+sizeBytes1, data2, sizeBytes2 );
}
bzero( data+sizeBytes1+sizeBytes2, (frames*bytesPerFrame) - sizeBytes1 - sizeBytes2 );
PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
0,
data,
inChan );
framesProcessed =
PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
&callbackResult );
PaUtil_AdvanceRingBufferReadIndex( &stream->inputRingBuffer,
framesReadable );
/* flag underflow */
stream->xrunFlags |= paInputUnderflow;
} else {
/*we got all the data, but split between buffers*/
PaUtil_SetInputFrameCount( &(stream->bufferProcessor), size1 );
PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
0,
data1,
inChan );
PaUtil_Set2ndInputFrameCount( &(stream->bufferProcessor), size2 );
PaUtil_Set2ndInterleavedInputChannels( &(stream->bufferProcessor),
0,
data2,
inChan );
framesProcessed =
PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
&callbackResult );
PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, framesReadable );
}
}
} 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.
*/
OSStatus 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. */
ring_buffer_size_t framesWritten = PaUtil_WriteRingBuffer( &stream->inputRingBuffer,
stream->inputAudioBufferList.mBuffers[0].mData,
inNumberFrames );
if( framesWritten != inNumberFrames )
{
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 )
{
Boolean isInput = FALSE;
CleanupDevicePropertyListeners( stream, stream->outputDevice, isInput );
}
if( stream->inputUnit )
{
Boolean isInput = TRUE;
CleanupDevicePropertyListeners( stream, stream->inputDevice, isInput );
}
if( stream->outputUnit ) {
int count = removeFromXRunListenerList( stream );
if( count == 0 )
AudioDeviceRemovePropertyListener( stream->outputDevice,
0,
false,
kAudioDeviceProcessorOverload,
xrunCallback );
}
if( stream->inputUnit && stream->outputUnit != stream->inputUnit ) {
int count = removeFromXRunListenerList( stream );
if( count == 0 )
AudioDeviceRemovePropertyListener( stream->inputDevice,
0,
true,
kAudioDeviceProcessorOverload,
xrunCallback );
}
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 );
if( stream->timingInformationMutexIsInitialized )
pthread_mutex_destroy( &stream->timingInformationMutex );
PaUtil_TerminateStreamRepresentation( &stream->streamRepresentation );
PaUtil_FreeMemory( stream );
}
return result;
}
static PaError StartStream( PaStream *s )
{
PaMacCoreStream *stream = (PaMacCoreStream*)s;
OSStatus 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
}
// 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;
OSStatus result = noErr;
PaError paErr;
VVDBUG(("StopStream()\n"));
VDBUG( ("Waiting for BLIO.\n") );
waitUntilBlioWriteBufferIsFlushed( &stream->blio );
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( 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;
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 );
}