pa_mac_core.c

一个开源的sip源代码

       long requested = 64;
       if( inputParameters )
          requested = MAX( requested, inputParameters->suggestedLatency * sampleRate / 2 );
       if( outputParameters )
          requested = MAX( requested, outputParameters->suggestedLatency *sampleRate / 2 );
       VDBUG( ("Block Size unspecified. Based on Latency, the user wants a Block Size near: %ld.\n",
              requested ) );
       if( requested <= 64 ) {
          /*requested a realtively low latency. make sure this is in range of devices */
          /*try to get the device's min natural buffer size and use that (but no smaller than 64).*/
          AudioValueRange audioRange;
          size_t size = sizeof( audioRange );
          if( inputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MAX( requested, audioRange.mMinimum );
          }
          if( outputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MAX( requested, audioRange.mMinimum );
          }
       } else {
          /* requested a realtively high latency. make sure this is in range of devices */
          /*try to get the device's max natural buffer size and use that (but no larger than 1024).*/
          AudioValueRange audioRange;
          size_t size = sizeof( audioRange );
          requested = MIN( requested, 1024 );
          if( inputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MIN( requested, audioRange.mMaximum );
          }
          if( outputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MIN( requested, audioRange.mMaximum );
          }
       }
       /* -- double check ranges -- */
       if( requested > 1024 ) requested = 1024;
       if( requested < 64 ) requested = 64;
       VDBUG(("After querying hardware, setting block size to %ld.\n", requested));
       framesPerBuffer = requested;
    }

    /* -- Now we actually open and setup streams. -- */
    if( inputParameters && outputParameters && outputParameters->device == inputParameters->device )
    { /* full duplex. One device. */
       result = OpenAndSetupOneAudioUnit( inputParameters,
                                          outputParameters,
                                          framesPerBuffer,
                                          &(stream->inputFramesPerBuffer),
                                          &(stream->outputFramesPerBuffer),
                                          auhalHostApi,
                                          &(stream->inputUnit),
                                          &(stream->inputSRConverter),
                                          &(stream->inputDevice),
                                          sampleRate,
                                          stream );
       stream->outputUnit = stream->inputUnit;
       stream->outputDevice = stream->inputDevice;
       if( result != paNoError )
           goto error;
    }
    else
    { /* full duplex, different devices OR simplex */
       result = OpenAndSetupOneAudioUnit( NULL,
                                          outputParameters,
                                          framesPerBuffer,
                                          NULL,
                                          &(stream->outputFramesPerBuffer),
                                          auhalHostApi,
                                          &(stream->outputUnit),
                                          NULL,
                                          &(stream->outputDevice),
                                          sampleRate,
                                          stream );
       if( result != paNoError )
           goto error;
       result = OpenAndSetupOneAudioUnit( inputParameters,
                                          NULL,
                                          framesPerBuffer,
                                          &(stream->inputFramesPerBuffer),
                                          NULL,
                                          auhalHostApi,
                                          &(stream->inputUnit),
                                          &(stream->inputSRConverter),
                                          &(stream->inputDevice),
                                          sampleRate,
                                          stream );
       if( result != paNoError )
           goto error;
    }

    if( stream->inputUnit ) {
       const size_t szfl = sizeof(float);
       /* setup the AudioBufferList used for input */
       bzero( &stream->inputAudioBufferList, sizeof( AudioBufferList ) );
       stream->inputAudioBufferList.mNumberBuffers = 1;
       stream->inputAudioBufferList.mBuffers[0].mNumberChannels
                 = inputChannelCount;
       stream->inputAudioBufferList.mBuffers[0].mDataByteSize
                 = stream->inputFramesPerBuffer*inputChannelCount*szfl;
       stream->inputAudioBufferList.mBuffers[0].mData
                 = (float *) calloc(
                               stream->inputFramesPerBuffer*inputChannelCount,
                               szfl );
       if( !stream->inputAudioBufferList.mBuffers[0].mData )
       {
          result = paInsufficientMemory;
          goto error;
       }
        
       /*
        * If input and output devs are different or we are doing SR conversion,
        * we also need a
        * ring buffer to store inpt data while waiting for output
        * data.
        */
       if( (stream->outputUnit && stream->inputUnit != stream->outputUnit)
           || stream->inputSRConverter )
       {
          /* May want the ringSize ot initial position in
             ring buffer to depend somewhat on sample rate change */

          void *data;
          long ringSize;

          ringSize = computeRingBufferSize( inputParameters,
                                            outputParameters,
                                            stream->inputFramesPerBuffer,
                                            stream->outputFramesPerBuffer,
                                            sampleRate );
          /*ringSize <<= 4; *//*16x bigger, for testing */


          /*now, we need to allocate memory for the ring buffer*/
          data = calloc( ringSize, szfl );
          if( !data )
          {
             result = paInsufficientMemory;
             goto error;
          }

          /* now we can initialize the ring buffer */
          PaUtil_InitializeRingBuffer( &stream->inputRingBuffer,
                                   ringSize*szfl, data ) ;
          /* advance the read point a little, so we are reading from the
             middle of the buffer */
          if( stream->outputUnit )
             PaUtil_AdvanceRingBufferWriteIndex( &stream->inputRingBuffer, ringSize*szfl / RING_BUFFER_ADVANCE_DENOMINATOR );
       }
    }

    /* -- initialize Blio Buffer Processors -- */
    if( !streamCallback )
    {
       long ringSize;

       ringSize = computeRingBufferSize( inputParameters,
                                         outputParameters,
                                         stream->inputFramesPerBuffer,
                                         stream->outputFramesPerBuffer,
                                         sampleRate );
       result = initializeBlioRingBuffers( &stream->blio,
              inputParameters?inputParameters->sampleFormat:0 ,
              outputParameters?outputParameters->sampleFormat:0 ,
              MAX(stream->inputFramesPerBuffer,stream->outputFramesPerBuffer),
              ringSize,
              inputParameters?inputChannelCount:0 ,
              outputParameters?outputChannelCount:0 ) ;
       if( result != paNoError )
          goto error;
    }

    /* -- initialize Buffer Processor -- */
    {
       unsigned long maxHostFrames = stream->inputFramesPerBuffer;
       if( stream->outputFramesPerBuffer > maxHostFrames )
          maxHostFrames = stream->outputFramesPerBuffer;
       result = PaUtil_InitializeBufferProcessor( &stream->bufferProcessor,
                 inputChannelCount, inputSampleFormat,
                 hostInputSampleFormat,
                 outputChannelCount, outputSampleFormat,
                 hostOutputSampleFormat,
                 sampleRate,
                 streamFlags,
                 framesPerBuffer,
                 /* If sample rate conversion takes place, the buffer size
                    will not be known. */
                 maxHostFrames,
                 stream->inputSRConverter
                              ? paUtilUnknownHostBufferSize
                              : paUtilBoundedHostBufferSize,
                 streamCallback ? streamCallback : BlioCallback,
                 streamCallback ? userData : &stream->blio );
       if( result != paNoError )
           goto error;
    }
    stream->bufferProcessorIsInitialized = TRUE;

    /*
        IMPLEMENT ME: initialise the following fields with estimated or actual
        values.
        I think this is okay the way it is br 12/1/05
        maybe need to change input latency estimate if IO devs differ
    */
    stream->streamRepresentation.streamInfo.inputLatency =
            PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor)/sampleRate;
    stream->streamRepresentation.streamInfo.outputLatency =
            PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor)/sampleRate;
    stream->streamRepresentation.streamInfo.sampleRate = sampleRate;

    stream->sampleRate  = sampleRate;
    stream->outDeviceSampleRate = 0;
    if( stream->outputUnit ) {
       Float64 rate;
       UInt32 size = sizeof( rate );
       result = ERR( AudioDeviceGetProperty( stream->outputDevice,
                                    0,
                                    FALSE,
                                    kAudioDevicePropertyNominalSampleRate,
                                    &size, &rate ) );
       if( result )
          goto error;
       stream->outDeviceSampleRate = rate;
    }
    stream->inDeviceSampleRate = 0;
    if( stream->inputUnit ) {
       Float64 rate;
       UInt32 size = sizeof( rate );
       result = ERR( AudioDeviceGetProperty( stream->inputDevice,
                                    0,
                                    TRUE,
                                    kAudioDevicePropertyNominalSampleRate,
                                    &size, &rate ) );
       if( result )
          goto error;
       stream->inDeviceSampleRate = rate;
    }
    stream->userInChan  = inputChannelCount;
    stream->userOutChan = outputChannelCount;

    stream->isTimeSet   = FALSE;
    stream->state = STOPPED;
    stream->xrunFlags = 0;

    *s = (PaStream*)stream;

    return result;

error:
    CloseStream( stream );
    return result;
}

PaTime GetStreamTime( PaStream *s )
{
   /* FIXME: I am not at all sure this timing info stuff is right.
             patest_sine_time reports negative latencies, which is wierd.*/
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    AudioTimeStamp timeStamp;

    VVDBUG(("GetStreamTime()\n"));

    if ( !stream->isTimeSet )
        return (PaTime)0;

    if ( stream->outputDevice ) {
        AudioDeviceGetCurrentTime( stream->outputDevice, &timeStamp);
        return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->outDeviceSampleRate;
    } else if ( stream->inputDevice ) {
        AudioDeviceGetCurrentTime( stream->inputDevice, &timeStamp);
    return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->inDeviceSampleRate;
    } else {
        return (PaTime)0;
    }
}

static void setStreamStartTime( PaStream *stream )
{
   /* FIXME: I am not at all sure this timing info stuff is right.
             patest_sine_time reports negative latencies, which is wierd.*/
   PaMacCoreStream *s = (PaMacCoreStream *) stream;
   VVDBUG(("setStreamStartTime()\n"));
   if( s->outputDevice )
      AudioDeviceGetCurrentTime( s->outputDevice, &s->startTime);
   else if( s->inputDevice )
      AudioDeviceGetCurrentTime( s->inputDevice, &s->startTime);
   else
      bzero( &s->startTime, sizeof( s->startTime ) );

   //FIXME: we need a memory barier here

   s->isTimeSet = TRUE;
}


static PaTime TimeStampToSecs(PaMacCoreStream *stream, const AudioTimeStamp* timeStamp)
{
    VVDBUG(("TimeStampToSecs()\n"));
    //printf( "ATS: %lu, %g, %g\n", timeStamp->mFlags, timeStamp->mSampleTime, timeStamp->mRateScalar );
    if (timeStamp->mF