//Allocate Audio Buffer List(s) to hold the data from input.
OSStatus CAPlayThrough::SetupBuffers()
{
OSStatus err = noErr;
UInt32 bufferSizeFrames,bufferSizeBytes,propsize;
CAStreamBasicDescription asbd,asbd_dev1_in,asbd_dev2_out;
Float64 rate=0;
//Get the size of the IO buffer(s)
UInt32 propertySize = sizeof(bufferSizeFrames);
err = AudioUnitGetProperty(mInputUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &bufferSizeFrames, &propertySize);
bufferSizeBytes = bufferSizeFrames * sizeof(Float32);
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev1_in);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &asbd_dev1_in, &propertySize);
//printf("=====Input DEVICE stream format\n" );
//asbd_dev1_in.Print();
//Get the Stream Format (client side)
propertySize = sizeof(asbd);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, &propertySize);
//printf("=====current Input (Client) stream format\n");
//asbd.Print();
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev2_out);
err = AudioUnitGetProperty(mOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd_dev2_out, &propertySize);
//printf("=====Output (Device) stream format\n");
//asbd_dev2_out.Print();
//////////////////////////////////////
//Set the format of all the AUs to the input/output devices channel count
//For a simple case, you want to set this to the lower of count of the channels
//in the input device vs output device
//////////////////////////////////////
asbd.mChannelsPerFrame =((asbd_dev1_in.mChannelsPerFrame < asbd_dev2_out.mChannelsPerFrame) ?asbd_dev1_in.mChannelsPerFrame :asbd_dev2_out.mChannelsPerFrame) ;
//printf("Info: Input Device channel count=%ld\t Input Device channel count=%ld\n",asbd_dev1_in.mChannelsPerFrame,asbd_dev2_out.mChannelsPerFrame);
//printf("Info: CAPlayThrough will use %ld channels\n",asbd.mChannelsPerFrame);
// We must get the sample rate of the input device and set it to the stream format of AUHAL
propertySize = sizeof(Float64);
AudioDeviceGetProperty(mInputDevice.mID, 0, 1, kAudioDevicePropertyNominalSampleRate, &propertySize, &rate);
asbd.mSampleRate =rate;
propertySize = sizeof(asbd);
//Set the new formats to the AUs...
err = AudioUnitSetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mVarispeedUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
//Set the correct sample rate for the output device, but keep the channel count the same
propertySize = sizeof(Float64);
AudioDeviceGetProperty(mOutputDevice.mID, 0, 0, kAudioDevicePropertyNominalSampleRate, &propertySize, &rate);
asbd.mSampleRate =rate;
propertySize = sizeof(asbd);
//Set the new audio stream formats for the rest of the AUs...
err = AudioUnitSetProperty(mVarispeedUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd, propertySize);
checkErr(err);
//set the audio stream to the mixer to be the same on both sides
err = AudioUnitSetProperty(mMixerUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mMixerUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
// turn metering ON for the mixer
UInt32 data = 1;
err = AudioUnitSetProperty(mMixerUnit, kAudioUnitProperty_MeteringMode, kAudioUnitScope_Global, 0, &data, propertySize);
checkErr(err);
//calculate number of buffers from channels
propsize = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) *asbd.mChannelsPerFrame);
//malloc buffer lists
mInputBuffer = (AudioBufferList *)malloc(propsize);
mInputBuffer->mNumberBuffers = asbd.mChannelsPerFrame;
//pre-malloc buffers for AudioBufferLists
for(UInt32 i =0; i< mInputBuffer->mNumberBuffers ; i++) {
mInputBuffer->mBuffers[i].mNumberChannels = 1;
mInputBuffer->mBuffers[i].mDataByteSize = bufferSizeBytes;
mInputBuffer->mBuffers[i].mData = malloc(bufferSizeBytes);
}
//Alloc ring buffer that will hold data between the two audio devices
mBuffer = new CARingBuffer();
mBuffer->Allocate(asbd.mChannelsPerFrame, asbd.mBytesPerFrame, bufferSizeFrames * 20);
return err;
}
//Allocate Audio Buffer List(s) to hold the data from input.
OSStatus CAPlayThrough::SetupBuffers()
{
OSStatus err = noErr;
UInt32 bufferSizeFrames,bufferSizeBytes,propsize;
CAStreamBasicDescription asbd,asbd_dev1_in,asbd_dev2_out;
Float64 rate=0;
//Get the size of the IO buffer(s)
UInt32 propertySize = sizeof(bufferSizeFrames);
err = AudioUnitGetProperty(mInputUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &bufferSizeFrames, &propertySize);
bufferSizeBytes = bufferSizeFrames * sizeof(Float32);
CAPT_DEBUG( "Input device buffer size is %ld frames.\n", bufferSizeFrames );
UInt32 outBufferSizeFrames;
propertySize = sizeof(outBufferSizeFrames);
err = AudioUnitGetProperty(mOutputUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &outBufferSizeFrames, &propertySize);
CAPT_DEBUG( "Output device buffer size is %ld frames.\n", outBufferSizeFrames );
outBufferSizeFrames = bufferSizeFrames;
propertySize = sizeof(outBufferSizeFrames);
err = AudioUnitSetProperty(mVarispeedUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &outBufferSizeFrames, propertySize);
CAPT_DEBUG( "Varispeed device buffer size is %ld frames.\n", outBufferSizeFrames );
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev1_in);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &asbd_dev1_in, &propertySize);
// CAPT_DEBUG("=====Input DEVICE stream format\n" );
// asbd_dev1_in.Print();
//Get the Stream Format (client side)
propertySize = sizeof(asbd);
err = AudioUnitGetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, &propertySize);
// CAPT_DEBUG("=====current Input (Client) stream format\n");
// asbd.Print();
//Get the Stream Format (Output client side)
propertySize = sizeof(asbd_dev2_out);
err = AudioUnitGetProperty(mOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd_dev2_out, &propertySize);
// CAPT_DEBUG("=====Output (Device) stream format\n");
// asbd_dev2_out.Print();
//////////////////////////////////////
//Set the format of all the AUs to the input/output devices channel count
//For a simple case, you want to set this to the lower of count of the channels
//in the input device vs output device
//////////////////////////////////////
asbd.mChannelsPerFrame =((asbd_dev1_in.mChannelsPerFrame < asbd_dev2_out.mChannelsPerFrame) ?asbd_dev1_in.mChannelsPerFrame :asbd_dev2_out.mChannelsPerFrame) ;
CAPT_DEBUG("Info: Input Device channel count=%ld\t Input Device channel count=%ld\n",asbd_dev1_in.mChannelsPerFrame,asbd_dev2_out.mChannelsPerFrame);
CAPT_DEBUG("Info: CAPlayThrough will use %ld channels\n",asbd.mChannelsPerFrame);
// We must get the sample rate of the input device and set it to the stream format of AUHAL
propertySize = sizeof(Float64);
AudioDeviceGetProperty(mInputDevice.mID, 0, 1, kAudioDevicePropertyNominalSampleRate, &propertySize, &rate);
asbd.mSampleRate =rate;
propertySize = sizeof(asbd);
//Set the new formats to the AUs...
err = AudioUnitSetProperty(mInputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mVarispeedUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
//Set the correct sample rate for the output device, but keep the channel count the same
propertySize = sizeof(Float64);
AudioDeviceGetProperty(mOutputDevice.mID, 0, 0, kAudioDevicePropertyNominalSampleRate, &propertySize, &rate);
asbd.mSampleRate =rate;
propertySize = sizeof(asbd);
//Set the new audio stream formats for the rest of the AUs...
err = AudioUnitSetProperty(mVarispeedUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &asbd, propertySize);
checkErr(err);
err = AudioUnitSetProperty(mOutputUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &asbd, propertySize);
checkErr(err);
//calculate number of buffers from channels
propsize = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) *asbd.mChannelsPerFrame);
//malloc buffer lists
mInputBuffer = (AudioBufferList *)malloc(propsize);
mInputBuffer->mNumberBuffers = asbd.mChannelsPerFrame;
//pre-malloc buffers for AudioBufferLists
for(UInt32 i =0; i< mInputBuffer->mNumberBuffers ; i++) {
mInputBuffer->mBuffers[i].mNumberChannels = 1;
mInputBuffer->mBuffers[i].mDataByteSize = bufferSizeBytes;
mInputBuffer->mBuffers[i].mData = malloc(bufferSizeBytes);
}
//Alloc ring buffer that will hold data between the two audio devices
mBuffer = new CARingBuffer();
mBuffer->Allocate(asbd.mChannelsPerFrame, asbd.mBytesPerFrame, bufferSizeFrames * 20 );
// Some test code to run the ring through its paces...
//
// CARingBuffer::SampleTime readTime = 0;
// CARingBuffer::SampleTime writeTime = 0;
// for ( int i = 0; i < 32768; i++ ) {
// printf( "run buf %d\n", i );
// UInt32 writeLength = 512;
// UInt32 readLength = 256;//(i%2==0) ? 440 : 441;
// OSStatus err = mBuffer->Store( mInputBuffer, writeLength, writeTime );
// if( err != kCARingBufferError_OK )
// {
// printf( "HIT ERR IN Store()\n" );
// }
// writeTime += writeLength;
//
// err = mBuffer->Fetch( mInputBuffer, readLength, readTime );
// if(err != kCARingBufferError_OK)
// {
// printf( "HIT ERR IN Fetch()\n" );
// MakeBufferSilent (mInputBuffer);
// SInt64 bufferStartTime, bufferEndTime;
// mBuffer->GetTimeBounds(bufferStartTime, bufferEndTime);
// }
// readTime += readLength;
// }
return err;
}
