我正在为一个项目编写一个 CoreAudio 后端名为 XAL 的音频库 http://libxal.googlecode.com/。输入缓冲器可以具有不同的采样率。我使用单个音频单元进行输出。想法是在将缓冲区发送到音频单元之前对其进行转换和混合。
只要输入缓冲区具有与输出音频单元相同的属性（采样率、通道数等），一切都可以正常工作。因此，混合部分起作用。
但是，我陷入了采样率和通道数转换的困境。据我所知，使用音频转换器服务 API 最容易做到这一点。我已经成功构建了一个转换器；这个想法是输出格式与输出单元格式相同，但可能会根据转换器的目的进行调整。
音频转换器已成功构建，但在调用时AudioConverterFillComplexBuffer()，我得到输出状态错误-50。
如果我能让另一群人关注这段代码，我会很高兴。问题可能出在下面的某个地方AudioConverterNew()。多变的stream包含传入（和传出）缓冲区数据，以及streamSize包含传入（和传出）缓冲区数据的字节大小。
我做错了什么？
void CoreAudio_AudioManager::_convertStream(Buffer* buffer, unsigned char** stream, int *streamSize)
{
if (buffer->getBitsPerSample() != unitDescription.mBitsPerChannel ||
buffer->getChannels() != unitDescription.mChannelsPerFrame ||
buffer->getSamplingRate() != unitDescription.mSampleRate)
{
printf("INPUT STREAM SIZE: %d\n", *streamSize);
// describe the input format's description
AudioStreamBasicDescription inputDescription;
memset(&inputDescription, 0, sizeof(inputDescription));
inputDescription.mFormatID = kAudioFormatLinearPCM;
inputDescription.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger;
inputDescription.mChannelsPerFrame = buffer->getChannels();
inputDescription.mSampleRate = buffer->getSamplingRate();
inputDescription.mBitsPerChannel = buffer->getBitsPerSample();
inputDescription.mBytesPerFrame = (inputDescription.mBitsPerChannel * inputDescription.mChannelsPerFrame) / 8;
inputDescription.mFramesPerPacket = 1; //*streamSize / inputDescription.mBytesPerFrame;
inputDescription.mBytesPerPacket = inputDescription.mBytesPerFrame * inputDescription.mFramesPerPacket;
printf("INPUT : %lu bytes per packet for sample rate %g, channels %d\n", inputDescription.mBytesPerPacket, inputDescription.mSampleRate, inputDescription.mChannelsPerFrame);
// copy conversion output format's description from the
// output audio unit's description.
// then adjust framesPerPacket to match the input we'll be passing.
// framecount of our input stream is based on the input bytecount.
// output stream will have same number of frames, but different
// number of bytes.
AudioStreamBasicDescription outputDescription = unitDescription;
outputDescription.mFramesPerPacket = 1; //inputDescription.mFramesPerPacket;
outputDescription.mBytesPerPacket = outputDescription.mBytesPerFrame * outputDescription.mFramesPerPacket;
printf("OUTPUT : %lu bytes per packet for sample rate %g, channels %d\n", outputDescription.mBytesPerPacket, outputDescription.mSampleRate, outputDescription.mChannelsPerFrame);
// create an audio converter
AudioConverterRef audioConverter;
OSStatus acCreationResult = AudioConverterNew(&inputDescription, &outputDescription, &audioConverter);
printf("Created audio converter %p (status: %d)\n", audioConverter, acCreationResult);
if(!audioConverter)
{
// bail out
free(*stream);
*streamSize = 0;
*stream = (unsigned char*)malloc(0);
return;
}
// calculate number of bytes required for output of input stream.
// allocate buffer of adequate size.
UInt32 outputBytes = outputDescription.mBytesPerPacket * (*streamSize / inputDescription.mBytesPerFrame); // outputDescription.mFramesPerPacket * outputDescription.mBytesPerFrame;
unsigned char *outputBuffer = (unsigned char*)malloc(outputBytes);
memset(outputBuffer, 0, outputBytes);
printf("OUTPUT BYTES : %d\n", outputBytes);
// describe input data we'll pass into converter
AudioBuffer inputBuffer;
inputBuffer.mNumberChannels = inputDescription.mChannelsPerFrame;
inputBuffer.mDataByteSize = *streamSize;
inputBuffer.mData = *stream;
// describe output data buffers into which we can receive data.
AudioBufferList outputBufferList;
outputBufferList.mNumberBuffers = 1;
outputBufferList.mBuffers[0].mNumberChannels = outputDescription.mChannelsPerFrame;
outputBufferList.mBuffers[0].mDataByteSize = outputBytes;
outputBufferList.mBuffers[0].mData = outputBuffer;
// set output data packet size
UInt32 outputDataPacketSize = outputDescription.mBytesPerPacket;
// convert
OSStatus result = AudioConverterFillComplexBuffer(audioConverter, /* AudioConverterRef inAudioConverter */
CoreAudio_AudioManager::_converterComplexInputDataProc, /* AudioConverterComplexInputDataProc inInputDataProc */
&inputBuffer, /* void *inInputDataProcUserData */
&outputDataPacketSize, /* UInt32 *ioOutputDataPacketSize */
&outputBufferList, /* AudioBufferList *outOutputData */
NULL /* AudioStreamPacketDescription *outPacketDescription */
);
printf("Result: %d wheee\n", result);
// change "stream" to describe our output buffer.
// even if error occured, we'd rather have silence than unconverted audio.
free(*stream);
*stream = outputBuffer;
*streamSize = outputBytes;
// dispose of the audio converter
AudioConverterDispose(audioConverter);
}
}
OSStatus CoreAudio_AudioManager::_converterComplexInputDataProc(AudioConverterRef inAudioConverter,
UInt32* ioNumberDataPackets,
AudioBufferList* ioData,
AudioStreamPacketDescription** ioDataPacketDescription,
void* inUserData)
{
printf("Converter\n");
if(*ioNumberDataPackets != 1)
{
xal::log("_converterComplexInputDataProc cannot provide input data; invalid number of packets requested");
*ioNumberDataPackets = 0;
ioData->mNumberBuffers = 0;
return -50;
}
*ioNumberDataPackets = 1;
ioData->mNumberBuffers = 1;
ioData->mBuffers[0] = *(AudioBuffer*)inUserData;
*ioDataPacketDescription = NULL;
return 0;
}
