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我正在爲audio library called XAL編寫CoreAudio後端。輸入緩衝器可以具有各種採樣率。我正在使用一個音頻單元來輸出。想法是轉換緩衝區並在將它們發送到音頻單元之前進行混合。OS X/iOS - 使用AudioConverterFillComplexBuffer對緩衝區進行採樣率轉換
只要輸入緩衝區具有與輸出音頻單元相同的屬性（採樣率，通道數等），一切都會正常工作。因此，混合部分起作用。
但是，我堅持採樣率和通道數轉換。從我發現的情況來看，這對於Audio Converter Services API來說是最簡單的。我設法構建了一個轉換器;其思想是輸出格式與輸出單元格式相同，但可能爲了轉換器的目的而進行調整。
音頻轉換器已成功構建，但撥打AudioConverterFillComplexBuffer()後，我得到輸出狀態錯誤-50。
如果我能在此代碼上獲得另一組眼球，我很樂意。問題可能在AudioConverterNew()以下。變量stream包含傳入（和傳出）緩衝區數據，並且streamSize包含傳入（和傳出）緩衝區數據的字節大小。
我做錯了什麼？
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;
}
在你的代碼中，我看到你有這個unitDescription.mSampleRate。我認爲這是你想要的輸出採樣率。但我似乎無法找到你告訴outputDescription所需採樣率的地方？ – user523234
該庫不需要也不需要進行任意轉換;它甚至進行轉換的唯一原因是它可以在「輸出音頻單元」上播放音頻 - 即揚聲器或耳機，大部分。因此，在進行轉換時，它將從輸出音頻單元中複製整個輸出音頻描述，並且只修改'mFramesPerPacket'。如果您需要獲得不同的速率，請嘗試更改'outputDescription'中的相應字段（這是一個'AudioStreamBasicDescription'類型的結構體）。並小心;在以後的實驗中，我有過很糟糕的時間。 –
但是當輸入格式中每個數據包的字節包含= 0時會發生什麼情況，VBR格式通常會出現這種情況？你不能除以零，除非你是查克·諾里斯！ ：p – abbood