那些遇到过的问题

OS X/iOS - 使用AudioConverterFillComplexBuffer的缓冲区的采样率转换

Iva*_*ica 9 iphone macos core-audio sample-rate audio-converter

我正在为一个名为XAL的音频库编写CoreAudio后端.输入缓冲区可以具有各种采样率.我正在使用单个音频单元进行输出.想法是转换缓冲区并在将它们发送到音频单元之前将它们混合.

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),一切都有效.因此,混合部分起作用.

但是,我坚持采样率和通道数转换.根据我的想法,这对于Audio Converter Services API来说是最容易的.我设法建造了一个转换器; 我们的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.

如果我能在这段代码上获得另一组眼球,我会很高兴.问题可能在下面的某个地方AudioConverterNew().变量stream包含传入(和传出)缓冲区数据,并包含传入(和传出)缓冲区数据的streamSize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;
}
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Iva*_*ica 10

Core Audio采样速率转换和通道数转换的工作代码,使用音频转换器服务(现在作为BSD许可的XAL音频库的一部分提供):

void CoreAudio_AudioManager::_convertStream(Buffer* buffer, unsigned char** stream, int *streamSize)
{
    if (buffer->getBitsPerSample() != unitDescription.mBitsPerChannel || 
        buffer->getChannels() != unitDescription.mChannelsPerFrame || 
        buffer->getSamplingRate() != unitDescription.mSampleRate)
    {
        // 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;

        // 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;

        // create an audio converter
        AudioConverterRef audioConverter;
        OSStatus acCreationResult = AudioConverterNew(&inputDescription, &outputDescription, &audioConverter);
        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.mBytesPerPacket); // outputDescription.mFramesPerPacket * outputDescription.mBytesPerFrame;
        unsigned char *outputBuffer = (unsigned char*)malloc(outputBytes);
        memset(outputBuffer, 0, 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 = outputBytes / outputDescription.mBytesPerPacket;

        // fill class members with data that we'll pass into
        // the InputDataProc
        _converter_currentBuffer = &inputBuffer;
        _converter_currentInputDescription = inputDescription;

        // convert
        OSStatus result = AudioConverterFillComplexBuffer(audioConverter, /* AudioConverterRef inAudioConverter */
                                                          CoreAudio_AudioManager::_converterComplexInputDataProc, /* AudioConverterComplexInputDataProc inInputDataProc */
                                                          this, /* void *inInputDataProcUserData */
                                                          &outputDataPacketSize, /* UInt32 *ioOutputDataPacketSize */
                                                          &outputBufferList, /* AudioBufferList *outOutputData */
                                                          NULL /* AudioStreamPacketDescription *outPacketDescription */
                                                          );

        // 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)
{
    if(ioDataPacketDescription)
    {
        xal::log("_converterComplexInputDataProc cannot provide input data; it doesn't know how to provide packet descriptions");
        *ioDataPacketDescription = NULL;
        *ioNumberDataPackets = 0;
        ioData->mNumberBuffers = 0;
        return 501;
    }

    CoreAudio_AudioManager *self = (CoreAudio_AudioManager*)inUserData;

    ioData->mNumberBuffers = 1;
    ioData->mBuffers[0] = *(self->_converter_currentBuffer);

    *ioNumberDataPackets = ioData->mBuffers[0].mDataByteSize / self->_converter_currentInputDescription.mBytesPerPacket;
    return 0;
}
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在标题中,作为CoreAudio_AudioManager类的一部分,这里是相关的实例变量:

    AudioStreamBasicDescription unitDescription;
    AudioBuffer *_converter_currentBuffer;
    AudioStreamBasicDescription _converter_currentInputDescription;
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几个月后,我正在看这个,我意识到我没有记录这些变化.

如果您对更改的内容感兴趣:

  • 看一下回调函数 CoreAudio_AudioManager::_converterComplexInputDataProc
  • 一个人必须正确指定输入数据包的数量 ioNumberDataPackets
  • 这需要引入新的实例变量来保存缓冲区(前一个inUserData)和输入描述(用于计算要输入Core Audio转换器的数据包数)
  • 这个"输出"数据包(馈入转换器的数据包)的计算是根据我们的回调接收的数据量以及输入格式包含的每个数据包的字节数完成的.

希望这个编辑将有助于未来的读者(包括我自己)!

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