将AudioBufferList转换为CMBlockBufferRef时出错

Question

我尝试使用AVAssetReader读取视频文件，并将音频传递给CoreAudio处理(添加效果和其他内容)，然后使用AVAssetWriter将其保存回磁盘。我想指出的是，如果我将输出节点的componentSubType设置为RemoteIO，则通过扬声器正确地播放。这使我有信心，我的AUGraph是正确的设置，因为我可以听到的东西工作。不过，我正在将subType设置为GenericOutput，这样我就可以自己进行渲染，并获得调整后的音频。

我正在读音频，我把CMSampleBufferRef传递给copyBuffer。这会将音频放入一个循环缓冲区中，稍后再读取。

- (void)copyBuffer:(CMSampleBufferRef)buf {  
    if (_readyForMoreBytes == NO)  
    {  
        return;  
    }  

    AudioBufferList abl;  
    CMBlockBufferRef blockBuffer;  
    CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(buf, NULL, &abl, sizeof(abl), NULL, NULL, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, &blockBuffer);  

    UInt32 size = (unsigned int)CMSampleBufferGetTotalSampleSize(buf);  
    BOOL bytesCopied = TPCircularBufferProduceBytes(&circularBuffer, abl.mBuffers[0].mData, size);  

    if (!bytesCopied){  
        /  
        _readyForMoreBytes = NO;  

        if (size > kRescueBufferSize){  
            NSLog(@"Unable to allocate enought space for rescue buffer, dropping audio frame");  
        } else {  
            if (rescueBuffer == nil) {  
                rescueBuffer = malloc(kRescueBufferSize);  
            }  

            rescueBufferSize = size;  
            memcpy(rescueBuffer, abl.mBuffers[0].mData, size);  
        }  
    }  

    CFRelease(blockBuffer);  
    if (!self.hasBuffer && bytesCopied > 0)  
    {  
        self.hasBuffer = YES;  
    }  
} 

接下来我叫processOutput。这将在outputUnit上执行手动修改。当AudioUnitRender被调用时，它会调用下面的playbackCallback，这就是在我的第一个节点上作为输入回调而连接起来的内容。playbackCallback从循环缓冲区中提取数据，并将其输入到传入的audioBufferList中。正如我之前所说，如果输出设置为RemoteIO，这将导致音频在扬声器上正确播放。当AudioUnitRender完成时，它返回noErr，bufferList对象包含有效数据。当我打电话给call时，得到了kCMSampleBufferError_RequiredParameterMissing (-12731)。

-(CMSampleBufferRef)processOutput  
{  
    if(self.offline == NO)  
    {  
        return NULL;  
    }  

    AudioUnitRenderActionFlags flags = 0;  
    AudioTimeStamp inTimeStamp;  
    memset(&inTimeStamp, 0, sizeof(AudioTimeStamp));  
    inTimeStamp.mFlags = kAudioTimeStampSampleTimeValid;  
    UInt32 busNumber = 0;  

    UInt32 numberFrames = 512;  
    inTimeStamp.mSampleTime = 0;  
    UInt32 channelCount = 2;  

    AudioBufferList *bufferList = (AudioBufferList*)malloc(sizeof(AudioBufferList)+sizeof(AudioBuffer)*(channelCount-1));  
    bufferList->mNumberBuffers = channelCount;  
    for (int j=0; j<channelCount; j++)  
    {  
        AudioBuffer buffer = {0};  
        buffer.mNumberChannels = 1;  
        buffer.mDataByteSize = numberFrames*sizeof(SInt32);  
        buffer.mData = calloc(numberFrames,sizeof(SInt32));  

        bufferList->mBuffers[j] = buffer;  

    }  
    CheckError(AudioUnitRender(outputUnit, &flags, &inTimeStamp, busNumber, numberFrames, bufferList), @"AudioUnitRender outputUnit");  

    CMSampleBufferRef sampleBufferRef = NULL;  
    CMFormatDescriptionRef format = NULL;  
    CMSampleTimingInfo timing = { CMTimeMake(1, 44100), kCMTimeZero, kCMTimeInvalid };  
    AudioStreamBasicDescription audioFormat = self.audioFormat;  
    CheckError(CMAudioFormatDescriptionCreate(kCFAllocatorDefault, &audioFormat, 0, NULL, 0, NULL, NULL, &format), @"CMAudioFormatDescriptionCreate");  
    CheckError(CMSampleBufferCreate(kCFAllocatorDefault, NULL, false, NULL, NULL, format, numberFrames, 1, &timing, 0, NULL, &sampleBufferRef), @"CMSampleBufferCreate");  
    CheckError(CMSampleBufferSetDataBufferFromAudioBufferList(sampleBufferRef, kCFAllocatorDefault, kCFAllocatorDefault, 0, bufferList), @"CMSampleBufferSetDataBufferFromAudioBufferList");  

    return sampleBufferRef;  
} 


static OSStatus playbackCallback(void *inRefCon,  
                                 AudioUnitRenderActionFlags *ioActionFlags,  
                                 const AudioTimeStamp *inTimeStamp,  
                                 UInt32 inBusNumber,  
                                 UInt32 inNumberFrames,  
                                 AudioBufferList *ioData)  
{  
    int numberOfChannels = ioData->mBuffers[0].mNumberChannels;  
    SInt16 *outSample = (SInt16 *)ioData->mBuffers[0].mData;  

    /  
    memset(outSample, 0, ioData->mBuffers[0].mDataByteSize);  

    MyAudioPlayer *p = (__bridge MyAudioPlayer *)inRefCon;  

    if (p.hasBuffer){  
        int32_t availableBytes;  
        SInt16 *bufferTail = TPCircularBufferTail([p getBuffer], &availableBytes);  

        int32_t requestedBytesSize = inNumberFrames * kUnitSize * numberOfChannels;  

        int bytesToRead = MIN(availableBytes, requestedBytesSize);  
        memcpy(outSample, bufferTail, bytesToRead);  
        TPCircularBufferConsume([p getBuffer], bytesToRead);  

        if (availableBytes <= requestedBytesSize*2){  
            [p setReadyForMoreBytes];  
        }  

        if (availableBytes <= requestedBytesSize) {  
            p.hasBuffer = NO;  
        }    
    }  
    return noErr;  
} 

我传入的CMSampleBufferRef看起来是有效的(下面是调试器中对象的转储)

CMSampleBuffer 0x7f87d2a03120 retainCount: 1 allocator: 0x103333180  
  invalid = NO  
  dataReady = NO  
  makeDataReadyCallback = 0x0  
  makeDataReadyRefcon = 0x0  
  formatDescription = <CMAudioFormatDescription 0x7f87d2a02b20 [0x103333180]> {  
  mediaType:'soun'  
  mediaSubType:'lpcm'  
  mediaSpecific: {  
  ASBD: {  
  mSampleRate: 44100.000000  
  mFormatID: 'lpcm'  
  mFormatFlags: 0xc2c  
  mBytesPerPacket: 2  
  mFramesPerPacket: 1  
  mBytesPerFrame: 2  
  mChannelsPerFrame: 1  
  mBitsPerChannel: 16 }  
  cookie: {(null)}  
  ACL: {(null)}  
  }  
  extensions: {(null)}  
}  
  sbufToTrackReadiness = 0x0  
  numSamples = 512  
  sampleTimingArray[1] = {  
  {PTS = {0/1 = 0.000}, DTS = {INVALID}, duration = {1/44100 = 0.000}},  
  }  
  dataBuffer = 0x0  

缓冲区列表如下所示

Printing description of bufferList:  
(AudioBufferList *) bufferList = 0x00007f87d280b0a0  
Printing description of bufferList->mNumberBuffers:  
(UInt32) mNumberBuffers = 2  
Printing description of bufferList->mBuffers:  
(AudioBuffer [1]) mBuffers = {  
  [0] = (mNumberChannels = 1, mDataByteSize = 2048, mData = 0x00007f87d3008c00)  
}  

真是不知所措，希望有人能帮忙。谢谢,

如果重要的话，我在ios 8.3模拟器上调试这个程序，音频来自我在iphone 6上拍摄的mp4，然后保存到我的笔记本上。

我读过以下几个问题，尽管仍然没有用，但事情并没有奏效。

How to convert AudioBufferList to CMSampleBuffer?

Converting an AudioBufferList to a CMSampleBuffer Produces Unexpected Results

CMSampleBufferSetDataBufferFromAudioBufferList returning error 12731

core audio offline rendering GenericOutput

更新

我四处打听了一下，注意到当我的AudioBufferList在AudioUnitRender运行之前运行时，如下所示：

bufferList->mNumberBuffers = 2,
bufferList->mBuffers[0].mNumberChannels = 1,
bufferList->mBuffers[0].mDataByteSize = 2048

mDataByteSize是数字框架*SInt32，它是512 * 4。当我查看在playbackCallback中传递的AudioBufferList时，列表如下：

bufferList->mNumberBuffers = 1,
bufferList->mBuffers[0].mNumberChannels = 1,
bufferList->mBuffers[0].mDataByteSize = 1024

不太确定另一个缓冲区的去向，或者其他1024字节大小.

如果我打完电话给雷德纳如果我做了这样的事

AudioBufferList newbuff;
newbuff.mNumberBuffers = 1;
newbuff.mBuffers[0] = bufferList->mBuffers[0];
newbuff.mBuffers[0].mDataByteSize = 1024;

然后传递给，错误就消失了。

如果我尝试将BufferList的大小设置为1 mNumberBuffers或其mDataByteSize设置为numberFrames*mDataByteSize(SInt16)，则在调用AudioUnitRender时将得到a -50。

更新2

我连接了一个渲染回调，这样，当我在扬声器上播放声音时，我可以检查输出。我注意到，到扬声器的输出还有一个带有2个缓冲区的AudioBufferList，输入回调期间的mDataByteSize是1024，在呈现回调中是2048，这与我手动调用AudioUnitRender时看到的相同。当我检查呈现的AudioBufferList中的数据时，我注意到两个缓冲区中的字节是相同的，这意味着我可以忽略第二个缓冲区。但是，我不知道如何处理这样一个事实，即数据在呈现后是2048大小，而不是在接收数据时的1024。知道为什么会发生这种事吗？在看完音频图之后，它是否以更原始的形式出现，这就是为什么大小加倍的原因吗？

Answer 1

听起来你要处理的问题是因为频道数量不一致。你看到数据块为2048年而不是1024的原因是因为它给你提供了两个通道(立体声)。检查以确保您的所有音频单元被正确地配置为在整个音频图中使用mono，包括音调单元和任何音频格式描述。

特别要注意的一点是，对AudioUnitSetProperty的调用可能会失败--所以一定要将这些调用包装在CheckError()中。