将 AudioBufferList 转换为 CMBlockBufferRef 时出错

error converting AudioBufferList to CMBlockBufferRef

我正在尝试使用 AVAssetReader 读取视频文件并将音频传递给 CoreAudio 进行处理(添加效果和内容),然后再使用 AVAssetWriter 将其保存回磁盘。我想指出的是,如果我将输出节点的 AudioComponentDescription 上的 componentSubType 设置为 RemoteIO,则可以通过扬声器正常播放。这让我确信我的 AUGraph 已正确设置,因为我可以听到一切正常。不过,我将子类型设置为 GenericOutput,这样我就可以自己进行渲染并取回调整后的音频。

我正在阅读音频并将 CMSampleBufferRef 传递给 copyBuffer。这会将音频放入一个循环缓冲区中,稍后将读取该缓冲区。

- (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;  
    }  
} 

接下来我调用processOutput。这将在 outputUnit 上进行手动渲染。当调用 AudioUnitRender 时,它会调用下面的 playbackCallback,这是我的第一个节点上作为输入回调连接的内容。 playbackCallback 从循环缓冲区中提取数据并将其馈送到传入的 audioBufferList 中。就像我之前说的,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 returns noErr 并且 bufferList 对象包含有效数据。 当我调用 CMSampleBufferSetDataBufferFromAudioBufferList 虽然我得到 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 看起来有效(下面是来自调试器的对象转储)

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)  
}  

在这里真的很茫然,希望有人能帮忙。谢谢,

以防万一,我正在 ios 8.3 模拟器中调试它,音频来自我在 iphone 6 上拍摄的 mp4,然后保存到我的笔记本电脑。

我已阅读以下问题,但仍然无济于事,一切都无法正常工作。

How to convert AudioBufferList to CMSampleBuffer?

CMSampleBufferSetDataBufferFromAudioBufferList returning error 12731

core audio offline rendering GenericOutput

更新

我又查了一下,注意到当我的 AudioBufferList 在 AudioUnitRender 运行之前看起来像这样:

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

mDataByteSize是numberFrames*sizeof(SInt32),也就是512*4,我看playbackCallback传入的AudioBufferList,列表是这样的:

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

不确定其他缓冲区或其他 1024 字节大小的去向...

如果当我完成呼叫 Redner 时如果我做这样的事情

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

并将 newbuff 传递给 CMSampleBufferSetDataBufferFromAudioBufferList,错误消失。

如果我尝试将 BufferList 的大小设置为具有 1 个 mNumberBuffers 或其 mDataByteSize 为 numberFrames*sizeof(SInt16) 我在调用 AudioUnitRender 时得到 -50

更新 2

我连接了一个渲染回调,这样我就可以在通过扬声器播放声音时检查输出。我注意到扬声器的输出也有一个带有 2 个缓冲区的 AudioBufferList,输入回调期间的 mDataByteSize 是 1024,在渲染回调中是 2048,这与我在手动调用 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。但是我不确定如何处理数据在渲染后大小为 2048 而不是在被接收时为 1024 的事实。关于为什么会发生这种情况的任何想法?通过音频图后它是否更像是原始形式,这就是大小加倍的原因?

听起来您遇到的问题是因为频道数量不一致。您在 2048 块而不是 1024 块中看到数据的原因是因为它向您反馈两个通道(立体声)。检查以确保所有音频单元都正确配置为在整个音频图中使用单声道,包括 Pitch Unit 和任何音频格式描述。

要特别注意的一件事是对 AudioUnitSetProperty 的调用可能会失败 - 因此请务必将它们也包含在 CheckError() 中。