我试图使用AVAssetReader读取视频文件并将音频传递给CoreAudio进行处理（添加特效等），然后使用AVAssetWriter将其保存回磁盘。我想指出的是，如果我在输出节点的AudioComponentDescription上设置componentSubType为RemoteIO，则可以正确播放内容。这让我有信心我的AUGraph已经正确设置，因为我可以听到它正在工作。但我将subType设置为GenericOutput，以便自己进行渲染并获得调整后的音频。
我正在读取音频，并将CMSampleBufferRef传递给copyBuffer。这将音频放入一个循环缓冲区，稍后将被读取。
我传递的CMSampleBufferRef看起来有效（下面是从调试器中转储的对象）。
缓冲区列表长这样。
我正在读取音频，并将CMSampleBufferRef传递给copyBuffer。这将音频放入一个循环缓冲区，稍后将被读取。
- (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;
}
}
接下来我调用processOutput。这将在outputUnit上手动渲染。当调用AudioUnitRender时，它会调用下面的playbackCallback，这是连接到我的第一个节点的输入回调。playbackCallback从循环缓冲区中取出数据，并将其馈入传递的audioBufferList中。正如我之前所说，如果输出设置为RemoteIO，则会导致音频正确播放到扬声器上。当AudioUnitRender完成后，它返回noErr，而bufferList对象包含有效数据。但是，当我调用CMSampleBufferSetDataBufferFromAudioBufferList时，我得到了kCMSampleBufferError_RequiredParameterMissing (-12731)的错误。
-(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;
}
我传递的CMSampleBufferRef看起来有效（下面是从调试器中转储的对象）。
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
缓冲区列表长这样。
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)
}
非常无助，希望有人能帮忙。谢谢。
如果重要的话，我是在iOS 8.3模拟器中进行调试的，音频来自于我用iPhone 6拍摄并保存到笔记本电脑上的mp4。
我已经阅读了以下问题，但仍然不起作用。
如何将AudioBufferList转换为CMSampleBuffer？
将AudioBufferList转换为CMSampleBuffer会产生意外结果
CMSampleBufferSetDataBufferFromAudioBufferList返回错误12731
更新
我继续查找并发现当我的AudioBufferList在执行AudioUnitRender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)的结果，即512 * 4。在我查看传递给playbackCallback的AudioBufferList时，列表如下所示：
bufferList->mNumberBuffers = 1,
bufferList->mBuffers[0].mNumberChannels = 1,
bufferList->mBuffers[0].mDataByteSize = 1024
我不太确定另一个缓冲区去了哪里，或者另一个1024字节大小的东西...
如果当我完成调用Redner后，我做了这样的事情
AudioBufferList newbuff;
newbuff.mNumberBuffers = 1;
newbuff.mBuffers[0] = bufferList->mBuffers[0];
newbuff.mBuffers[0].mDataByteSize = 1024;
如果我将BufferList的大小设置为1 mNumberBuffers或将其mDataByteSize设置为numberFrames*sizeof(SInt16)，则在调用AudioUnitRender时会出现-50的错误。
当我播放声音并通过渲染回调检查输出时，我发现输出到扬声器的输出也具有带有2个缓冲区的AudioBufferList，并且在输入回调期间的mDataByteSize为1024，在渲染回调中的mDataByteSize为2048，这与手动调用AudioUnitRender时看到的情况相同。当我检查渲染的AudioBufferList中的数据时，我注意到2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，我不确定如何处理渲染后数据的大小为2048而不是1024的事实。对于这种情况，您有什么想法吗？是否是在通过音频图形后以更原始的形式出现，这就是为什么大小会加倍的原因？
如果将newbuff传递给CMSampleBufferSetDataBufferFromAudioBufferList，则错误将消失。