從CMSampleBufferRef中提取PCM數(shù)據(jù)

脈沖編碼調(diào)制，其實(shí)是將不規(guī)則的模擬信號(hào)轉(zhuǎn)換成數(shù)字信號(hào)，這樣就可以通過(guò)物理介質(zhì)存儲(chǔ)起來(lái)。
而聲音也是一種特定頻率（20-20000HZ）的模擬信號(hào)，也可以通過(guò)這種技術(shù)轉(zhuǎn)換成數(shù)字信號(hào)，從而保存下來(lái)。
PCM格式，就是錄制聲音時(shí)，保存的最原始的聲音數(shù)據(jù)格式。比如 wav格式的音頻，它其實(shí)就是給PCM數(shù)據(jù)流加上一段header數(shù)據(jù)，就成為了wav格式。而wav格式有時(shí)候之所以被稱為無(wú)損格式，就是因?yàn)樗４娴氖窃紁cm數(shù)據(jù)（也跟采樣率和比特率有關(guān)）。像我們耳熟能詳?shù)哪切┮纛l格式，mp3，aac等等，都是有損壓縮，為了節(jié)約占用空間，在很少損失音效的基礎(chǔ)上，進(jìn)行最大程度的壓縮。
所有的音頻編碼器，都支持pcm編碼，而且錄制的聲音，默認(rèn)也是PCM格式，所以我們下一步就是要獲取錄制的PCM數(shù)據(jù)。

-(NSData *) convertAudioSmapleBufferToPcmData:(CMSampleBufferRef) audioSample{

AudioStreamBasicDescription inAudioStreamBasicDescription = *CMAudioFormatDescriptionGetStreamBasicDescription((CMAudioFormatDescriptionRef)CMSampleBufferGetFormatDescription(pcmData));
    
//獲取CMBlockBufferRef
    CMBlockBufferRef blockBufferRef = CMSampleBufferGetDataBuffer(pcmData);
//獲取pcm數(shù)據(jù)大小
    size_t length = CMBlockBufferGetDataLength(blockBufferRef);
    
//分配空間
    char buffer[length];
//直接將數(shù)據(jù)copy至我們自己分配的內(nèi)存中
    CMBlockBufferCopyDataBytes(blockBufferRef, 0, length, buffer);
    
    if ((inAudioStreamBasicDescription.mFormatFlags & kAudioFormatFlagIsBigEndian) == kAudioFormatFlagIsBigEndian)
    {
        for (int i = 0; i < length; i += 2)
        {
            char tmp = buffer[i];
            buffer[i] = buffer[i+1];
            buffer[i+1] = tmp;
        }
    }
    
    uint32_t ch = inAudioStreamBasicDescription.mChannelsPerFrame;
    uint32_t fs = inAudioStreamBasicDescription.mSampleRate;

    //返回?cái)?shù)據(jù)
    return [NSData dataWithBytesNoCopy:buffer length:audioDataSize];
}

PCM填充CMSampleBufferRef

根據(jù)采樣精度我們可以知道一個(gè)采樣點(diǎn)的數(shù)據(jù)量，比如16位精度，即一個(gè)采樣點(diǎn)需要2子節(jié)，則有200ms需要的數(shù)據(jù)量為:

//200ms 采樣點(diǎn)數(shù)量
NSUInteger samples = self->mSampleRate * 200 * self->mChannelsPerFrame/1000;
//200ms pcm數(shù)量量
int len = samples*2;

PCM填充CMSampleBufferRef 代碼示例：

- (CMSampleBufferRef)createAudioSampleBuffer:(char*) buf withLen:(int) len withASBD:(AudioStreamBasicDescription) asbd{
    
    AudioBufferList audioData;
    audioData.mNumberBuffers = 1;
    char* tmp = malloc(len);
    memcpy(tmp, buf, len);
    
    audioData.mBuffers[0].mData = tmp;
    audioData.mBuffers[0].mNumberChannels = asbd.mChannelsPerFrame;
    audioData.mBuffers[0].mDataByteSize = len;
    
    
    CMSampleBufferRef buff = NULL;
    CMFormatDescriptionRef format =NULL;
    OSStatus status = CMAudioFormatDescriptionCreate(kCFAllocatorDefault, &asbd,0, NULL, 0, NULL, NULL, &format);
    
    if (status) {
        return nil;
    }
    CMSampleTimingInfo timing = {CMTimeMake(asbd.mFramesPerPacket,asbd.mSampleRate), kCMTimeZero, kCMTimeInvalid };
    
    
    status = CMSampleBufferCreate(kCFAllocatorDefault,NULL, false,NULL, NULL, format, (CMItemCount)asbd.mFramesPerPacket,1, &timing, 0,NULL, &buff);
    
    if (status) { //失敗
        return nil;
    }
    
    status = CMSampleBufferSetDataBufferFromAudioBufferList(buff,kCFAllocatorDefault,kCFAllocatorDefault,0, &audioData);
    
    if (tmp) {
        free(tmp);
    }
    CFRelease(format);
    
    return buff;
}