一、iOS中的音視訊都是採用一個叫做CMSampleBuffer的格式封裝的

　　比如回撥的App音訊

Printing description of sampleBuffer:
CMSampleBuffer 0x143d06560 retainCount: 1 allocator: 0x25bae95e0
	invalid = NO
	dataReady = YES
	makeDataReadyCallback = 0x0
	makeDataReadyRefcon = 0x0
	formatDescription = <CMAudioFormatDescription 0x281940990 [0x25bae95e0]> {
	mediaType:'soun' 
	mediaSubType:'lpcm' 
	mediaSpecific: {
		ASBD: {
			mSampleRate: 44100.000000 
			mFormatID: 'lpcm' 
			mFormatFlags: 0xe 
			mBytesPerPacket: 2 
			mFramesPerPacket: 1 
			mBytesPerFrame: 2 
			mChannelsPerFrame: 1 
			mBitsPerChannel: 16 	} 
		cookie: {(null)} 
		ACL: {(null)}
		FormatList Array: {(null)} 
	} 
	extensions: {(null)}
}
	sbufToTrackReadiness = 0x0
	numSamples = 22528
	sampleTimingArray[1] = {
		{PTS = {619877934457375/1000000000 = 619877.934}, DTS = {INVALID}, duration = {1/44100 = 0.000}},
	}
	dataBuffer = 0x281940360

　　比如回撥的micphone音訊

Printing description of sampleBuffer:
CMSampleBuffer 0x149e053b0 retainCount: 1 allocator: 0x25bae95e0
	invalid = NO
	dataReady = YES
	makeDataReadyCallback = 0x0
	makeDataReadyRefcon = 0x0
	formatDescription = <CMAudioFormatDescription 0x281b47b10 [0x25bae95e0]> {
	mediaType:'soun' 
	mediaSubType:'lpcm' 
	mediaSpecific: {
		ASBD: {
			mSampleRate: 44100.000000 
			mFormatID: 'lpcm' 
			mFormatFlags: 0xc 
			mBytesPerPacket: 2 
			mFramesPerPacket: 1 
			mBytesPerFrame: 2 
			mChannelsPerFrame: 1 
			mBitsPerChannel: 16 	} 
		cookie: {(null)} 
		ACL: {(null)}
		FormatList Array: {(null)} 
	}
 
 
	extensions: {(null)}
}
	sbufToTrackReadiness = 0x0
	numSamples = 1024
	sampleTimingArray[1] = {
		{PTS = {27273930501/44100 = 618456.474, rounded}, DTS = {INVALID}, duration = {1/44100 = 0.000}},
	}
	dataBuffer = 0x281b47c30

　從這個結構中，可以得到這個音訊資料的描述資訊

  // Get samples.
        CMBlockBufferRef audioBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
        size_t lengthAtOffset;
        size_t totalLength;
        char *samples;
        CMBlockBufferGetDataPointer(audioBuffer, 0, &lengthAtOffset, &totalLength, &samples);

        // Get format.
        CMAudioFormatDescriptionRef format = CMSampleBufferGetFormatDescription(sampleBuffer);
        const AudioStreamBasicDescription *description = CMAudioFormatDescriptionGetStreamBasicDescription(format);

　　輸出為格式的描述資訊

Printing description of format:
<CMAudioFormatDescription 0x28282f7b0 [0x25bae95e0]> {
	mediaType:'soun' 
	mediaSubType:'lpcm' 
	mediaSpecific: {
		ASBD: {
			mSampleRate: 44100.000000 
			mFormatID: 'lpcm' 
			mFormatFlags: 0xc 
			mBytesPerPacket: 2 
			mFramesPerPacket: 1 
			mBytesPerFrame: 2 
			mChannelsPerFrame: 1 
			mBitsPerChannel: 16 	} 
		cookie: {(null)} 
		ACL: {(null)}
		FormatList Array: {(null)} 
	} 
	extensions: {(null)}
}

　　關於音訊格式有幾個關鍵資訊：格式、取樣率、取樣精度、聲道數

　　關於音訊資料除了上面的格式資訊之外，還有 PTS 、 samples（幀數）描述資訊

　　比如上面的PCM資料中，取樣率是44100hz、取樣深度是16位、聲道數是1，samples是1024幀，最後計算這段聲音的長度資訊是 (1/44100) * 1024 = 23ms , 這個聲音的資料大小是 1024 * 2byte = 2014 Byte

　　如果是錄製聲音的話，需要先設定需要錄製的音訊格式資訊　　

AudioStreamBasicDescription format;
memset(&format, 0, sizeof(format));
format.mSampleRate = 44100;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
format.mChannelsPerFrame = 1;
format.mBitsPerChannel = 16;
format.mBytesPerFrame = (format.mBitsPerChannel/8) * format.mChannelsPerFrame;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;

　　需要設定關鍵引數、取樣率、取樣精度、聲道數、資料排列格式、　一個packet的幀數

附錄：

　　關於取樣精度

關於取樣率