iOS平台上音頻編碼成aac

標籤：AudioConverter   pcm   aac   寫入程式碼   

小程之前介紹解碼aac時，曾經使用了fadd，並且有提到，如果想編碼成aac格式，可以使用facc、fdk-aac等，但使用fdk-aac等編碼方式，都是軟編碼，在cpu的消耗上會明顯大於硬體編碼。

硬式編碼優勢是可以用硬體晶片整合的功能，高速且低功耗地完成編碼任務。

在iOS平台，也提供了硬式編碼能力，APP開發時只需要調用相應的SDK介面就可以了。

這個SDK介面就是AudioConverter。

本文介紹iOS平台上，如何調用AudioConverter來完成aac的寫入程式碼。

從名字來看，AudioConverter就是格式轉換器，這裡小程使用它，把pcm格式的資料，轉換成aac格式的資料。

對於媒體格式（編碼格式或封裝格式），讀者可以關注“廣州小程”公眾號，並在“音視頻->基礎概念與流程”菜單中查閱相關文章。

AudioConverter在記憶體中實現轉換，並不需要寫檔案，而ExtAudioFile介面則是對檔案的操作，並且內部使用AudioConerter來轉換格式，也就是說，讀者在某種情境下，也可以使用ExtAudioFile介面。

如何使用AudioConverter呢？基本上，對介面的調用都需要閱讀對應的標頭檔，通過看文檔注釋來理解怎麼調用。

小程這裡示範一下，怎麼把pcm格式的資料轉換成aac格式的資料。

在示範代碼之後，小程只做簡單的解釋，有需要的讀者請耐心閱讀代碼來理解，並應用到自己的開發情境中。

下面的例子示範從pcm轉aac的實現（比如把錄音資料儲存成aac的實現）。

typedef struct{    void *source;    UInt32 sourceSize;    UInt32 channelCount;    AudioStreamPacketDescription *packetDescriptions;}FillComplexInputParam;// 填寫來源資料，即pcm資料OSStatus audioConverterComplexInputDataProc(  AudioConverterRef               inAudioConverter,                                            UInt32*                         ioNumberDataPackets,                                            AudioBufferList*                ioData,                                            AudioStreamPacketDescription**  outDataPacketDescription,                                            void*                           inUserData){    FillComplexInputParam* param = (FillComplexInputParam*)inUserData;    if (param->sourceSize <= 0) {        *ioNumberDataPackets = 0;        return -1;    }    ioData->mBuffers[0].mData = param->source;    ioData->mBuffers[0].mNumberChannels = param->channelCount;    ioData->mBuffers[0].mDataByteSize = param->sourceSize;    *ioNumberDataPackets = 1;    param->sourceSize = 0;    param->source = NULL;    return noErr;}typedef struct _tagConvertContext {    AudioConverterRef converter;    int samplerate;    int channels;}ConvertContext;// init// 最終用AudioConverterNewSpecific建立ConvertContext，並設定位元速率之類的屬性void* convert_init(int sample_rate, int channel_count){    AudioStreamBasicDescription sourceDes;    memset(&sourceDes, 0, sizeof(sourceDes));    sourceDes.mSampleRate = sample_rate;    sourceDes.mFormatID = kAudioFormatLinearPCM;    sourceDes.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger;    sourceDes.mChannelsPerFrame = channel_count;    sourceDes.mBitsPerChannel = 16;    sourceDes.mBytesPerFrame = sourceDes.mBitsPerChannel/8*sourceDes.mChannelsPerFrame;    sourceDes.mBytesPerPacket = sourceDes.mBytesPerFrame;    sourceDes.mFramesPerPacket = 1;    sourceDes.mReserved = 0;    AudioStreamBasicDescription targetDes;    memset(&targetDes, 0, sizeof(targetDes));    targetDes.mFormatID = kAudioFormatMPEG4AAC;    targetDes.mSampleRate = sample_rate;    targetDes.mChannelsPerFrame = channel_count;    UInt32 size = sizeof(targetDes);    AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &targetDes);    AudioClassDescription audioClassDes;    memset(&audioClassDes, 0, sizeof(AudioClassDescription));    AudioFormatGetPropertyInfo(kAudioFormatProperty_Encoders, sizeof(targetDes.mFormatID), &targetDes.mFormatID, &size);    int encoderCount = size / sizeof(AudioClassDescription);    AudioClassDescription descriptions[encoderCount];    AudioFormatGetProperty(kAudioFormatProperty_Encoders, sizeof(targetDes.mFormatID), &targetDes.mFormatID, &size, descriptions);    for (int pos = 0; pos < encoderCount; pos ++) {        if (targetDes.mFormatID == descriptions[pos].mSubType && descriptions[pos].mManufacturer == kAppleSoftwareAudioCodecManufacturer) {            memcpy(&audioClassDes, &descriptions[pos], sizeof(AudioClassDescription));            break;        }    }    ConvertContext *convertContex = malloc(sizeof(ConvertContext));    OSStatus ret = AudioConverterNewSpecific(&sourceDes, &targetDes, 1, &audioClassDes, &convertContex->converter);    if (ret == noErr) {        AudioConverterRef converter = convertContex->converter;        tmp = kAudioConverterQuality_High;        AudioConverterSetProperty(converter, kAudioConverterCodecQuality, sizeof(tmp), &tmp);        UInt32 bitRate = 96000;        UInt32 size = sizeof(bitRate);        ret = AudioConverterSetProperty(converter, kAudioConverterEncodeBitRate, size, &bitRate);    }    else {        free(convertContex);        convertContex = NULL;    }    return convertContex;}// convertingvoid convert(void* convertContext, void* srcdata, int srclen, void** outdata, int* outlen){    ConvertContext* convertCxt = (ConvertContext*)convertContext;    if (convertCxt && convertCxt->converter) {        UInt32 theOuputBufSize = srclen;          UInt32 packetSize = 1;        void *outBuffer = malloc(theOuputBufSize);        memset(outBuffer, 0, theOuputBufSize);        AudioStreamPacketDescription *outputPacketDescriptions = NULL;        outputPacketDescriptions = (AudioStreamPacketDescription*)malloc(sizeof(AudioStreamPacketDescription) * packetSize);        FillComplexInputParam userParam;        userParam.source = srcdata;        userParam.sourceSize = srclen;        userParam.channelCount = convertCxt->channels;        userParam.packetDescriptions = NULL;        OSStatus ret = noErr;        AudioBufferList* bufferList = malloc(sizeof(AudioBufferList));        AudioBufferList outputBuffers = *bufferList;        outputBuffers.mNumberBuffers = 1;        outputBuffers.mBuffers[0].mNumberChannels = convertCxt->channels;        outputBuffers.mBuffers[0].mData = outBuffer;        outputBuffers.mBuffers[0].mDataByteSize = theOuputBufSize;        ret = AudioConverterFillComplexBuffer(convertCxt->converter, audioConverterComplexInputDataProc, &userParam, &packetSize, &outputBuffers, outputPacketDescriptions);        if (ret == noErr) {            if (outputBuffers.mBuffers[0].mDataByteSize > 0) {                NSData* rawAAC = [NSData dataWithBytes:outputBuffers.mBuffers[0].mData length:outputBuffers.mBuffers[0].mDataByteSize];                *outdata = malloc([rawAAC length]);                memcpy(*outdata, [rawAAC bytes], [rawAAC length]);                *outlen = (int)[rawAAC length];// 測試轉換出來的aac資料，儲存成adts-aac檔案#if 1                int headerLength = 0;                char* packetHeader = newAdtsDataForPacketLength((int)[rawAAC length], convertCxt->samplerate, convertCxt->channels, &headerLength);                NSData* adtsPacketHeader = [NSData dataWithBytes:packetHeader length:headerLength];                free(packetHeader);                NSMutableData* fullData = [NSMutableData dataWithData:adtsPacketHeader];                [fullData appendData:rawAAC];                NSFileManager *fileMgr = [NSFileManager defaultManager];                NSString *filepath = [NSHomeDirectory() stringByAppendingFormat:@"/Documents/test%p.aac", convertCxt->converter];                NSFileHandle *file = nil;                if (![fileMgr fileExistsAtPath:filepath]) {                    [fileMgr createFileAtPath:filepath contents:nil attributes:nil];                }                file = [NSFileHandle fileHandleForWritingAtPath:filepath];                [file seekToEndOfFile];                [file writeData:fullData];                [file closeFile];#endif            }        }        free(outBuffer);        if (outputPacketDescriptions) {            free(outputPacketDescriptions);        }    }}// uninit// ...int freqIdxForAdtsHeader(int samplerate){    /**     0: 96000 Hz     1: 88200 Hz     2: 64000 Hz     3: 48000 Hz     4: 44100 Hz     5: 32000 Hz     6: 24000 Hz     7: 22050 Hz     8: 16000 Hz     9: 12000 Hz     10: 11025 Hz     11: 8000 Hz     12: 7350 Hz     13: Reserved     14: Reserved     15: frequency is written explictly     */    int idx = 4;    if (samplerate >= 7350 && samplerate < 8000) {        idx = 12;    }    else if (samplerate >= 8000 && samplerate < 11025) {        idx = 11;    }    else if (samplerate >= 11025 && samplerate < 12000) {        idx = 10;    }    else if (samplerate >= 12000 && samplerate < 16000) {        idx = 9;    }    else if (samplerate >= 16000 && samplerate < 22050) {        idx = 8;    }    else if (samplerate >= 22050 && samplerate < 24000) {        idx = 7;    }    else if (samplerate >= 24000 && samplerate < 32000) {        idx = 6;    }    else if (samplerate >= 32000 && samplerate < 44100) {        idx = 5;    }    else if (samplerate >= 44100 && samplerate < 48000) {        idx = 4;    }    else if (samplerate >= 48000 && samplerate < 64000) {        idx = 3;    }    else if (samplerate >= 64000 && samplerate < 88200) {        idx = 2;    }    else if (samplerate >= 88200 && samplerate < 96000) {        idx = 1;    }    else if (samplerate >= 96000) {        idx = 0;    }    return idx;}int channelIdxForAdtsHeader(int channelCount){    /**     0: Defined in AOT Specifc Config     1: 1 channel: front-center     2: 2 channels: front-left, front-right     3: 3 channels: front-center, front-left, front-right     4: 4 channels: front-center, front-left, front-right, back-center     5: 5 channels: front-center, front-left, front-right, back-left, back-right     6: 6 channels: front-center, front-left, front-right, back-left, back-right, LFE-channel     7: 8 channels: front-center, front-left, front-right, side-left, side-right, back-left, back-right, LFE-channel     8-15: Reserved     */    int ret = 2;    if (channelCount == 1) {        ret = 1;    }    else if (channelCount == 2) {        ret = 2;    }    return ret;}/** *  Add ADTS header at the beginning of each and every AAC packet. *  This is needed as MediaCodec encoder generates a packet of raw *  AAC data. * *  Note the packetLen must count in the ADTS header itself. *  See: http://wiki.multimedia.cx/index.php?title=ADTS *  Also: http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio#Channel_Configurations **/char* newAdtsDataForPacketLength(int packetLength, int samplerate, int channelCount, int* ioHeaderLen) {    int adtsLength = 7;    char *packet = malloc(sizeof(char) * adtsLength);    // Variables Recycled by addADTStoPacket    int profile = 2;  //AAC LC    //39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;    int freqIdx = freqIdxForAdtsHeader(samplerate);    int chanCfg = channelIdxForAdtsHeader(channelCount);  //MPEG-4 Audio Channel Configuration.    NSUInteger fullLength = adtsLength + packetLength;    // fill in ADTS data    packet[0] = (char)0xFF;// 11111111  = syncword    packet[1] = (char)0xF9;// 1111 1 00 1  = syncword MPEG-2 Layer CRC    packet[2] = (char)(((profile-1)<<6) + (freqIdx<<2) +(chanCfg>>2));    packet[3] = (char)(((chanCfg&3)<<6) + (fullLength>>11));    packet[4] = (char)((fullLength&0x7FF) >> 3);    packet[5] = (char)(((fullLength&7)<<5) + 0x1F);    packet[6] = (char)0xFC;    *ioHeaderLen = adtsLength;    return packet;}

以上代碼，有兩個函數比較重要，一個是初始化函數，這個函數建立了AudioConverterRef，另一個是轉換函式，這個函數應該被反覆調用，對不同的pcm資料進行轉換。

另外，樣本中，把pcm轉換出來的aac資料，進行了儲存，儲存出來的檔案可以用於播放。

注意，AudioConverter轉換出來的都是音頻裸資料，至於組合成adts-aac，還是封裝成蘋果的m4a檔案，由程式決定。

這裡解釋一下，adts-aac是aac資料的一種表示方式，也就是在每幀aac裸資料前面，增加一個幀資訊（包括每幀的長度、採樣率、聲道數等），加上幀資訊後，每幀aac可以單獨播放。而且，adts-aac是沒有封裝的，也就是沒有特定的檔案頭以及檔案結構等。

adts是Audio Data Transport Stream的縮寫。

當然，讀者也可以把轉換出來的aac資料，封裝成m4a格式，這種封裝格式，先是檔案頭，然後就是祼音頻資料：

{packet-table}{audio_data}{trailer}，頭資訊之後就是音頻裸資料，音頻資料不帶packet資訊。

至此，iOS平台把pcm轉換成aac資料的實現就介紹完畢了。

總結一下，本文介紹了如何使用iOS平台提供的AudioConverter介面，把pcm格式的資料轉換成aac格式。文章也介紹了怎麼儲存成adts-aac檔案，讀者可以通過這個辦法檢驗轉換出來的aac資料是否正確。

iOS平台上音頻編碼成aac