多媒体开发（13）：iOS上音频编码成aac

标签：mpeg4   dex   com   rate   variable   test   文件   efi   source   

如前面我所说，对于音频的解码，一般你都不用考虑硬解，用软解就足够了，这时可以选择faad或FFmpeg等。但是，如果是音频的编码呢？这可不一样，编码比解码明显耗时，为了快跟低功耗（特别对于低端机器），要优先考虑硬编码（不能再使用fdk-aac或faac之类的软编码），硬编码的优势是可以用硬件芯片集成的功能，高速且低功耗地完成编码任务。

iOS平台，也提供了硬编码的能力，APP开发时只需要调用相应的SDK接口就能达成目标，这个SDK接口就是AudioConverter。

本文介绍iOS平台上，如何调用AudioConverter来完成aac的硬编码。

从名字来看，AudioConverter就是格式转换器，那就对了，这里把pcm格式的数据，转换成aac格式的数据。

AudioConverter在内存中实现转换，并不需要写文件，而ExtAudioFile接口则是对文件的操作，并且内部使用AudioConerter来转换格式，也就是说，你在某种场景下，也可以使用ExtAudioFile接口并接受临时文件的过程。

要独立操作，就要理解细节。具体如何使用AudioConverter呢？基本上，对接口的调用都需要阅读对应的头文件，通过看文档注释来理解怎么调用。

小程这里演示一下，怎么把pcm转换成aac。在演示代码之后，我只做简单的解释，如果你有需要，请耐心阅读代码来理解，并应用到自己的开发场景中。

typedef struct
{
    void *source;
    UInt32 sourceSize;
    UInt32 channelCount;
    AudioStreamPacketDescription *packetDescriptions;
}FillComplexInputParam;

// 填写源数据，即pcm数据
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
// 最终用AudioConverterNewSpecific创建ConvertContext，并设置比特率之类的属性
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;
}

// converting
void 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];
// 测试转换出来的aac数据，保存成adts-aac文件
#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;
}

以上代码，有两个函数比较重要，一个是初始化函数，这个函数创建了AudioConverterRef，另一个是转换函数，这个函数应该被反复调用，对不同的pcm数据进行转换。

另外，示例中，把pcm转换出来的aac数据，进行了保存，保存出来的文件可以用于播放。注意，AudioConverter转换出来的都是音频裸数据，至于组合成adts-aac，还是封装成苹果的m4a文件，由你的程序决定。

这里解释一下，adts-aac是aac数据的一种表示方式，也就是在每帧aac裸数据前面，增加一个帧信息（包括每帧的长度、采样率、声道数等），加上帧信息后，每帧aac可以单独播放。而且，adts-aac没有特定的文件头以及文件结构等。adts是Audio Data Transport Stream的缩写。

当然，你也可以把转换出来的aac数据，封装成m4a格式，这种封装格式，先是文件头，然后是box的组合（包括音频数据mdat等），可参考mp4封装格式。

至此，iOS平台把pcm转换成aac数据的实现就介绍完毕了。

总结一下，本文介绍了如何使用iOS平台提供的AudioConverter接口，把pcm格式的数据转换成aac格式。文章也介绍了怎么保存成adts-aac文件，你可以通过这个办法检验转换出来的aac数据是否正确。

多媒体开发（13）：iOS上音频编码成aac

标签：mpeg4   dex   com   rate   variable   test   文件   efi   source   

原文地址：https://www.cnblogs.com/freeself/p/10979118.html