多媒体开发（13）：iOS上音频编码成aac

# 移动开发 2024-05-05 12:16 0 17 来源： 云博客

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;}// 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];// 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。

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