AAC（Advanced Audio Coding），中文名：高级音频编码，出现于1997年，基于MPEG-2的音频编码技术。由Fraunhofer
IIS、杜比实验室、AT&T、Sony等公司共同开发，目的是取代MP3格式。2000年，MPEG-4标准出现后，AAC重新集成了其特性，加入了SBR技术和PS技术，为了区别于传统的MPEG-2
AAC又称为MPEG-4 AAC。
iOS平台支持AAC编码器，主要使用AudioToolbox中的AudioConverter API。之所以做AAC编码器是因为在做一个HLS的功能，HLS要求的TS文件，需要视频采用H264编码，音频采用AAC编码。H264可以使用硬件或软件编码器，前面已经介绍。AAC也可以使用硬件或者软件编码，iOS全都支持。
首先需要创建一个Converter，也就是一个AAC Encoder，使用如下接口：
extern OSStatus
AudioConverterNew( const AudioStreamBasicDescription* inSourceFormat,
const AudioStreamBasicDescription* inDestinationFormat,
AudioConverterRef* outAudioConverter) __OSX_AVAILABLE_STARTING(__MAC_10_1,__IPHONE_2_0);
输入参数分别是源和目的的数据格式。
在AAC编码的场景下，源格式就是采集到的PCM数据，目的格式就是AAC。
AudioStreamBasicDescription inAudioStreamBasicDescription;
// FillOutASBDForLPCM()
inAudioStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
inAudioStreamBasicDescription.mSampleRate = 44100;
inAudioStreamBasicDescription.mBitsPerChannel = 16;
inAudioStreamBasicDescription.mFramesPerPacket = 1;
inAudioStreamBasicDescription.mBytesPerFrame = 2;
inAudioStreamBasicDescription.mBytesPerPacket = inAudioStreamBasicDescription.mBytesPerFrame * inAudioStreamBasicDescription.mFramesPerPacket;
inAudioStreamBasicDescription.mChannelsPerFrame = 1;
inAudioStreamBasicDescription.mFormatFlags = kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsNonInterleaved;
inAudioStreamBasicDescription.mReserved = 0;
AudioStreamBasicDescription outAudioStreamBasicDescription = {0}; // Always initialize the fields of a new audio stream basic description structure to zero, as shown here: ...
outAudioStreamBasicDescription.mChannelsPerFrame = 1;
outAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC;
UInt32 size = sizeof(outAudioStreamBasicDescription);
AudioFormatGetProperty(kAudioFormatProperty_FormatInfo, 0, NULL, &size, &outAudioStreamBasicDescription);
OSStatus status = AudioConverterNew(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, &_audioConverter);
if(status != 0) {NSLog(@"setup converter failed: %d", (int)status);}
这样就创建了AAC编码器，默认情况下，Apple会创建一个硬件编码器，如果硬件不可用，会创建软件编码器。
经过我的测试，硬件AAC编码器的编码时延很高，需要buffer大约2秒的数据才会开始编码。而软件编码器的编码时延就是正常的，只要喂给1024个样点，就会开始编码。
那么如何在创建的时候指定使用软件编码器呢？需要用到下面的接口：
- (AudioClassDescription *)getAudioClassDescriptionWithType:(UInt32)type
fromManufacturer:(UInt32)manufacturer
{
static AudioClassDescription desc;
UInt32 encoderSpecifier = type;
OSStatus st;
UInt32 size;
st = AudioFormatGetPropertyInfo(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size);
if (st) {
NSLog(@"error getting audio format propery info: %d", (int)(st));
return nil;
}
unsigned int count = size / sizeof(AudioClassDescription);
AudioClassDescription descriptions[count];
st = AudioFormatGetProperty(kAudioFormatProperty_Encoders,
sizeof(encoderSpecifier),
&encoderSpecifier,
&size,
descriptions);
if (st) {
NSLog(@"error getting audio format propery: %d", (int)(st));
return nil;
}
for (unsigned int i = 0; i < count; i++) {
if ((type == descriptions[i].mSubType) &&
(manufacturer == descriptions[i].mManufacturer)) {
memcpy(&desc, &(descriptions[i]), sizeof(desc));
return &desc;
}
}
return nil;
}
AudioClassDescription *desc = [self getAudioClassDescriptionWithType:kAudioFormatMPEG4AAC fromManufacturer:kAppleSoftwareAudioCodecManufacturer]; OSStatus status = AudioConverterNewSpecific(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, 1, desc, &_audioConverter);
如果要正确的编码，编码码率参数是必须设置的。否则编码时会返回560226676错误码（!dat）。
UInt32 ulBitRate = 64000;
UInt32 ulSize = sizeof(ulBitRate);
status = AudioConverterSetProperty(_audioConverter, kAudioConverterEncodeBitRate, ulSize, &ulBitRate);
需要注意，AAC并不是随便的码率都可以支持。比如如果PCM采样率是44100KHz，那么码率可以设置64000bps，如果是16K，可以设置为32000bps。
创建完成Converter和设置完Bitrate之后，可以查询一下最大编码输出的大小，后续会用到。
UInt32 value = 0;
size = sizeof(value);
AudioConverterGetProperty(_audioConverter, kAudioConverterPropertyMaximumOutputPacketSize, &size, &value);
获取出来的Value表示编码器最大输出的包大小。
然后调用AudioConverterFillCOmplexBuffer进行编码：
AudioBufferList outAudioBufferList = {0};
outAudioBufferList.mNumberBuffers = 1;
outAudioBufferList.mBuffers[0].mNumberChannels = 1;
outAudioBufferList.mBuffers[0].mDataByteSize = value;//value是上面查询到的值
outAudioBufferList.mBuffers[0].mData = new int8[value];
UInt32 ioOutputDataPacketSize = 1;
status = AudioConverterFillComplexBuffer(_audioConverter, inInputDataProc, (__bridge void *)(self), &ioOutputDataPacketSize, &outAudioBufferList, NULL);
编码接口中，inInputDataProc是一个输入数据的回调函数。用来喂PCM数据给Converter，ioOutputDataPacketSize为1表示编码产生1帧数据即返回。outAudioBufferList用来存放编码后的数据。
inInputDataProc中的处理如下：
static OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
AACEncoder *encoder = (__bridge AACEncoder *)(inUserData);
UInt32 requestedPackets = *ioNumberDataPackets;
uint8_t *buffer;
uint32_t bufferLength = requestedPackets * 2;
uint32_t bufferRead;
bufferRead = [encoder.pcmPool readBuffer:&buffer withLength:bufferLength];
if (bufferRead == 0) {
*ioNumberDataPackets = 0;
return -1;
}
ioData->mBuffers[0].mData = buffer;
ioData->mBuffers[0].mDataByteSize = bufferRead;
ioData->mNumberBuffers = 1;
ioData->mBuffers[0].mNumberChannels = 1;
*ioNumberDataPackets = bufferRead >> 1;
return noErr;
}
pcmPool是一个用于存放PCM数据的环形缓冲区。
因为采集输入每次不一定有1024样点，所以可以将数据缓存起来，再满足1024样点时再调用编码。
另外，对于TS文件来说，每个AAC数据需要增加一个adts头，adts头是一个7bit的数据，通过adts可以得知AAC数据的编码参数，方便解码器进行解码。
adts头的计算方法如下：
- (NSData*) adtsDataForPacketLength:(NSUInteger)packetLength {
int adtsLength = 7;
char *packet = (char *)malloc(sizeof(char) * adtsLength);
// Variables Recycled by addADTStoPacket
int profile = 2; //AAC LC
//39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 8; //16KHz
int chanCfg = 1; //MPEG-4 Audio Channel Configuration. 1 Channel front-center
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;
NSData *data = [NSData dataWithBytesNoCopy:packet length:adtsLength freeWhenDone:YES];
return data;
}
adts头的计算需要几个参数：profile／frequency／channels／length，具体可参考http://wiki.multimedia.cx/index.php?title=ADTS
参考文献：
Audio Converter Services Reference
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时间： 2024-11-08 21:36:44