多媒體開發（13）：iOS上音頻編碼成aac
如前面我所說，對於音頻的解碼，通常你都不用考慮硬解，用軟解就足夠了，這時能夠選擇faad或FFmpeg等。可是，若是是音頻的編碼呢？這可不同，編碼比解碼明顯耗時，爲了快跟低功耗（特別對於低端機器），要優先考慮硬編碼（不能再使用fdk-aac或faac之類的軟編碼），硬編碼的優點是能夠用硬件芯片集成的功能，高速且低功耗地完成編碼任務。php
iOS平臺，也提供了硬編碼的能力，APP開發時只須要調用相應的SDK接口就能達成目標，這個SDK接口就是AudioConverter。app
本文介紹iOS平臺上，如何調用AudioConverter來完成aac的硬編碼。ide
從名字來看，AudioConverter就是格式轉換器，那就對了，這裏把pcm格式的數據，轉換成aac格式的數據。函數
AudioConverter在內存中實現轉換，並不須要寫文件，而ExtAudioFile接口則是對文件的操做，而且內部使用AudioConerter來轉換格式，也就是說，你在某種場景下，也可使用ExtAudioFile接口並接受臨時文件的過程。測試
要獨立操做，就要理解細節。具體如何使用AudioConverter呢？基本上，對接口的調用都須要閱讀對應的頭文件，經過看文檔註釋來理解怎麼調用。編碼
小程這裏演示一下，怎麼把pcm轉換成aac。在演示代碼以後，我只作簡單的解釋，若是你有須要，請耐心閱讀代碼來理解，並應用到本身的開發場景中。code
typedef struct
{
void *source;
UInt32 sourceSize;
UInt32 channelCount;
AudioStreamPacketDescription *packetDescriptions;
}FillComplexInputParam;
// 填寫源數據，即pcm數據
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，並設置比特率之類的屬性
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;
}
// 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];
// 測試轉換出來的aac數據，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，另外一個是轉換函數，這個函數應該被反覆調用，對不一樣的pcm數據進行轉換。orm
另外，示例中，把pcm轉換出來的aac數據，進行了保存，保存出來的文件能夠用於播放。注意，AudioConverter轉換出來的都是音頻裸數據，至於組合成adts-aac，仍是封裝成蘋果的m4a文件，由你的程序決定。blog
這裏解釋一下，adts-aac是aac數據的一種表示方式，也就是在每幀aac裸數據前面，增長一個幀信息（包括每幀的長度、採樣率、聲道數等），加上幀信息後，每幀aac能夠單獨播放。並且，adts-aac沒有特定的文件頭以及文件結構等。adts是Audio Data Transport Stream的縮寫。接口
固然，你也能夠把轉換出來的aac數據，封裝成m4a格式，這種封裝格式，先是文件頭，而後是box的組合（包括音頻數據mdat等），可參考mp4封裝格式。
至此，iOS平臺把pcm轉換成aac數據的實現就介紹完畢了。
總結一下，本文介紹瞭如何使用iOS平臺提供的AudioConverter接口，把pcm格式的數據轉換成aac格式。文章也介紹了怎麼保存成adts-aac文件，你能夠經過這個辦法檢驗轉換出來的aac數據是否正確。
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