开发中通常会遇到的音频流的处理，我们知道从苹果设备麦克风上面获取的声音是驳杂的，我们利用原生框架<AVFoundation/AVFoundation.h>可以把音频源经过PCM编码后得到音频裸流，这个音频流理论上你也可以在网络上传输（推流），但是这个裸流内容量太大，不仅传输慢，而且服务器承受不了。这时候我们就需要将其编码，进而引入AAC高级音频编码！
1、PCM编码
通常我们采用的是脉冲代码调制编码，即PCM编码。PCM通过抽样、量化、编码三个步骤将连续变化的模拟信号转换为数字编码。
抽样：对模拟信号进行周期性扫描，把时间上连续的信号变成时间上离散的信号；
量化：用一组规定的电平，把瞬时抽样值用最接近的电平值来表示,通常是用二进制表示；
编码：用一组二进制码组来表示每一个有固定电平的量化值；
采样后的数据大小 = 采样率值×采样大小值×声道数 bps。
一个采样率为44.1KHz，采样大小为16bit，双声道的PCM编码的WAV文件，它的数据速率=44.1K×16×2 bps=1411.2 Kbps= 176.4 KB/s。
这个速率和压缩后的视频数据速率差不多！
延伸出来AAC高级音频编码。
2、AAC高级音频编码
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。更具体的可以看看AAC的维基百科。
3、AAC音频格式
AAC音频格式有ADIF和ADTS：
ADIF：Audio Data Interchange Format 音频数据交换格式。这种格式的特征是可以确定的找到这个音频数据的开始，不需进行在音频数据流中间开始的解码，即它的解码必须在明确定义的开始处进行。故这种格式常用在磁盘文件中。
ADTS：Audio Data Transport Stream 音频数据传输流。这种格式的特征是它是一个有同步字的比特流，解码可以在这个流中任何位置开始，更利于网络传输。它的特征类似于mp3数据流格式。
4、iOS上把PCM音频编码成AAC音频流
1.设置编码器（codec），并开始录制；
2.收集到PCM数据，传给编码器；
3.编码完成回调callback，写入文件。
具体实现步骤：
ViewController.m
1.创建文件夹句柄
#pragma mark 创建文件夹句柄
- (void)createFileToDocument{
NSString *audioFile = [[NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject] stringByAppendingPathComponent:@"abc.aac"];
// 有就移除掉
[[NSFileManager defaultManager] removeItemAtPath:audioFile error:nil];
// 移除之后再创建
[[NSFileManager defaultManager] createFileAtPath:audioFile contents:nil attributes:nil];
self.audioFileHandle = [NSFileHandle fileHandleForWritingAtPath:audioFile];
}2.创建并配置AVCaptureSession
#pragma mark - 设置音频
- (void)setupAudioCapture {
self.aacEncoder = [[AACEncoder alloc] init];
self.session = [[AVCaptureSession alloc] init];
AVCaptureDevice *audioDevice = [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeAudio];
NSError *error = nil;
AVCaptureDeviceInput *audioInput = [[AVCaptureDeviceInput alloc]initWithDevice:audioDevice error:&error];
if (error) {
NSLog(@"Error getting audio input device:%@",error.description);
}
if ([self.session canAddInput:audioInput]) {
[self.session addInput:audioInput];
}
self.AudioQueue = dispatch_queue_create("Audio Capture Queue", DISPATCH_QUEUE_SERIAL);
AVCaptureAudioDataOutput *audioOutput = [AVCaptureAudioDataOutput new];
[audioOutput setSampleBufferDelegate:self queue:self.AudioQueue];
if ([self.session canAddOutput:audioOutput]) {
[self.session addOutput:audioOutput];
}
self.audioConnection = [audioOutput connectionWithMediaType:AVMediaTypeAudio];
}
3.接收到外界音频后回调代理方法
#pragma mark - 实现 AVCaptureOutputDelegate：
- (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection {
if (connection == _audioConnection)
{
// 音频
[self.aacEncoder encodeSampleBuffer:sampleBuffer completionBlock:^(NSData *encodedData, NSError *error) {
if (encodedData) {
//NSLog(@"Audio data (%lu):%@", (unsigned long)encodedData.length,encodedData.description);
[self.audioFileHandle writeData:encodedData];
}else {
NSLog(@"Error encoding AAC: %@", error);
}
}];
}else{
// 视频
}
}AACEncoder.m
4、创建转换器
AudioStreamBasicDescription是输出流的结构体描述，
配置好outAudioStreamBasicDescription后，
根据AudioClassDescription（编码器），
调用AudioConverterNewSpecific创建转换器。
- (void) setupEncoderFromSampleBuffer:(CMSampleBufferRef)sampleBuffer {
AudioStreamBasicDescription inAudioStreamBasicDescription = *CMAudioFormatDescriptionGetStreamBasicDescription((CMAudioFormatDescriptionRef)CMSampleBufferGetFormatDescription(sampleBuffer));
AudioStreamBasicDescription outAudioStreamBasicDescription = {0}; // 初始化输出流的结构体描述为0. 很重要。
outAudioStreamBasicDescription.mSampleRate = inAudioStreamBasicDescription.mSampleRate; // 音频流，在正常播放情况下的帧率。如果是压缩的格式，这个属性表示解压缩后的帧率。帧率不能为0。
outAudioStreamBasicDescription.mFormatID = kAudioFormatMPEG4AAC; // 设置编码格式
outAudioStreamBasicDescription.mFormatFlags = kMPEG4Object_AAC_LC; // 无损编码 ，0表示没有
outAudioStreamBasicDescription.mBytesPerPacket = 0; // 每一个packet的音频数据大小。如果的动态大小，设置为0。动态大小的格式，需要用AudioStreamPacketDescription 来确定每个packet的大小。
outAudioStreamBasicDescription.mFramesPerPacket = 1024; // 每个packet的帧数。如果是未压缩的音频数据，值是1。动态帧率格式，这个值是一个较大的固定数字，比如说AAC的1024。如果是动态大小帧数（比如Ogg格式）设置为0。
outAudioStreamBasicDescription.mBytesPerFrame = 0; // 每帧的大小。每一帧的起始点到下一帧的起始点。如果是压缩格式，设置为0 。
outAudioStreamBasicDescription.mChannelsPerFrame = 1; // 声道数
outAudioStreamBasicDescription.mBitsPerChannel = 0; // 压缩格式设置为0
outAudioStreamBasicDescription.mReserved = 0; // 8字节对齐，填0.
AudioClassDescription *description = [self
getAudioClassDescriptionWithType:kAudioFormatMPEG4AAC
fromManufacturer:kAppleSoftwareAudioCodecManufacturer]; //软编
OSStatus status = AudioConverterNewSpecific(&inAudioStreamBasicDescription, &outAudioStreamBasicDescription, 1, description, &_audioConverter); // 创建转换器
if (status != 0) {
NSLog(@"setup converter: %d", (int)status);
}
}获取编码器的方法
- (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;
}5、获取到PCM数据并传入编码器
用CMSampleBufferGetDataBuffer获取到CMSampleBufferRef里面的CMBlockBufferRef，
再通过CMBlockBufferGetDataPointer获取到_pcmBufferSize和_pcmBuffer；
调用AudioConverterFillComplexBuffer传入数据，并在callBack函数调用填充buffer的方法。
- (void) encodeSampleBuffer:(CMSampleBufferRef)sampleBuffer completionBlock:(void (^)(NSData * encodedData, NSError* error))completionBlock {
CFRetain(sampleBuffer);
dispatch_async(_encoderQueue, ^{
if (!_audioConverter) {
[self setupEncoderFromSampleBuffer:sampleBuffer];
}
CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer);
CFRetain(blockBuffer);
OSStatus status = CMBlockBufferGetDataPointer(blockBuffer, 0, NULL, &_pcmBufferSize, &_pcmBuffer);
NSError *error = nil;
if (status != kCMBlockBufferNoErr) {
error = [NSError errorWithDomain:NSOSStatusErrorDomain code:status userInfo:nil];
}
//NSLog(@"PCM Buffer Size: %zu", _pcmBufferSize);
memset(_aacBuffer, 0, _aacBufferSize);
AudioBufferList outAudioBufferList = {0};
outAudioBufferList.mNumberBuffers = 1;
outAudioBufferList.mBuffers[0].mNumberChannels = 1;
outAudioBufferList.mBuffers[0].mDataByteSize = (int)_aacBufferSize;
outAudioBufferList.mBuffers[0].mData = _aacBuffer;
AudioStreamPacketDescription *outPacketDescription = NULL;
UInt32 ioOutputDataPacketSize = 1;
status = AudioConverterFillComplexBuffer(_audioConverter, inInputDataProc, (__bridge void *)(self), &ioOutputDataPacketSize, &outAudioBufferList, outPacketDescription);
//NSLog(@"ioOutputDataPacketSize: %d", (unsigned int)ioOutputDataPacketSize);
NSData *data = nil;
if (status == 0) {
NSData *rawAAC = [NSData dataWithBytes:outAudioBufferList.mBuffers[0].mData length:outAudioBufferList.mBuffers[0].mDataByteSize];
NSData *adtsHeader = [self adtsDataForPacketLength:rawAAC.length];
NSMutableData *fullData = [NSMutableData dataWithData:adtsHeader];
[fullData appendData:rawAAC];
data = fullData;
} else {
error = [NSError errorWithDomain:NSOSStatusErrorDomain code:status userInfo:nil];
}
if (completionBlock) {
dispatch_async(_callbackQueue, ^{
completionBlock(data, error);
});
}
CFRelease(sampleBuffer);
CFRelease(blockBuffer);
});
}Callback函数
static OSStatus inInputDataProc(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void *inUserData)
{
AACEncoder *encoder = (__bridge AACEncoder *)(inUserData);
UInt32 requestedPackets = *ioNumberDataPackets;
//NSLog(@"Number of packets requested: %d", (unsigned int)requestedPackets);
size_t copiedSamples = [encoder copyPCMSamplesIntoBuffer:ioData];
if (copiedSamples < requestedPackets) {
//NSLog(@"PCM buffer isn't full enough!");
*ioNumberDataPackets = 0;
return -1;
}
*ioNumberDataPackets = 1;
//NSLog(@"Copied %zu samples into ioData", copiedSamples);
return noErr;
}
/**
* 填充PCM到缓冲区
*/
- (size_t) copyPCMSamplesIntoBuffer:(AudioBufferList*)ioData {
size_t originalBufferSize = _pcmBufferSize;
if (!originalBufferSize) {
return 0;
}
ioData->mBuffers[0].mData = _pcmBuffer;
ioData->mBuffers[0].mDataByteSize = (int)_pcmBufferSize;
_pcmBuffer = NULL;
_pcmBufferSize = 0;
return originalBufferSize;
}6、得到rawAAC码流，添加ADTS头，并写入文件
AudioConverterFillComplexBuffer返回的是AAC原始码流，需要在AAC每帧添加ADTS头，调用adtsDataForPacketLength方法生成，最后把数据写入audioFileHandle的文件。
/**
* 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
**/
- (NSData*) adtsDataForPacketLength:(NSUInteger)packetLength {
int adtsLength = 7;
char *packet = malloc(sizeof(char) * adtsLength);
// Variables Recycled by addADTStoPacket
int profile = 2; //AAC LC
//39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 4; //44.1KHz
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;
}
到这里基本上就实现了从麦克风录制音频并编码成AAC码流，代码实现起来不难，大都可以在网上找到，但是想要深究的话。还是需要细细的琢磨，要熟悉C语言才能更好的理解。这里感谢简书的落影loyinglin大牛，他对音视频编解码以及OpenGL都有很深的理解，值得学习！