iOS音视频开发-音频硬编码-AudioToolbox-PCMToAAC

之前几篇文章记录了视频的软、硬编码过程，接下来将记录下音频的软、硬编码过程，学习、工作之余，以免忘记。
视频编码地址：
iOS音视频开发-视频会话捕捉
iOS音视频开发-视频硬编码(H264)
iOS音视频开发-视频软编码(x264编码H.264文件)
iOS音视频开发-视频软编码(FFmpeg+x264编码H.264文件)

PCM数据
PCM(Pulse Code Modulation)也被称为脉冲编码调制。PCM音频数据是未经压缩的音频采样数据裸流，它是由模拟信号经过采样、量化、编码转换成的标准的数字音频数据。
移动端对音频的实时采集编码传输，一般为将采集的音频数据设置为PCM格式数据，然后将PCM编码为AAC格式数据，以便后续传输。
PCM的数据格式，这里有篇文章介绍的很好，后续代码中的采样率、声道等均参考此文章设置。
PCM数据格式文章地址：点这里

ADTS
ADTS全称是(Audio Data Transport Stream)，是AAC的一种十分常见的传输格式。
将PCM数据编码为AAC的时候需要将每帧的AAC数据添加ADTS header，否则将无法解码播放。
ADTS数据格式分为两部分：
固定头部：adts_fixed_header
可变头部：adts_variable_header
详见Wiki，地址在文章末尾。

主要代码
1、音频捕获代码

#import "BBAudioCapture.h"
#import <AVFoundation/AVFoundation.h>
#import "BBAudioConfig.h"
#import "BBAudioHardEncoder.h"
#import "BBAudioHardEncoder.h"

@interface BBAudioCapture ()
{
    AudioComponentInstance _outInstance;
}
@property (nonatomic, assign) AudioComponent        component;
@property (nonatomic, strong) AVAudioSession        *session;
@property (nonatomic, strong) BBAudioHardEncoder    *encoder;
@property (nonatomic, strong) NSFileHandle          *handle;
@end

@implementation BBAudioCapture

#pragma mark -- 对象销毁方法
- (void)dealloc{
    AudioComponentInstanceDispose(_outInstance);
}

#pragma mark -- 对外API（控制是否捕捉音频数据）
- (void)startRunning{
    AudioOutputUnitStart(_outInstance);
}

-(void)stopRunning{
    AudioOutputUnitStop(_outInstance);
}

#pragma mark -- 对外API（设置捕获音频数据配置项）
- (void)setConfig:(BBAudioConfig *)config{
    _config = config;
    [self private_setupAudioSession];
}

#pragma mark -- 私有API（初始化音频会话）
- (void)private_setupAudioSession{

    //0.初始化编码器
    self.encoder = [[BBAudioHardEncoder alloc] init];
    self.encoder.config = self.config;

    //1.获取音频会话实例
    self.session = [AVAudioSession sharedInstance];

    NSError *error = nil;
    [self.session setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:AVAudioSessionCategoryOptionMixWithOthers | AVAudioSessionCategoryOptionDefaultToSpeaker error:&error];

    if (error) {
        NSLog(@"AVAudioSession setupError");
        error = nil;
        return;
    }

    //2.激活会话
    [self.session setActive:YES error:&error];

    if (error) {
        NSLog(@"AVAudioSession setActiveError");
        error = nil;
        return;
    }

    //3.设置模式
    [self.session setMode:AVAudioSessionModeVideoRecording error:&error];

    if (error) {
        NSLog(@"AVAudioSession setModeError");
        error = nil;
        return;
    }

    //4.设置音频单元
    AudioComponentDescription acd = {
        .componentType = kAudioUnitType_Output,
        .componentSubType = kAudioUnitSubType_RemoteIO,
        .componentManufacturer = kAudioUnitManufacturer_Apple,
        .componentFlags = 0,
        .componentFlagsMask = 0,
    };

    //5.查找音频单元
    self.component = AudioComponentFindNext(NULL, &acd);

    //6.获取音频单元实例
    OSStatus status = AudioComponentInstanceNew(self.component, &_outInstance);

    if (status != noErr) {
        NSLog(@"AudioSource new AudioComponent error");
        status = noErr;
        return;
    }

    //7.设置音频单元属性-->可读写 0-->不可读写 1-->可读写
    UInt32 flagOne = 1;
    AudioUnitSetProperty(_outInstance, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &flagOne, sizeof(flagOne));

    //8.设置音频单元属性-->音频流
    AudioStreamBasicDescription asbd = {0};
    asbd.mSampleRate = self.config.sampleRate;//采样率
    asbd.mFormatID = kAudioFormatLinearPCM;//原始数据为PCM格式
    asbd.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
    asbd.mChannelsPerFrame = (UInt32)self.config.channels;//每帧的声道数量
    asbd.mFramesPerPacket = 1;//每个数据包多少帧
    asbd.mBitsPerChannel = 16;//16位
    asbd.mBytesPerFrame = asbd.mChannelsPerFrame * asbd.mBitsPerChannel / 8;//每帧多少字节 bytes -> bit / 8
    asbd.mBytesPerPacket = asbd.mFramesPerPacket * asbd.mBytesPerFrame;//每个包多少字节

    status = AudioUnitSetProperty(_outInstance, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &asbd, sizeof(asbd));

    if (status != noErr) {
        NSLog(@"AudioUnitSetProperty StreamFormat error");
        status = noErr;
        return;
    }

    //9.设置回调函数
    AURenderCallbackStruct cb;
    cb.inputProcRefCon = (__bridge void *)self;
    cb.inputProc = audioBufferCallBack;

    status = AudioUnitSetProperty(_outInstance, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 1, &cb, sizeof(cb));

    if(status != noErr){
        NSLog(@"AudioUnitSetProperty StreamFormat InputCallback error");
        status = noErr;
        return;
    }

    //10.初始化音频单元
    status = AudioUnitInitialize(_outInstance);

    if (status != noErr) {
        NSLog(@"AudioUnitInitialize error");
        status = noErr;
        return;
    }

    //11.设置优先采样率
    [self.session setPreferredSampleRate:self.config.sampleRate error:&error];

    if (error) {
        NSLog(@"AudioSource setPreferredSampleRate error");
        error = nil;
        return;
    }

    //12.aac文件夹地址
    NSString *audioPath = [[NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject] stringByAppendingPathComponent:@"test.aac"];
    [[NSFileManager defaultManager] removeItemAtPath:audioPath error:nil];
    [[NSFileManager defaultManager] createFileAtPath:audioPath contents:nil attributes:nil];
    self.handle = [NSFileHandle fileHandleForWritingAtPath:audioPath];

}

#pragma mark -- 音频流回调函数
static OSStatus audioBufferCallBack(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
    @autoreleasepool {
        BBAudioCapture *capture = (__bridge BBAudioCapture *)inRefCon;
        if(!capture) return -1;

        AudioBuffer buffer;
        buffer.mData = NULL;
        buffer.mDataByteSize = 0;
        buffer.mNumberChannels = 1;

        AudioBufferList buffers;
        buffers.mNumberBuffers = 1;
        buffers.mBuffers[0] = buffer;

        OSStatus status = AudioUnitRender(capture->_outInstance,
        ioActionFlags,
        inTimeStamp,
        inBusNumber,
        inNumberFrames,
        &buffers);

        if(status == noErr) {
            [capture.encoder encodeWithBufferList:buffers completianBlock:^(NSData *encodedData, NSError *error) {
                if (error) {
                    NSLog(@"error:%@",error);
                    return;
                }

            NSLog(@"write to file!");
            [capture.handle writeData:encodedData];
            }];
        }
        return status;
    }
}

@end

2、编码代码

#import "BBAudioHardEncoder.h"
#import "BBAudioConfig.h"

@interface BBAudioHardEncoder ()
@property (nonatomic, assign) AudioConverterRef converterRef;
@end
@implementation BBAudioHardEncoder

- (AudioConverterRef)converterRef{
    if (_converterRef == nil) {
        [self private_setupAudioConvert];
    }
    return _converterRef;
}

- (void)dealloc {
    AudioConverterDispose(_converterRef);
}

- (void)private_setupAudioConvert{

    //1.输入流
    AudioStreamBasicDescription inputFormat = {0};
    inputFormat.mSampleRate = self.config.sampleRate;//采样率
    inputFormat.mFormatID = kAudioFormatLinearPCM;//PCM采样
    inputFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
    inputFormat.mChannelsPerFrame = (UInt32)self.config.channels;//每帧声道数
    inputFormat.mFramesPerPacket = 1;//每包帧数
    inputFormat.mBitsPerChannel = 16;//每声道位数
    inputFormat.mBytesPerFrame = inputFormat.mBitsPerChannel / 8 * inputFormat.mChannelsPerFrame;//每帧的字节数
    inputFormat.mBytesPerPacket = inputFormat.mBytesPerFrame * inputFormat.mFramesPerPacket;//每包字节数

    //2.输出流
    AudioStreamBasicDescription outputFormat;
    //2.1初始清零
    memset(&outputFormat, 0, sizeof(outputFormat));
    //2.2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。
    outputFormat.mSampleRate       = inputFormat.mSampleRate;
    //2.3AAC编码 kAudioFormatMPEG4AAC kAudioFormatMPEG4AAC_HE_V2
    outputFormat.mFormatID         = kAudioFormatMPEG4AAC;
    //2.4无损编码，0则无
    outputFormat.mFormatFlags      = kMPEG4Object_AAC_LC;
    //2.5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。
    outputFormat.mBytesPerPacket   = 0;
    //2.6每帧的声道数
    outputFormat.mChannelsPerFrame = (UInt32)self.config.channels;
    //2.7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。
    outputFormat.mFramesPerPacket  = 1024;
    //2.8每帧的bytes数，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 。
    outputFormat.mBytesPerFrame = 0;
    //2.9语音每采样点占用位数 压缩格式设置为0
    outputFormat.mBitsPerChannel = 0;
    //2.10字节对齐，填0.
    outputFormat.mReserved = 0;

    //3.编码器参数
    const OSType subtype = kAudioFormatMPEG4AAC;
    AudioClassDescription requestedCodecs[2] = {
        {
            kAudioEncoderComponentType,
            subtype,
            kAppleSoftwareAudioCodecManufacturer
        },
        {
            kAudioEncoderComponentType,
            subtype,
            kAppleHardwareAudioCodecManufacturer
        }
    };

    //4.编码器
    OSStatus result = AudioConverterNewSpecific(&inputFormat, &outputFormat, 2, requestedCodecs, &_converterRef);

    if (result == noErr) {
        NSLog(@"creat convert success!");
    }else{
        NSLog(@"creat convert error!");
        _converterRef = nil;
    }

}

- (void)encodeWithBufferList:(AudioBufferList)bufferList completianBlock:(void (^)(NSData *encodedData, NSError *error))completionBlock{
    if (!self.converterRef) {
        return;
    }
    int size = bufferList.mBuffers[0].mDataByteSize;

    if (size <= 0) {
        return;
    }

    char *aacBuf = malloc(size);

    //1.初始化一个输出缓冲列表
    AudioBufferList outBufferList;
    outBufferList.mNumberBuffers              = 1;
    outBufferList.mBuffers[0].mNumberChannels = bufferList.mBuffers[0].mNumberChannels;
    outBufferList.mBuffers[0].mDataByteSize   = bufferList.mBuffers[0].mDataByteSize; // 设置缓冲区大小
    outBufferList.mBuffers[0].mData           = aacBuf; // 设置AAC缓冲区
    UInt32 outputDataPacketSize               = 1;

    NSData *data = nil;
    NSError *error = nil;
    OSStatus status = AudioConverterFillComplexBuffer(_converterRef, inputDataProc, &bufferList, &outputDataPacketSize, &outBufferList, NULL);
    if (status == 0){
        NSData *rawAAC = [NSData dataWithBytes:outBufferList.mBuffers[0].mData length:outBufferList.mBuffers[0].mDataByteSize];
        NSData *adtsHeader = [self getADTSDataWithPacketLength:rawAAC.length];
        NSMutableData *fullData = [NSMutableData dataWithData:adtsHeader];
        [fullData appendData:rawAAC];
        data = fullData;
    }else{
        error = [NSError errorWithDomain:NSOSStatusErrorDomain code:status userInfo:nil];
        NSLog(@"音频编码失败");
        return;
    }

    if (completionBlock) {
        dispatch_async(dispatch_get_global_queue(0, 0), ^{
        completionBlock(data, error);
        });
    }
    free(aacBuf);
}

#pragma mark -- AudioCallBack
OSStatus inputDataProc(AudioConverterRef inConverter, UInt32 *ioNumberDataPackets, AudioBufferList *ioData,AudioStreamPacketDescription **outDataPacketDescription, void *inUserData) {
    //填充PCM到缓冲区
    AudioBufferList bufferList = *(AudioBufferList*)inUserData;
    ioData->mBuffers[0].mNumberChannels = 1;
    ioData->mBuffers[0].mData           = bufferList.mBuffers[0].mData;
    ioData->mBuffers[0].mDataByteSize   = bufferList.mBuffers[0].mDataByteSize;
    ioData->mNumberBuffers              = 1;
    return noErr;
}

/**
*  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 *)getADTSDataWithPacketLength:(NSInteger)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;
}
@end

以上代码为主要代码，配置代码无非就是采样率：44.1kHz，声道：双声道。
完整代码地址：https://github.com/ibabyblue/PCMHardEncodeToAAC
将编码的AAC数据写入本地文件，利用VLC播放器可以直接播放.aac格式文件，测试很方便。

写在最后，一路学习，坎坷良多，非常感谢无私分享的开发者们，感谢Jovins、iossigner分享的干货，感恩！
参考地址：
http://blog.csdn.net/ownwell/article/details/8114121/
https://wiki.multimedia.cx/index.php?title=ADTS
https://developer.apple.com/documentation/audiotoolbox/audio_converter_services?language=objc