利用AudioQueue做音頻采集編碼和播放(附完整demo)

概述

在直播應(yīng)用開(kāi)發(fā)中我們經(jīng)常需要實(shí)時(shí)對(duì)音頻做處理，比如音頻錄制、播放、編碼等。本文介紹的是使用AudioQueue對(duì)音頻做上述處理。

PCM和AAC是音頻的兩種不同的格式，PCM是無(wú)損音頻數(shù)據(jù)，AAC是壓縮編碼過(guò)的數(shù)據(jù)。我們?cè)诮榻BAudioQueue的用法之前，首先對(duì)音頻的這兩種格式做大致了解。關(guān)于音頻的基礎(chǔ)請(qǐng)參考 音頻基礎(chǔ)知識(shí)

文章目錄：

  • AAC音頻
  • AudioQueue錄制音頻原始幀PCM數(shù)據(jù)
  • AudioQueue播放PCM音頻文件
  • AudioQueue錄制音頻PCM數(shù)據(jù)，同時(shí)轉(zhuǎn)化為aac數(shù)據(jù)保存到沙盒中，最后利用ffmpeg命令查看并播放aac文件

示例代碼：

代碼結(jié)構(gòu):

image

AAC音頻

因?yàn)槲覀儠?huì)介紹從PCM轉(zhuǎn)化為AAC，所以我們先對(duì)AAC有一個(gè)大致的了解。

AAC的音頻文件格式有兩種分別是ADIF 和 ADTS 。

ADIF和ADTS

  • ADIF : (Audio Data Interchange Format )音頻數(shù)據(jù)交換格式。這種格式的特征是可以確定的找到這個(gè)音頻數(shù)據(jù)的開(kāi)始，不需進(jìn)行在音頻數(shù)據(jù)流中間開(kāi)始的解碼，即它的解碼必須在明確定義的開(kāi)始處進(jìn)行。故這種格式常用在磁盤(pán)文件中。
  • ADTS : (Audio Data Transport Stream)音頻數(shù)據(jù)傳輸流。這種格式的特征是它是一個(gè)有同步字的比特流，解碼可以在這個(gè)流中任何位置開(kāi)始。它的特征類似于mp3數(shù)據(jù)流格式。

簡(jiǎn)單說(shuō)，ADTS可以在任意幀解碼，也就是說(shuō)它每一幀都有頭信息。ADIF只有一個(gè)統(tǒng)一的頭，所以必須得到所有的數(shù)據(jù)后解碼。且這兩種的header的格式也是不同的，目前一般編碼后的和抽取出的都是ADTS格式的音頻流。

ADIF格式:

image

ADTS格式:

image

ADIF和ADTS的頭信息

ADIF的頭信息如圖：

image

ADIF頭信息位于AAC文件的起始處，接下來(lái)就是連續(xù)的 raw data blocks。組成 ADIF頭信息的各個(gè)域如下所示：

image

ADTS的固定頭信息：

image

ADTS的可變頭信息：

image
  • 幀同步目的在于找出幀頭在比特流中的位置，13818-7規(guī)定，aac ADTS格式的幀頭。同步字為12比特的“1111 1111 1111”
  • ADTS的頭信息為兩部分組成，其一為固定頭信息，緊接著是可變頭信息。固定頭信息中的數(shù)據(jù)每一幀都相同，而可變頭信息則在幀與幀之間可變

AAC元素信息

在AAC中，原始數(shù)據(jù)塊的組成可能有六種不同的元素：

  • SCE: Single Channel Element單通道元素。單通道元素基本上只由一個(gè)ICS組成。一個(gè)原始數(shù)據(jù)塊最可能由16個(gè)SCE組成。
  • CPE: Channel Pair Element 雙通道元素，由兩個(gè)可能共享邊信息的ICS和一些聯(lián)合立體聲編碼信息組成。一個(gè)原始數(shù)據(jù)塊最多可能由16個(gè)SCE組成。
  • CCE: Coupling Channel Element 藕合通道元素。代表一個(gè)塊的多通道聯(lián)合立體聲信息或者多語(yǔ)種程序的對(duì)話信息。
  • LFE: Low Frequency Element 低頻元素。包含了一個(gè)加強(qiáng)低采樣頻率的通道。
  • DSE: Data Stream Element 數(shù)據(jù)流元素，包含了一些并不屬于音頻的附加信息。
  • PCE: Program Config Element 程序配置元素。包含了聲道的配置信息。它可能出現(xiàn)在ADIF 頭部信息中。
  • FIL: Fill Element 填充元素。包含了一些擴(kuò)展信息。如SBR，動(dòng)態(tài)范圍控制信息等。

AudioQueue錄制PCM音頻

關(guān)于什么是AudioQueue及其介紹在此不再描述，請(qǐng)查閱蘋(píng)果官方文檔或譯:Audio Queue Services Programming Guide

設(shè)置錄制音頻格式

音頻格式：PCM ; 采樣率：48K ; 每幀數(shù)據(jù)通道數(shù)：1 ; 一幀數(shù)據(jù)中每個(gè)通道的樣本數(shù)據(jù)位數(shù): 16 ; 下面是具體代碼：

- (void)settingAudioFormat
{
    /*** setup audio sample rate , channels number, and format ID ***/
    memset(&dataFormat, 0, sizeof(dataFormat));
    UInt32 size = sizeof(dataFormat.mSampleRate);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate, &size, &dataFormat.mSampleRate);
    dataFormat.mSampleRate = kAudioSampleRate;
    size = sizeof(dataFormat.mChannelsPerFrame);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareInputNumberChannels, &size, &dataFormat.mChannelsPerFrame);
    dataFormat.mFormatID = kAudioFormatLinearPCM;
    dataFormat.mChannelsPerFrame = 1;
    dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    dataFormat.mBitsPerChannel = 16;
    dataFormat.mBytesPerPacket = dataFormat.mBytesPerFrame = (dataFormat.mBitsPerChannel / 8) * dataFormat.mChannelsPerFrame;
    dataFormat.mFramesPerPacket = kAudioFramesPerPacket; // AudioQueue collection pcm data , need to set as this
}

創(chuàng)建AudioQueue

extern OSStatus             
AudioQueueNewInput(                 const AudioStreamBasicDescription *inFormat,
                                    AudioQueueInputCallback         inCallbackProc,
                                    void * __nullable               inUserData,
                                    CFRunLoopRef __nullable         inCallbackRunLoop,
                                    CFStringRef __nullable          inCallbackRunLoopMode,
                                    UInt32                          inFlags,
                                    AudioQueueRef __nullable * __nonnull outAQ);
  • inFormt: 所錄制音頻的格式，是AudioStreamBasicDescription的實(shí)例。AudioStreamBasicDescription是對(duì)音頻格式的描述。
  • inCallbackProc : 是一個(gè)回調(diào)，當(dāng)一個(gè)buffer被填充完成時(shí)，會(huì)觸發(fā)這個(gè)回調(diào)。
  • inCallbackRunLoop：要調(diào)用inCallbackProc的事件循環(huán)。如果指定NULL，則在其中一個(gè)音頻隊(duì)列的內(nèi)部線程上調(diào)用回調(diào)。這個(gè)參數(shù)一般填寫(xiě)NULL
  • inCallbackRunLoopMode：為RunLoop模式，如果傳入NULL就相當(dāng)于kCFRunLoopCommonModes，一般這個(gè)參數(shù)也是填寫(xiě)NULL
  • inFlags : 保留字段，直接傳0
  • outAQ: 返回生成的AudioQueue實(shí)例,返回值用來(lái)判斷是否成功創(chuàng)建（OSStatus == noErr)

下面代碼時(shí)創(chuàng)建錄制的AudioQueue的代碼：

- (void)settingCallBackFunc
{
    /*** 設(shè)置錄音回調(diào)函數(shù) ***/
    OSStatus status = 0;
    //    int bufferByteSize = 0;
    UInt32 size = sizeof(dataFormat);
    status = AudioQueueNewInput(&dataFormat, inputAudioQueueBufferHandler, (__bridge void *)self, NULL, NULL, 0, &mQueue);
    if (status != noErr) {
        NSLog(@"AppRecordAudio,%s,AudioQueueNewInput failed status:%d ",__func__,(int)status);
    }
    
    for (int i = 0 ; i < kQueueBuffers; i++) {
        status = AudioQueueAllocateBuffer(mQueue, kAudioPCMTotalPacket * kAudioBytesPerPacket * dataFormat.mChannelsPerFrame, &mBuffers[i]);
        status = AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL);
    }
}

音頻錄制回調(diào)函數(shù)

在此回調(diào)中，我們直接把PCM數(shù)據(jù)寫(xiě)到沙盒中(只寫(xiě)前800幀，數(shù)目達(dá)到時(shí)會(huì)停止錄制)。

/*!
 @discussion
 AudioQueue 音頻錄制回調(diào)函數(shù)
 @param      inAQ
 回調(diào)函數(shù)的音頻隊(duì)列.
 @param      inBuffer
 是一個(gè)被音頻隊(duì)列填充新的音頻數(shù)據(jù)的音頻隊(duì)列緩沖區(qū)，它包含了回調(diào)函數(shù)寫(xiě)入文件所需要的新數(shù)據(jù).
 @param      inStartTime
 是緩沖區(qū)中的一采樣的參考時(shí)間
 @param      inNumberPacketDescriptions
 參數(shù)中包描述符（packet descriptions）的數(shù)量，如果你正在錄制一個(gè)VBR(可變比特率（variable bitrate））格式, 音頻隊(duì)列將會(huì)提供這個(gè)參數(shù)給你的回調(diào)函數(shù)，這個(gè)參數(shù)可以讓你傳遞給AudioFileWritePackets函數(shù). CBR (常量比特率（constant bitrate）) 格式不使用包描述符。對(duì)于CBR錄制，音頻隊(duì)列會(huì)設(shè)置這個(gè)參數(shù)并且將inPacketDescs這個(gè)參數(shù)設(shè)置為NULL
 
 */
static void inputAudioQueueBufferHandler(void * __nullable               inUserData,
                               AudioQueueRef                   inAQ,
                               AudioQueueBufferRef             inBuffer,
                               const AudioTimeStamp *          inStartTime,
                               UInt32                          inNumberPacketDescriptions,
                               const AudioStreamPacketDescription * __nullable inPacketDescs)
{
    if (!inUserData) {
        NSLog(@"AppRecordAudio,%s,inUserData is null",__func__);
        return;
    }
    
    NSLog(@"%s, audio length: %d",__func__,inBuffer->mAudioDataByteSize);
    static int createCount = 0;
    static FILE *fp_pcm = NULL;
    if (createCount == 0) {
        NSString *paths = [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) objectAtIndex:0];
        NSString *debugUrl = [paths stringByAppendingPathComponent:@"debug"] ;
        NSFileManager *fileManager = [NSFileManager defaultManager];
        [fileManager createDirectoryAtPath:debugUrl withIntermediateDirectories:YES attributes:nil error:nil];
        
        NSString *audioFile = [paths stringByAppendingPathComponent:@"debug/queue_pcm_48k.pcm"] ;
        fp_pcm = fopen([audioFile UTF8String], "wb++");
    }
    createCount++;
    
    MIAudioQueueRecord *miAQ = (__bridge MIAudioQueueRecord *)inUserData;
    if (createCount <= 800) {
        void *bufferData = inBuffer->mAudioData;
        UInt32 buffersize = inBuffer->mAudioDataByteSize;
        fwrite((uint8_t *)bufferData, 1, buffersize, fp_pcm);
    }else{
        fclose(fp_pcm);
        NSLog(@"AudioQueue, close PCM file ");
        [miAQ stopRecorder];
        createCount = 0;
    }
    
    if (miAQ.m_isRunning) {
        AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
    }
}

開(kāi)始錄制和停止錄制

開(kāi)始錄制：

- (void)startRecorder
{
    [self createAudioSession];
    [self settingAudioFormat];
    [self settingCallBackFunc];
    
    if (self.m_isRunning) {
        return;
    }

    /*** start audioQueue ***/
    OSStatus status = AudioQueueStart(mQueue, NULL);
    if (status != noErr) {
        NSLog(@"AppRecordAudio,%s,AudioQueueStart failed status:%d  ",__func__,(int)status);
    }
    self.m_isRunning = YES;
}

停止錄制：

- (void)stopRecorder
{
    if (!self.m_isRunning) {
        return;
    }
    self.m_isRunning = NO;
    
    if (mQueue) {
        OSStatus stopRes = AudioQueueStop(mQueue, true);
        
        if (stopRes == noErr) {
            for (int i = 0; i < kQueueBuffers; i++) {
                AudioQueueFreeBuffer(mQueue, mBuffers[i]);
            }
        }else{
            NSLog(@"AppRecordAudio,%s,stop AudioQueue failed.  ",__func__);
        }
        
        AudioQueueDispose(mQueue, true);
        mQueue = NULL;
    }
}

利用ffplay播放錄制的音頻

我們此時(shí)進(jìn)入到沙河目錄中先用ffplay命令播放一下錄制好的pcm音頻，后面我們會(huì)利用AudioQueue來(lái)播放pcm文件。

ffplay -f s16le -ar 48000 -ac 1 queue_pcm_48k.pcm
image

AudioQueue播放PCM文件

我們?cè)诒拘〗Y(jié)中，就播放上面剛剛錄制的pcm文件。

設(shè)置待播放的音頻格式

- (void)settingAudioFormat
{
    /*** setup audio sample rate , channels number, and format ID ***/
    memset(&dataFormat, 0, sizeof(dataFormat));
    UInt32 size = sizeof(dataFormat.mSampleRate);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate, &size, &dataFormat.mSampleRate);
    dataFormat.mSampleRate = kAudioSampleRate;
    size = sizeof(dataFormat.mChannelsPerFrame);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareInputNumberChannels, &size, &dataFormat.mChannelsPerFrame);
    dataFormat.mFormatID = kAudioFormatLinearPCM;
    dataFormat.mChannelsPerFrame = 1;
    dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    dataFormat.mBitsPerChannel = 16;
    dataFormat.mBytesPerPacket = dataFormat.mBytesPerFrame = (dataFormat.mBitsPerChannel / 8) * dataFormat.mChannelsPerFrame;
    dataFormat.mFramesPerPacket = kAudioFramesPerPacket; // AudioQueue collection pcm data , need to set as this
}

創(chuàng)建播放的AudioQueue并設(shè)置callback

- (void)settingCallBackFunc
{
    /*** 設(shè)置錄音回調(diào)函數(shù) ***/
    OSStatus status = 0;
    //    int bufferByteSize = 0;
    UInt32 size = sizeof(dataFormat);
    
    /*** 設(shè)置播放回調(diào)函數(shù) ***/
    status = AudioQueueNewOutput(&dataFormat,
                                 miAudioPlayCallBack,
                                 (__bridge void *)self,
                                 NULL,
                                 NULL,
                                 0,
                                 &mQueue);
    if (status != noErr) {
        NSLog(@"AppRecordAudio,%s, AudioQueueNewOutput failed status:%d",__func__,(int)status);
    }
    
    for (int i = 0 ; i < kQueueBuffers; i++) {
        status = AudioQueueAllocateBuffer(mQueue, kAudioPCMTotalPacket * kAudioBytesPerPacket * dataFormat.mChannelsPerFrame, &mBuffers[i]);
        status = AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL);
    }

}

從pcm文件中讀音頻數(shù)據(jù)

- (void)initPlayedFile
{
    NSString *paths = [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) objectAtIndex:0];
    NSString *audioFile = [paths stringByAppendingPathComponent:@"debug/queue_pcm_48k.pcm"] ;
    NSFileManager *manager = [NSFileManager defaultManager];
    NSLog(@"file exist = %d",[manager fileExistsAtPath:audioFile]);
    NSLog(@"file size = %lld",[[manager attributesOfItemAtPath:audioFile error:nil] fileSize]) ;
    file  = fopen([audioFile UTF8String], "r");
    if(file)
    {
        fseek(file, 0, SEEK_SET);
        pcmDataBuffer = malloc(1024);
    }
    else{
        NSLog(@"!!!!!!!!!!!!!!!!");
    }
    synlock = [[NSLock alloc] init];
}

把pcm數(shù)據(jù)送給AudioQueue Buffer播放

- (void)startPlay
{
    [self initPlayedFile];
    [self createAudioSession];
    [self settingAudioFormat];
    [self settingCallBackFunc];
    
    AudioQueueStart(mQueue, NULL);
    for (int i = 0; i < kQueueBuffers; i++) {
        [self readPCMAndPlay:mQueue buffer:mBuffers[i]];
    }
}

-(void)readPCMAndPlay:(AudioQueueRef)outQ buffer:(AudioQueueBufferRef)outQB
{
    [synlock lock];
    int readLength = fread(pcmDataBuffer, 1, 1024, file);//讀取文件
    NSLog(@"read raw data size = %d",readLength);
    outQB->mAudioDataByteSize = readLength;
    Byte *audiodata = (Byte *)outQB->mAudioData;
    for(int i=0;i<readLength;i++)
    {
        audiodata[i] = pcmDataBuffer[i];
    }
    /*
     將創(chuàng)建的buffer區(qū)添加到audioqueue里播放
     AudioQueueBufferRef用來(lái)緩存待播放的數(shù)據(jù)區(qū)，AudioQueueBufferRef有兩個(gè)比較重要的參數(shù)，AudioQueueBufferRef->mAudioDataByteSize用來(lái)指示數(shù)據(jù)區(qū)大小，AudioQueueBufferRef->mAudioData用來(lái)保存數(shù)據(jù)區(qū)
     */
    AudioQueueEnqueueBuffer(outQ, outQB, 0, NULL);
    [synlock unlock];
}

運(yùn)行

image

如上圖所是，點(diǎn)擊播放會(huì)播放上一次錄制的pcm音頻。

AudioQueue實(shí)時(shí)編碼PCM數(shù)據(jù)為AAC并保存到沙盒中

在本小結(jié)中，我們首先啟動(dòng)AudioQueue錄制pcm音頻，同時(shí)創(chuàng)建一個(gè)轉(zhuǎn)化器把PCM數(shù)據(jù)轉(zhuǎn)化成AAC，最后把AAC保存到沙盒中。流程大致如下：

image

采樣率等參數(shù)依然和前面錄制和播放的保持一致，都是48K

設(shè)置輸入輸出音頻編碼參數(shù)并創(chuàng)建轉(zhuǎn)化器

- (void)settingInputAudioFormat
{
    /*** setup audio sample rate , channels number, and format ID ***/
    memset(&inAudioStreamDes, 0, sizeof(inAudioStreamDes));
    UInt32 size = sizeof(inAudioStreamDes.mSampleRate);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate, &size, &inAudioStreamDes.mSampleRate);
    inAudioStreamDes.mSampleRate = kAudioSampleRate;
    size = sizeof(inAudioStreamDes.mChannelsPerFrame);
    AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareInputNumberChannels, &size, &inAudioStreamDes.mChannelsPerFrame);
    inAudioStreamDes.mFormatID = kAudioFormatLinearPCM;
    inAudioStreamDes.mChannelsPerFrame = 1;
    inAudioStreamDes.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    inAudioStreamDes.mBitsPerChannel = 16;
    inAudioStreamDes.mBytesPerPacket = inAudioStreamDes.mBytesPerFrame = (inAudioStreamDes.mBitsPerChannel / 8) * inAudioStreamDes.mChannelsPerFrame;
    inAudioStreamDes.mFramesPerPacket = kAudioFramesPerPacket; // AudioQueue collection pcm data , need to set as this
}

- (void)settingDestAudioStreamDescription
{
    outAudioStreamDes.mSampleRate = kAudioSampleRate;
    outAudioStreamDes.mFormatID = kAudioFormatMPEG4AAC;
    outAudioStreamDes.mBytesPerPacket = 0;
    outAudioStreamDes.mFramesPerPacket = 1024;
    outAudioStreamDes.mBytesPerFrame = 0;
    outAudioStreamDes.mChannelsPerFrame = 1;
    outAudioStreamDes.mBitsPerChannel = 0;
    outAudioStreamDes.mReserved = 0;
    AudioClassDescription *des = [self getAudioClassDescriptionWithType:kAudioFormatMPEG4AAC
                                                       fromManufacturer:kAppleSoftwareAudioCodecManufacturer];
    OSStatus status = AudioConverterNewSpecific(&inAudioStreamDes, &outAudioStreamDes, 1, des, &miAudioConvert);
    if (status != 0) {
        NSLog(@"create convert failed...\n");
    }
    
    UInt32 targetSize   = sizeof(outAudioStreamDes);
    UInt32 bitRate  =  64000;
    targetSize      = sizeof(bitRate);
    status          = AudioConverterSetProperty(miAudioConvert,
                                                kAudioConverterEncodeBitRate,
                                                targetSize, &bitRate);
    if (status != noErr) {
        NSLog(@"set bitrate error...");
        return;
    }
}

獲取編解碼器

/**
 *  獲取編解碼器
 *  @param type         編碼格式
 *  @param manufacturer 軟/硬編
 *  @return 指定編碼器
 */
- (AudioClassDescription *)getAudioClassDescriptionWithType:(UInt32)type
                                           fromManufacturer:(UInt32)manufacturer
{
    static AudioClassDescription desc;
    
    UInt32 encoderSpecifier = type;
    OSStatus st;
    
    UInt32 size;
    // 取得給定屬性的信息
    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];
    // 取得給定屬性的數(shù)據(jù)
    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;
}

填充PCM到緩沖區(qū)

/**
 *  填充PCM到緩沖區(qū)
 */
- (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;
}

AAC的DTS計(jì)算

具體可參考：

下面代碼時(shí)支持采樣率為48K，通道數(shù)為1的DTS計(jì)算：

- (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 = 3;  //48KHz
    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;
}

PCM轉(zhuǎn)AAC并把AAC寫(xiě)入到沙盒

static int initTime = 0;
- (void)encodePCMToAAC:(MIAudioQueueConvert *)convert
{
    if (initTime == 0) {
        initTime = 1;
        [self settingDestAudioStreamDescription];
    }
    OSStatus status;
    memset(_aacBuffer, 0, _aacBufferSize);
    
    AudioBufferList *bufferList             = (AudioBufferList *)malloc(sizeof(AudioBufferList));
    bufferList->mNumberBuffers              = 1;
    bufferList->mBuffers[0].mNumberChannels = outAudioStreamDes.mChannelsPerFrame;
    bufferList->mBuffers[0].mData           = _aacBuffer;
    bufferList->mBuffers[0].mDataByteSize   = (int)_aacBufferSize;
    
    AudioStreamPacketDescription outputPacketDescriptions;
    UInt32 inNumPackets = 1;
    status = AudioConverterFillComplexBuffer(miAudioConvert,
                                             pcmEncodeConverterInputCallback,
                                             (__bridge void *)(self),//inBuffer->mAudioData,
                                             &inNumPackets,
                                             bufferList,
                                             &outputPacketDescriptions);
    
    if (status == noErr) {
        NSData *aacData = [NSData dataWithBytes:bufferList->mBuffers[0].mData length:bufferList->mBuffers[0].mDataByteSize];
        static int createCount = 0;
        static FILE *fp_aac = NULL;
        if (createCount == 0) {
            NSString *paths = [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) objectAtIndex:0];
            NSString *debugUrl = [paths stringByAppendingPathComponent:@"debug"] ;
            NSFileManager *fileManager = [NSFileManager defaultManager];
            [fileManager createDirectoryAtPath:debugUrl withIntermediateDirectories:YES attributes:nil error:nil];
            
            NSString *audioFile = [paths stringByAppendingPathComponent:@"debug/queue_aac_48k.aac"] ;
            fp_aac = fopen([audioFile UTF8String], "wb++");
        }
        createCount++;
        if (createCount <= 800) {
            NSData *rawAAC = [NSData dataWithBytes:bufferList->mBuffers[0].mData length:bufferList->mBuffers[0].mDataByteSize];
            NSData *adtsHeader = [self adtsDataForPacketLength:rawAAC.length];
            NSMutableData *fullData = [NSMutableData dataWithData:adtsHeader];
            [fullData appendData:rawAAC];
            
            void * bufferData = fullData.bytes;
            int buffersize = fullData.length;
            
            fwrite((uint8_t *)bufferData, 1, buffersize, fp_aac);
        }else{
            fclose(fp_aac);
            NSLog(@"AudioQueue, close aac file ");
            [self stopRecorder];
            createCount = 0;
        }
    }
}

利用ffplay播放

ffplay命令：

ffplay -ar 48000 queue_aac_48k.aac
image
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