PortAudio+webrtc+lame实现采集降噪增益mp3

一、使用PortAudio采集
首先你应该下载这个库编出动态库放好头文件和链接库lib，这些都不是重点不赘述。

//定义pa的sample类型为int16,这个可以配合webrtc模块
#define PA_SAMPLE_TYPE paInt16      //采用双字节,一个sample=sizeof(int)=2字节
#define SAMPLE_RATE  (32000)        //代表1秒有32000个sample
#define FRAMES_PER_BUFFER (6400)    //代表回调一次提供6400个sample
//说明:input是从设备读到的数据,字节长度是frameCount*sizeof(SAMPLE)
//如果要echo,需要在初始化的时候指定好设备,把input拷到output即可.
static int PACallback(
        const void *input, void *output,
        unsigned long frameCount,
        const PaStreamCallbackTimeInfo* timeInfo,
        PaStreamCallbackFlags statusFlags,
        void *userData )
{
    Q_UNUSED(output);
    Q_UNUSED(timeInfo);
    Q_UNUSED(statusFlags);
    //进一步处理数据
    WcRecordMedia2 *media=(WcRecordMedia2*)userData;
    bool rt = media->tHandle(input,frameCount);
    return rt?paContinue:paComplete;
}

    PaError err;
    err = Pa_Initialize();
    if( err != paNoError ) goto error;
    PaDeviceIndex didx = Pa_GetDefaultInputDevice();
    if(didx == paNoDevice){
        //没有录音设备
        return false;
    }
    PaStreamParameters inputDev;
    inputDev.device = Pa_GetDefaultInputDevice();
    inputDev.channelCount = 1;
    inputDev.sampleFormat = PA_SAMPLE_TYPE;
    inputDev.suggestedLatency = 1;
    inputDev.hostApiSpecificStreamInfo = NULL;

    PaStreamParameters outputDev;
    //以下写paNoDevice会导致Pa_OpenStream失败,所以如果没喇叭输出设备,录音也玩不了啦?!
    outputDev.device = Pa_GetDefaultOutputDevice();
    outputDev.channelCount = 1;
    outputDev.sampleFormat = PA_SAMPLE_TYPE;
    outputDev.suggestedLatency = 1;
    outputDev.hostApiSpecificStreamInfo = NULL;
    PaStream* paStream = NULL;
    err = Pa_OpenStream(
                &paStream,
                &inputDev,
                &outputDev,
                SAMPLE_RATE,
                FRAMES_PER_BUFFER, /* frames per buffer */
                paDitherOff,    /* paDitherOff, // flags */
                PACallback,
                this);   //userdata,这里是我的类指针.
    if( err != paNoError ) goto error;
    err = Pa_StartStream( paStream );
    if( err != paNoError ) goto error;
    //如果是demo,可以紧接着睡一分钟.
    for(int i=0;i<6000;i++){
        Pa_Sleep(10);
    }
    //...结束的时候
    PaError err;
    if(paStream_){
        err = Pa_CloseStream( paStream );
        qDebug()<<"Pa_CloseStream:"<<Pa_GetErrorText(err);
        paStream = NULL;
    }
    Pa_Terminate();

有用经验：①webrtc只支持SAMPLE_RATE 为8000,16000,32000,常见的44100罢工!
②FRAMES_PER_BUFFER 可以比较自由,但是给webrtc处理的frameCount有限制..咳咳..请坐稳你的小板凳: webrtc提供的Ns(降噪),每次给的frameCount最多是480,因为拆分高低频后,count减半不能超过240; AGC(增益)只支持160或者320; 所以我选择了320的20倍,分20次把6400个sample给进去.

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二、webrtc处理
这部分代码是从另一个帖子下载抄回来的，作者从webrtc代码里抠出来需要的部分，神！原贴在这里，遥谢一下原始作者！
http://www.jianshu.com/p/77a363960711
以下代码是我根据原贴封装的C++类

//webrtc的音频处理模块,支持降噪和增益.
//只接受int16的sample,其他的..谁知道能不能支持呢!
class WebRtcAudioModule
{
public:
    WebRtcAudioModule()
        : nsInst_(NULL)
        , agcHandle_(NULL)
        , micLevel_(0)
    {
        memset(internalBuf,0,sizeof(int)*24);
        filter_state1 = internalBuf + 0;
        filter_state2 = internalBuf + 6;
        synthesis_state1 = internalBuf + 12;
        synthesis_state2 = internalBuf + 18;

        memset(handlerBuf,0,sizeof(short)*240*4);
        //以下4个buf长度只需要单次处理frame的一半长度.(单位short2字节)
        shInL = handlerBuf + 0;
        shInH = handlerBuf + 240;
        shOutL = handlerBuf + 240*2;
        shOutH = handlerBuf + 240*3;
    }

    ~WebRtcAudioModule(){
        if(nsInst_) WebRtcNsx_Free(nsInst_);
        if(agcHandle_) WebRtcAgc_Free(agcHandle_);
    }

    bool initNs(uint32_t sampleRate)
    {
        int ret = WebRtcNsx_Create(&nsInst_);
        if(ret != 0) return false;
        ret = WebRtcNsx_Init(nsInst_,sampleRate);
        if(ret != 0) return false;
        ret = WebRtcNsx_set_policy(nsInst_,1);
        if(ret != 0) return false;
        return true;
    }

    bool initAgc(uint32_t sampleRate)
    {
        int ret = WebRtcAgc_Create(&agcHandle_);
        if(ret != 0) return false;
        int minLevel = 0;
        int maxLevel = 255;
        int agcMode  = kAgcModeFixedDigital;
        ret = WebRtcAgc_Init(agcHandle_, minLevel, maxLevel, agcMode, sampleRate);
        if(ret != 0) return false;
        WebRtcAgc_config_t agcConfig;
        agcConfig.compressionGaindB = 20;
        agcConfig.limiterEnable     = 1;
        agcConfig.targetLevelDbfs   = 3;
        ret = WebRtcAgc_set_config(agcHandle_, agcConfig);
        if(ret != 0) return false;
        return true;
    }

    //处理数据. data里面的有效长度是frameCount*sizeof(short). 也就是为什么只支持int16的sample
    bool handle(short* data, unsigned long frameCount)
    {       //记住,一个frame目前是2字节!
        int ret = 0;
        Q_ASSERT(frameCount%320==0); //agc操作要求frame必须是160, 320.
        Q_ASSERT(nsInst_ || agcHandle_);
        if(frameCount%320!=0) return false;
        for(unsigned i=0;i<frameCount;i+=320)
        {
            //分拆高频低频.这个函数一定会成功,否则崩掉.
            WebRtcSpl_AnalysisQMF(data+i,320,shInL,shInH,
                                  filter_state1,filter_state2);
            if(nsInst_){
                ret = WebRtcNsx_Process(nsInst_, shInL, shInH, shOutL, shOutH);
                if(ret != 0) break;
            }
            if(agcHandle_){
                if(nsInst_){
                    qSwap(shInL,shOutL);
                    qSwap(shInH,shOutH);
                }
                uint8_t saturationWarning;
                int inMicLevel = micLevel_;
                ret = WebRtcAgc_Process(agcHandle_, shInL, shInH, 160,
                                        shOutL ,shOutH, inMicLevel,
                                        &micLevel_, 0, &saturationWarning);
                if(ret != 0) break;
            }

            WebRtcSpl_SynthesisQMF(shOutL,shOutH, 160, data+i, synthesis_state1,synthesis_state2);
        }
        return (ret==0);
    }

private:
    NsxHandle *nsInst_;
    int internalBuf[24];
    int *filter_state1;//[6];
    int *filter_state2;//[6];
    int *synthesis_state1;//[6];
    int *synthesis_state2;//[6];
    short handlerBuf[240*4]; //webrtc降噪函数限制最大240个short
    short *shInL,*shInH,*shOutL,*shOutH;

    void *agcHandle_;
    int micLevel_;  //micLevel输入换输出,下一个处理再作为输入.
};

有用经验：①我这里是先降噪后增益,噪音降低后增益把原声和降低的噪音又都放大了,效果是降噪不太明显,主声增大比较显著.
②如果想要降噪–增益–再降噪, 你需要多初始化一套NsxHandle *nsInst2_, 然后照着样子调用,否则不行.
③WebRtcNsx_Process和WebRtcNs_Process听不出差别,但是有x是不是要更牛一点呢.
④webrtc这部分代码免费取用http://download.csdn.net/detail/jinzeyu_cn/9894495
三、lame编码mp3

//sdk要求lame的buffer最少要7200用于lame_encode_flush
#define MIN_LAME_BUFFERSIZE (7200)
class LameMp3Encoder
{
public:
    LameMp3Encoder()
        : flags_(NULL)
        , mp3buf(NULL)
        , fp_(NULL)
    {}

    ~LameMp3Encoder()
    {
        this->finish();
        if(flags_) lame_close(flags_);
        if(mp3buf) JFree(mp3buf);
        if(fp_) fclose(fp_);
    }

    bool init(const char* filePath, uint32_t sampleRate, int bufferSize=FRAMES_PER_BUFFER*2)
    {
        fp_ = fopen(filePath,"wb+");
        if(fp_ == NULL) return false;

        bufferSize_ = qMax(bufferSize,MIN_LAME_BUFFERSIZE);
        mp3buf = (unsigned char*)JMalloc(bufferSize_);

        flags_ = lame_init();
        if(flags_==NULL) return false;
        int ret = 0;
        lame_set_in_samplerate(flags_,sampleRate);
        lame_set_num_channels(flags_,1);
        lame_set_VBR(flags_,vbr_default); //vbr_abr);
        //关闭tag自动写入,因为lame_mp3_tags_fid函数崩溃.
        //参考 http://mp3-encoding.31853.n2.nabble.com/Re-lame-mp3-tags-fid-and-file-access-callbacks-td34000.html
        lame_set_write_id3tag_automatic(flags_,0);
        lame_set_brate(flags_,32);   //好像是影响最低比特率.
        lame_set_mode(flags_,MONO);  //单声道即可,源都是单声道.
        lame_set_quality(flags_,2);  //最佳效果,webrtc的效果占主导,其实听不出来差别.
        ret = lame_init_params(flags_);
        if(ret < 0) return false;
        ret = lame_get_id3v2_tag(flags_,mp3buf,bufferSize_);
        if(ret > 0){
            fwrite(mp3buf,1,ret,fp_);
        }
        tagPos_ = ftell(fp_);  //记录位置,结束前要回头写.
        return true;
    }

    bool handle(short* data, unsigned long frameCount)
    {
        if(fp_ == NULL) return false;
        Q_ASSERT(frameCount*2 <= bufferSize_);
        //int mp3bytes  = lame_encode_buffer_interleaved(flags_,data,frameCount,mp3buf,bufferSize_);
        int mp3bytes = lame_encode_buffer(flags_,data,data,frameCount,mp3buf,bufferSize_);
        if(mp3bytes < 0) return false;
        fwrite(mp3buf,1,mp3bytes,fp_);
        return true;
    }

    void finish()
    {
        if(flags_==NULL || bufferSize_<7200 || fp_==NULL) return;
        int mp3bytes = lame_encode_flush(flags_,mp3buf,bufferSize_);
        if(mp3bytes>0){
            fwrite(mp3buf,1,mp3bytes,fp_);
        }
        //lame_mp3_tags_fid(flags_,fp_);  此函数崩溃,看堆栈在fseek.使用手动tag写入解决.
        mp3bytes = lame_get_id3v1_tag(flags_,mp3buf,bufferSize_);
        if(mp3bytes>0){
            fwrite(mp3buf,1,mp3bytes,fp_);
        }
        mp3bytes = lame_get_lametag_frame(flags_,mp3buf,bufferSize_);
        if(mp3bytes>0){
            fseek(fp_,tagPos_,SEEK_SET);
            fwrite(mp3buf,1,mp3bytes,fp_);
        }
        fclose(fp_);
        fp_ = NULL;
    }

private:
    lame_global_flags* flags_;
    unsigned bufferSize_;
    unsigned char* mp3buf;
    FILE* fp_;
    int tagPos_;
};

有用经验：如果不用我的代码，mp3文件编完后很容易出现时长偏差，播放器和文件夹里看的时间不一致，那是因为没给mp3写好tag文件头。简单的方法是结束的时候调用lame_mp3_tags_fid，但是我一次也没成功，后找到老外一个帖子，手工添加tag完美解决，就是略麻烦一点，参见以上代码和url

最后

如果您和我有一样的需求,以上的经验足以让您节省至少一天的工时.算一下您月薪假如是12000,工作日22天,为您节省500+的人民币,真实在!!
当然为了我的工作,我将PortAudio的调用也C++类化,并且支持录音设备的热插拔,用空白语音填充无设备的情况.这一部分的有用经验：一旦Pa_Initialize();成功后Pa_GetDefaultInputDevice()返回值不会变,无法发现设备插入、移除或变更哦。手头没代码，先卖个关子吧！