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Android 音视频04 --- SoundTouch

Android 音视频04 --- SoundTouch

作者: 沪漂意哥哥 | 来源:发表于2022-04-19 00:23 被阅读0次

一、 SoundTouch详解

是一个用C++编写的开源的音频处理库，可以改变音频文件或实时音频流的节拍(Tempo)、音调(Pitch)、
回放率(Playback Rates)，还支持估算音轨的稳定节拍率(BPM rate)。ST的3个效果互相独立，也可以一
起使用。这些效果通过采样率转换、时间拉伸结合实现。

1.1Tempo节拍 ：

通过拉伸时间，改变声音的播放速率而不影响音调。

1.2Playback Rate回放率 :

以不同的转率播放唱片（DJ打碟？），通过采样率转换实现。

1.3Pitch音调 ：

在保持节拍不变的前提下改变声音的音调，结合采样率转换+时间拉伸实现。如：增
高音调的处理过程是：将原音频拉伸时长，再通过采样率转换，同时减少时长与增高音调变为原时
长。

二、处理对象

ST处理的对象是PCM（Pulse Code Modulation，脉冲编码调制），.wav文件中主要是这种格式，因此
ST的示例都是处理wav音频。mp3等格式经过了压缩，需转换为PCM后再用ST处理。

三、主要特性

3.1易于实现：

ST为所有支持gcc编译器或者visual Studio的处理器或操作系统进行了编译，支持Windows、Mac OS、Linux、Android、Apple iOS等。

3.2完全开源：

ST库与示例工程完全开源可下载

3.3容易使用：

编程接口使用单一的C++类

3.4支持16位整型或32位浮点型的单声道、立体声、多通道的音频格式
3.5可实现实时音频流处理：

输入/输出延迟约为100ms；
实时处理44.1kHz/16bit的立体声，需要133Mhz英特尔奔腾处理器或更好；

四、调用接口与参数

官网提供了ST的可执行程序、C++源码、说明文档、不同操作系统的示例工程，几个重要链接：

4.1采样：

setChannels(int) 设置声道，1 = mono单声道, 2 = stereo立体声
setSampleRate(uint) 设置采样率

4.2速率：

setRate(double) 指定播放速率，原始值为1.0，大快小慢
setTempo(double) 指定节拍，原始值为1.0，大快小慢
setRateChange(double) 、 setTempoChange(double) 在原速1.0基础上，按百分比做增量，取值(-50 .. +100 %)

4.3音调：

setPitch(double) 指定音调值，原始值为1.0
setPitchOctaves(double) 在原音调基础上以八度音为单位进行调整，取值为[-1.00,+1.00]
setPitchSemiTones(int) 在原音调基础上以半音为单位进行调整，取值为[-12,+12]

五、SoundTouch实时处理音频流

ST对音频的处理是输入函数putSamples()与输出函数receiveSamples()。实时处理音频流的思路就是，
循环读取音频数据段，放入ST进行输出，输出处理后的数据段用于播放。

六、Cmakelist.txt

cmake_minimum_required(VERSION 3.4.1)

include_directories(include)

include_directories(soundtouch/include)
include_directories(soundtouch/SoundTouch)

#添加系统环境变量
set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -L${CMAKE_SOURCE_DIR}/libs/${ANDROID_ABI}")

aux_source_directory(. SOURCE)
aux_source_directory(./soundtouch/SoundTouch SOURCE1)
add_library(
        native-lib
        SHARED
        ${SOURCE}
        ${SOURCE1})

find_library(
        log-lib
        log)

target_link_libraries(
        native-lib
        avcodec
        avdevice
        avfilter
        avformat
        avutil
        swresample
        swscale
        ${log-lib}
        OpenSLES
        android)

七、LYAudio.cpp

LYAudio::LYAudio(LYPlaystatus *playstatus, int sample_rate, LYCallJava *callJava) {
    this->playstatus = playstatus;
    this->sample_rate = sample_rate;
    queue = new LYQueue(playstatus);
    buffer = (uint8_t *) av_malloc(sample_rate * 2 * 2);
    this->callJava = callJava;
    sampleBuffer= static_cast<SAMPLETYPE *>(malloc(sample_rate * 2 * 2));
    soundTouch = new SoundTouch();
    soundTouch->setSampleRate(sample_rate);
    soundTouch->setChannels(2);
    soundTouch->setTempo(speed);
    soundTouch->setPitch(pitch);
}


LYAudio::~LYAudio() {

}

void *decodPlay(void *data) {
    LYAudio *wlAudio = (LYAudio *) data;
    wlAudio->initOpenSLES();
    pthread_exit(&wlAudio->thread_play);
}
void LYAudio::play() {
    pthread_create(&thread_play, NULL, decodPlay, this);
}

void pcmBufferCallBack(SLAndroidSimpleBufferQueueItf bf, void * context) {
    LYAudio *wlAudio = (LYAudio *) context;
    if(wlAudio != NULL) {
//        int buffersize = wlAudio->resampleAudio();
        int buffersize = wlAudio->getSoundTouchData();
        if(buffersize > 0) {
            wlAudio->clock+=buffersize/ ((double)(wlAudio->sample_rate * 2 * 2));
            if(wlAudio->clock - wlAudio->last_time >= 0.25){
                wlAudio->last_time = wlAudio->clock;
                wlAudio->callJava->onCallTimeInfo(CHILD_THREAD,wlAudio->clock,wlAudio->duration);
            }
            (* wlAudio-> pcmBufferQueue)->Enqueue( wlAudio->pcmBufferQueue,(char *) wlAudio-> sampleBuffer, buffersize*2*2);
        }
    }
}

void LYAudio::initOpenSLES() {
    LOGE("initOpenSLES begin");
    SLresult result;
    result = slCreateEngine(&engineObject, 0, 0, 0, 0, 0);
    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);

    //第二步，创建混音器
    const SLInterfaceID mids[1] = {SL_IID_ENVIRONMENTALREVERB};
    const SLboolean mreq[1] = {SL_BOOLEAN_FALSE};
    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, mids, mreq);
    (void)result;
    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
    (void)result;
    result = (*outputMixObject)->GetInterface(outputMixObject, SL_IID_ENVIRONMENTALREVERB, &outputMixEnvironmentalReverb);
    if (SL_RESULT_SUCCESS == result) {
        result = (*outputMixEnvironmentalReverb)->SetEnvironmentalReverbProperties(
                outputMixEnvironmentalReverb, &reverbSettings);
        (void)result;
    }
    SLDataLocator_OutputMix outputMix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
    SLDataSink audioSnk = {&outputMix, 0};


    // 第三步，配置PCM格式信息
    SLDataLocator_AndroidSimpleBufferQueue android_queue={SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,2};

    SLDataFormat_PCM pcm={
            SL_DATAFORMAT_PCM,//播放pcm格式的数据
            2,//2个声道（立体声）
            static_cast<SLuint32>(getCurrentSampleRateForOpensles(sample_rate)),//44100hz的频率
            SL_PCMSAMPLEFORMAT_FIXED_16,//位数 16位
            SL_PCMSAMPLEFORMAT_FIXED_16,//和位数一致就行
            SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,//立体声（前左前右）
            SL_BYTEORDER_LITTLEENDIAN//结束标志
    };
    SLDataSource slDataSource = {&android_queue, &pcm};


    const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE,SL_IID_VOLUME,SL_IID_MUTESOLO};
    const SLboolean req[3] = {SL_BOOLEAN_TRUE,SL_BOOLEAN_TRUE,SL_BOOLEAN_TRUE};

    (*engineEngine)->CreateAudioPlayer(engineEngine, &pcmPlayerObject, &slDataSource, &audioSnk, 2, ids, req);
    //初始化播放器
    (*pcmPlayerObject)->Realize(pcmPlayerObject, SL_BOOLEAN_FALSE);

//    得到接口后调用  获取Player接口
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject, SL_IID_PLAY, &pcmPlayerPlay);
    if(pcmPlayerPlay == NULL) {
        LOGE("pcmPlayerPlay == NULL");
    }
//    获取声道操作接口
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject, SL_IID_MUTESOLO, &pcmMutePlay);
    if(pcmMutePlay == NULL) {
        LOGE("pcmMutePlay == NULL");
    }
//   拿控制  播放暂停恢复的句柄
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject,SL_IID_VOLUME,&pcmVolumePlay);
    if(pcmVolumePlay == NULL) {
        LOGE("pcmVolumePlay == NULL");
    }
//    注册回调缓冲区 获取缓冲队列接口
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject, SL_IID_BUFFERQUEUE, &pcmBufferQueue);
    if(pcmBufferQueue == NULL) {
        LOGE("pcmBufferQueue == NULL");
    }
    //缓冲接口回调
    (*pcmBufferQueue)->RegisterCallback(pcmBufferQueue, pcmBufferCallBack, this);
//    获取播放状态接口
    (*pcmPlayerPlay)->SetPlayState(pcmPlayerPlay, SL_PLAYSTATE_PLAYING);
    pcmBufferCallBack(pcmBufferQueue, this);
    LOGE("initOpenSLES end");
}

int LYAudio::getCurrentSampleRateForOpensles(int sample_rate) {
    int rate = 0;
    switch (sample_rate)
    {
        case 8000:
            rate = SL_SAMPLINGRATE_8;
            break;
        case 11025:
            rate = SL_SAMPLINGRATE_11_025;
            break;
        case 12000:
            rate = SL_SAMPLINGRATE_12;
            break;
        case 16000:
            rate = SL_SAMPLINGRATE_16;
            break;
        case 22050:
            rate = SL_SAMPLINGRATE_22_05;
            break;
        case 24000:
            rate = SL_SAMPLINGRATE_24;
            break;
        case 32000:
            rate = SL_SAMPLINGRATE_32;
            break;
        case 44100:
            rate = SL_SAMPLINGRATE_44_1;
            break;
        case 48000:
            rate = SL_SAMPLINGRATE_48;
            break;
        case 64000:
            rate = SL_SAMPLINGRATE_64;
            break;
        case 88200:
            rate = SL_SAMPLINGRATE_88_2;
            break;
        case 96000:
            rate = SL_SAMPLINGRATE_96;
            break;
        case 192000:
            rate = SL_SAMPLINGRATE_192;
            break;
        default:
            rate =  SL_SAMPLINGRATE_44_1;
    }
    return rate;
}

int LYAudio::resampleAudio(void **pcmbuf) {
    while(playstatus != NULL && !playstatus->exit) {
        avPacket = av_packet_alloc();
        if(queue->getAvpacket(avPacket) != 0) {
            av_packet_free(&avPacket);
            av_free(avPacket);
            avPacket = NULL;
            continue;
        }
        ret = avcodec_send_packet(avCodecContext, avPacket);
        if(ret != 0) {
            av_packet_free(&avPacket);
            av_free(avPacket);
            avPacket = NULL;
            continue;
        }
        avFrame = av_frame_alloc();
        ret = avcodec_receive_frame(avCodecContext, avFrame);
        if(ret == 0) {
            if(avFrame->channels && avFrame->channel_layout == 0) {
                avFrame->channel_layout = av_get_default_channel_layout(avFrame->channels);
            } else if(avFrame->channels == 0 && avFrame->channel_layout > 0){
                avFrame->channels = av_get_channel_layout_nb_channels(avFrame->channel_layout);
            }

            SwrContext *swr_ctx;
            swr_ctx = swr_alloc_set_opts(
                    NULL,
                    AV_CH_LAYOUT_STEREO,
                    AV_SAMPLE_FMT_S16,
                    avFrame->sample_rate,
                    avFrame->channel_layout,
                    (AVSampleFormat) avFrame->format,
                    avFrame->sample_rate,
                    NULL, NULL
            );
            if(!swr_ctx || swr_init(swr_ctx) <0) {
                av_packet_free(&avPacket);
                av_free(avPacket);
                avPacket = NULL;
                av_frame_free(&avFrame);
                av_free(avFrame);
                avFrame = NULL;
                swr_free(&swr_ctx);
                continue;
            }

            nb = swr_convert(
                    swr_ctx,
                    &buffer,
                    avFrame->nb_samples,
                    (const uint8_t **) avFrame->data,
                    avFrame->nb_samples);

            int out_channels = av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);
            data_size = nb * out_channels * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
            now_time= avFrame->pts * av_q2d(time_base);
            if(now_time < clock)  {
                now_time = clock;
            }
            clock = now_time;
            *pcmbuf = buffer;
            av_packet_free(&avPacket);
            av_free(avPacket);
            avPacket = NULL;
            av_frame_free(&avFrame);
            av_free(avFrame);
            avFrame = NULL;
            swr_free(&swr_ctx);
            break;
        }  else{
            av_packet_free(&avPacket);
            av_free(avPacket);
            avPacket = NULL;
            av_frame_free(&avFrame);
            av_free(avFrame);
            avFrame = NULL;
            continue;
        }
    }
    return data_size;
}

void LYAudio::pause() {
    if(pcmPlayerPlay != NULL) {
        (*pcmPlayerPlay)->SetPlayState(pcmPlayerPlay, SL_PLAYSTATE_PAUSED);
    }
}

void LYAudio::resume() {
    if(pcmPlayerPlay != NULL) {
        (*pcmPlayerPlay)->SetPlayState(pcmPlayerPlay, SL_PLAYSTATE_PLAYING);
    }
}

void LYAudio::setMute(jint mute) {

    if(pcmMutePlay == NULL) {
        return;
    }
    this->mute = mute;
    if(mute == 0){
        //right
        LOGE("0000");
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 1, false);
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 0, true);
    } else if(mute == 1) {
        LOGE("11111");
        //left
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 1, true);
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 0, false);
    } else if(mute == 2) {
        LOGE("2222");
        //center
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 1, false);
        (*pcmMutePlay)->SetChannelMute(pcmMutePlay, 0, false);
    }
}

void LYAudio::setVolume(int percent) {
    LOGE("setVolume %d", percent);
    if(pcmVolumePlay != NULL) {
        if (percent > 30) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -20);
        } else if (percent > 25) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -22);
        } else if (percent > 20) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -25);
        } else if (percent > 15) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -28);
        } else if (percent > 10) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -30);
        } else if (percent > 5) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -34);
        } else if (percent > 3) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -37);
        } else if (percent > 0) {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -40);
        } else {
            (*pcmVolumePlay)->SetVolumeLevel(pcmVolumePlay, (100 - percent) * -100);
        }
    }
}

void LYAudio::setPitch(float pitch) {
    this->pitch = pitch;
    if(soundTouch != NULL)  {
        soundTouch->setPitch(pitch);
    }
}

void LYAudio::setSpeed(float speed) {
    this->speed = speed;
    if (soundTouch != NULL) {
        soundTouch->setTempo(speed);
    }
}

int LYAudio::getSoundTouchData() {
    while(playstatus != NULL && !playstatus->exit){
        out_buffer = NULL;
        if(finished){
            finished = false;
            data_size = this->resampleAudio(reinterpret_cast<void **>(&out_buffer));
            if (data_size > 0) {
                for(int i = 0; i < data_size / 2 + 1; i++){
                    sampleBuffer[i]=(out_buffer[i * 2] | ((out_buffer[i * 2 + 1]) << 8));
                }
                soundTouch->putSamples(sampleBuffer, nb);
                num=soundTouch->receiveSamples(sampleBuffer, data_size / 4);
            } else{
                soundTouch->flush();
            }
        }

        if (num == 0) {
            finished = true;
            continue;
        } else{
            if(out_buffer == NULL){
                num=soundTouch->receiveSamples(sampleBuffer, data_size / 4);
                if(num == 0)  {
                    finished = true;
                    continue;
                }
            }
            return num;
        }
    }
    return 0;
}

八、代码地址

https://gitee.com/luisliuyi/android-ffmpeg-opensl-southtouch.git

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