音视频系列--OpenSL ES基础用法总结

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一、前置

OpenSL ES全称为Open Sound Library for Embedded Systems，及嵌入式音频加速标准。OpenSL ES是无授权费、跨平台、针对嵌入式系统封精心优化的硬件音频加速API。它为嵌入式移动多媒体设备上的本地应用程序开发提供了标准化、高性能、低响应时间的音频功能实现方法，同时还实现了软/硬件音频性能的直接跨平台部署，降低了执行难度。

在android中，High Level Audio Libs是音频Java层API输入输出，属于高级API，相对来说，OpenSL ES则是比较底层级的API，属于C语言API。这里记录下用法

二、OpenSL ES引入

Android的OpenSL ES库是在NDK的platforms文件夹对应android平台相应cpu类型里面，如：
在这里插入图片描述

CmakeList中引入

在这里插入图片描述

三、开发流程

3.1、OpenSL ES的开发流程主要有如下6个步骤

1、创建接口对象
2、设置混音器
3、创建播放器（录音器）
4、设置缓冲队列和回调函数
5、设置播放状态
6、启动回调函数
其中4和6是播放PCM等数据格式的音频是需要用到的。

四、播放PCM文件

4.1、 创建播放器和混音器


SLObjectItf engineObject = NULL;//用SLObjectItf声明引擎接口对象
SLEngineItf engineEngine = NULL;//声明具体的引擎对象实例

void createEngine()
{
    SLresult result;//返回结果
    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);//第一步创建引擎
    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);//实现（Realize）engineObject接口对象
    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);//通过engineObject的GetInterface方法初始化engineEngine
}

//第一步，创建引擎
    createEngine();
 
    //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, NULL};

4.2、设置pcm格式的频率位数等信息并创建播放器

// 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）
            SL_SAMPLINGRATE_44_1,//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_EFFECTSEND, SL_IID_VOLUME};
        const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
        
        
    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &pcmPlayerObject, &slDataSource, &audioSnk, 3, 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);

4.3、设置缓冲队列和回调函数

//    注册回调缓冲区 获取缓冲队列接口
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject, SL_IID_BUFFERQUEUE, &pcmBufferQueue);
    //缓冲接口回调
    (*pcmBufferQueue)->RegisterCallback(pcmBufferQueue, pcmBufferCallBack, NULL);
    

//回调函数
//喇叭没数据时候会回调这个函数，把数据写入到opensl es
void pcmBufferCallBack(SLAndroidSimpleBufferQueueItf bf, void * context)
{
    Audio *wlAudio = (Audio *) context;
    if(wlAudio != NULL)
    {
        int buffersize = wlAudio->resampleAudio();
        if(buffersize > 0)
        {
            (* wlAudio-> pcmBufferQueue)->Enqueue( wlAudio->pcmBufferQueue, (char *) wlAudio-> buffer, buffersize);
        }
    }
}

4.4、设置播放状态并手动开始调用回调函数

pcmBufferCallBack(pcmBufferQueue, NULL);

4.5、音频重采样

FFmpeg从网络解码的音频格式和OpenSL ES设置的音频播放格式是不一样的，这里有个重采样的过程，可以看示例程序的resampleAudio方法

五、示例代码

OpenSL ES代码基本是固定的，写好基本不用修改

#include "Audio.h"

Audio::Audio(Playstatus *playstatus, int sample_rate) {
    this->playstatus = playstatus;
    this->sample_rate = sample_rate;
    queue = new Queue(playstatus);
    buffer = (uint8_t *) av_malloc(sample_rate * 2 * 2);
}

Audio::~Audio() {

}

void *decodPlay(void *data)
{
    Audio *wlAudio = (Audio *) data;

    wlAudio->initOpenSLES();

    pthread_exit(&wlAudio->thread_play);
}

void Audio::play() {

    pthread_create(&thread_play, NULL, decodPlay, this);

}

//重采样
int Audio::resampleAudio() {

    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;
            }

            int 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);

            if(LOG_DEBUG)
            {
                LOGE("data_size is %d", data_size);
            }
            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;
}

//喇叭没数据时候会回调这个函数，把数据写入到opensl es
void pcmBufferCallBack(SLAndroidSimpleBufferQueueItf bf, void * context)
{
    Audio *wlAudio = (Audio *) context;
    if(wlAudio != NULL)
    {
        int buffersize = wlAudio->resampleAudio();
        if(buffersize > 0)
        {
            (* wlAudio-> pcmBufferQueue)->Enqueue( wlAudio->pcmBufferQueue, (char *) wlAudio-> buffer, buffersize);
        }
    }
}

void Audio::initOpenSLES() {

    //第一步 创建一个引擎接口对象
    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，通过获取的引擎接口对象去创建混音器
    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[1] = {SL_IID_BUFFERQUEUE};
    const SLboolean req[1] = {SL_BOOLEAN_TRUE};

    (*engineEngine)->CreateAudioPlayer(engineEngine, &pcmPlayerObject, &slDataSource, &audioSnk, 1, 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);

//    注册回调缓冲区 获取缓冲队列接口
    (*pcmPlayerObject)->GetInterface(pcmPlayerObject, SL_IID_BUFFERQUEUE, &pcmBufferQueue);
    //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);


}

int Audio::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;
}

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