ijkplayer音频解码与播放

解码线程

音频解码线程的入口函数是audio_thread()

static int audio_thread(void *arg)
{
//...
    do {
        ffp_audio_statistic_l(ffp);
        if ((got_frame = decoder_decode_frame(ffp, &is->auddec, frame, NULL)) < 0)
            goto the_end;
            //...
            while ((ret = av_buffersink_get_frame_flags(is->out_audio_filter, frame, 0)) >= 0) {
    //...
                if (!(af = frame_queue_peek_writable(&is->sampq)))
                    goto the_end;

        //...

                av_frame_move_ref(af->frame, frame);
                frame_queue_push(&is->sampq);
//...
        }
    } while (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF);
 the_end:
//...
    av_frame_free(&frame);
    return ret;
}

一开始就进入循环，然后调用decoder_decode_frame()进行解码，解码后的帧存放到frame中，然后调用frame_queue_peek_writable()判断是否能把刚刚解码的frame写入is->sampq中（判断sampq队列是否满了），如果没位置放frame的话，会调用pthread_cond_wait()阻塞队列。is->sampq是音频解码帧列表，播放线程直接从这里读取数据然后播放出来。最后调用av_frame_move_ref(af->frame, frame)，把frame放入到sampq的相应位置。前面的af = frame_queue_peek_writable(&is->sampq)，af就是指向这一帧frame应该放的位置的指针，所以直接把值赋值给它的结构体里面的frame就行了。

然后frame_queue_push(&is->sampq)里面是一个唤醒线程操作，如果音频播放线程因为sampq队列为空而阻塞，这里可以唤醒它。

在decoder_decode_frame()里面调用传进去的codec的codec->decode()方法解码。

播放流程

初始化IjkMediaPlayer的时候，在ijkmp_android_create()方法：

IjkMediaPlayer *ijkmp_android_create(int(*msg_loop)(void*))
{
    IjkMediaPlayer *mp = ijkmp_create(msg_loop);

    mp->ffplayer->vout = SDL_VoutAndroid_CreateForAndroidSurface();
    if (!mp->ffplayer->vout)
        goto fail;

    mp->ffplayer->pipeline = ffpipeline_create_from_android(mp->ffplayer);

    ffpipeline_set_vout(mp->ffplayer->pipeline, mp->ffplayer->vout);

    return mp;

}

在ffpipeline_create_from_android()里面有一句：

pipeline->func_open_audio_output  = func_open_audio_output;

func_open_audio_output()

static SDL_Aout *func_open_audio_output(IJKFF_Pipeline *pipeline, FFPlayer *ffp)
{
    SDL_Aout *aout = NULL;
    if (ffp->opensles) {
        aout = SDL_AoutAndroid_CreateForOpenSLES();
    } else {
        aout = SDL_AoutAndroid_CreateForAudioTrack();
    }
    if (aout)
        SDL_AoutSetStereoVolume(aout, pipeline->opaque->left_volume, pipeline->opaque->right_volume);
    return aout;
}

可以看出，音频播放也分为opensles，audiotrack，看一下audiotrack:

SDL_Aout *SDL_AoutAndroid_CreateForAudioTrack()
{
    SDL_Aout *aout = SDL_Aout_CreateInternal(sizeof(SDL_Aout_Opaque));
    if (!aout)
        return NULL;

    SDL_Aout_Opaque *opaque = aout->opaque;
    opaque->wakeup_cond  = SDL_CreateCond();
    opaque->wakeup_mutex = SDL_CreateMutex();
    opaque->speed        = 1.0f;

    aout->opaque_class = &g_audiotrack_class;
    aout->free_l       = aout_free_l;
    // 
    aout->open_audio   = aout_open_audio;
    aout->pause_audio  = aout_pause_audio;
    aout->flush_audio  = aout_flush_audio;
    aout->set_volume   = aout_set_volume;
    aout->close_audio  = aout_close_audio;
    aout->func_get_audio_session_id = aout_get_audio_session_id;
    aout->func_set_playback_rate    = func_set_playback_rate;

    return aout;
}

前面stream_component_open()里面相当于直接调用了aout_open_audio()，里面接着会调用aout_open_audio_n()，然后

SDL_CreateThreadEx(&opaque->_audio_tid, aout_thread, aout, "ff_aout_android");

这里创建的线程就是播放线程。aout_thread这个函数内部会调用aout_thread_n()

static int aout_thread_n(JNIEnv *env, SDL_Aout *aout)
{
  SDL_Aout_Opaque *opaque = aout->opaque;
    SDL_Android_AudioTrack *atrack = opaque->atrack;
    SDL_AudioCallback audio_cblk = opaque->spec.callback;
    void *userdata = opaque->spec.userdata;
    uint8_t *buffer = opaque->buffer;
  //...
    if (!opaque->abort_request && !opaque->pause_on)
        SDL_Android_AudioTrack_play(env, atrack);

    while (!opaque->abort_request) {
        SDL_LockMutex(opaque->wakeup_mutex);
        if (!opaque->abort_request && opaque->pause_on) {
            SDL_Android_AudioTrack_pause(env, atrack);
            while (!opaque->abort_request && opaque->pause_on) {
                SDL_CondWaitTimeout(opaque->wakeup_cond, opaque->wakeup_mutex, 1000);
            }
            if (!opaque->abort_request && !opaque->pause_on)
                SDL_Android_AudioTrack_play(env, atrack);
        }
        if (opaque->need_flush) {
            opaque->need_flush = 0;
            SDL_Android_AudioTrack_flush(env, atrack);
        }
        if (opaque->need_set_volume) {
            opaque->need_set_volume = 0;
            SDL_Android_AudioTrack_set_volume(env, atrack, opaque->left_volume, opaque->right_volume);
        }
        if (opaque->speed_changed) {
            opaque->speed_changed = 0;
            if (J4A_GetSystemAndroidApiLevel(env) >= 23) {
                SDL_Android_AudioTrack_setSpeed(env, atrack, opaque->speed);
            }
        }
        SDL_UnlockMutex(opaque->wakeup_mutex);

        audio_cblk(userdata, buffer, copy_size);
        if (opaque->need_flush) {
            SDL_Android_AudioTrack_flush(env, atrack);
            opaque->need_flush = false;
        }

        if (opaque->need_flush) {
            opaque->need_flush = 0;
            SDL_Android_AudioTrack_flush(env, atrack);
        } else {
            int written = SDL_Android_AudioTrack_write(env, atrack, buffer, copy_size);
            if (written != copy_size) {
                ALOGW("AudioTrack: not all data copied %d/%d", (int)written, (int)copy_size);
            }
        }

        // TODO: 1 if callback return -1 or 0
    }
}

SDL_Android_AudioTrack_set_xxx()，主要是设置播放器相关的配置，比如播放速度，声音大小等。
我们来看一下audio_cblk()，在stream_component_open()的里面调用的audio_open()，会有这么一句代码:

wanted_spec.callback = sdl_audio_callback;

上面aout_thread_n()里调用的audio_cblk(),实际上就是调用的opaque->spec.callback,其实就是调用到sdl_audio_callback()这个函数来了。

static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
//...
    while (len > 0) {
        if (is->audio_buf_index >= is->audio_buf_size) {
           audio_size = audio_decode_frame(is);
           if (audio_size < 0) {
                /* 发生错误，就输出silence */
           //...
           } else {
               if (is->show_mode != SHOW_MODE_VIDEO)
                   update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
               is->audio_buf_size = audio_size;
           }
           is->audio_buf_index = 0;
        }
        len1 = is->audio_buf_size - is->audio_buf_index;
        if (len1 > len)
            len1 = len;
        if (!is->muted && is->audio_buf && is->audio_volume == SDL_MIX_MAXVOLUME)
            memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
        else {
           //...
        }
        len -= len1;
        stream += len1;
        is->audio_buf_index += len1;
    }
    is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
    /* Let's assume the audio driver that is used by SDL has two periods. */
  //...
}

其中重要代码是：

audio_size = audio_decode_frame(ffp);
memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);

audio_decode_frame()

/**
 * Decode one audio frame and return its uncompressed size.
 *
 * The processed audio frame is decoded, converted if required, and
 * stored in is->audio_buf, with size in bytes given by the return
 * value.
 */
static int audio_decode_frame(FFPlayer *ffp)｛
    af = frame_queue_peek_readable(&is->sampq)
    is->audio_buf = af->frame->data[0];
｝

这里主要是判断解码后的is->sampq是否为空（解码时放入is->sampq判断是否满），如果为空就阻塞（解码的时候，每向is->sampq放入一frame，就唤醒线程），不为空就返回队列的第一个frame，然后赋值给ffp->is->audio_buf。
接着把刚刚赋值的ffp->is->audio_buf copy到stream中，流的另一头在哪里？
再返回到aout_thread_n()中：

SDL_Android_AudioTrack_write(env, atrack, buffer, copy_size);

这里的buffer就是刚刚的stream，该函数调用：

 (*env)->SetByteArrayRegion(env, atrack->byte_buffer, 0, (int)size_in_byte, (jbyte*) data);
J4AC_AudioTrack__write(env, atrack->thiz, atrack->byte_buffer, 0, (int)size_in_byte);

这里先是把buffer(data)拷贝到数组中，后面会把这个数组也就是音频帧传递给java。
J4AC_AudioTrack__write()中继续跟踪会发现：

jint J4AC_android_media_AudioTrack__write(JNIEnv *env, jobject thiz, jbyteArray audioData, jint offsetInBytes, jint sizeInBytes)
{
    return (*env)->CallIntMethod(env, thiz, class_J4AC_android_media_AudioTrack.method_write, audioData, offsetInBytes, sizeInBytes);
}

这里又调用到java里面去了，这里调用了java层的AudioTrack.java中的write()函数。