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ffplay音频SDL播放处理

1、从解码数组获取到解码后的数据
 
static int audio_decode_frame(VideoState *is)
{
    int data_size, resampled_data_size;
    av_unused double audio_clock0;
    int wanted_nb_samples;
    Frame *af;

    if (is->paused)
        return -1;

    //音频数组队列获取数据
    do {
#if defined(_WIN32)
        while (frame_queue_nb_remaining(&is->sampq) == 0) {
            if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)
                return -1;
            av_usleep (1000);
        }
#endif
        if (!(af = frame_queue_peek_readable(&is->sampq)))
            return -1;
        frame_queue_next(&is->sampq);
    } while (af->serial != is->audioq.serial);

    // 获取音频数据
    data_size = av_samples_get_buffer_size(NULL, af->frame->ch_layout.nb_channels, af->frame->nb_samples, af->frame->format, 1);

    // 同步视频的转换
    wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);

    // 格式是否与SDL输出格式一样
    if (af->frame->format        != is->audio_src.fmt            ||
        av_channel_layout_compare(&af->frame->ch_layout, &is->audio_src.ch_layout) ||
        af->frame->sample_rate   != is->audio_src.freq           ||
        (wanted_nb_samples       != af->frame->nb_samples && !is->swr_ctx)) {
        swr_free(&is->swr_ctx);

        // 重采样上下文
        swr_alloc_set_opts2(&is->swr_ctx,
                            &is->audio_tgt.ch_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
                            &af->frame->ch_layout, af->frame->format, af->frame->sample_rate,
                            0, NULL);
        if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
            av_log(NULL, AV_LOG_ERROR,
                   "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
                    af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), af->frame->ch_layout.nb_channels,
                    is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.ch_layout.nb_channels);
            swr_free(&is->swr_ctx);
            return -1;
        }
        if (av_channel_layout_copy(&is->audio_src.ch_layout, &af->frame->ch_layout) < 0)
            return -1;
        is->audio_src.freq = af->frame->sample_rate;
        is->audio_src.fmt = af->frame->format;
    }

    if (is->swr_ctx) {// 格式不同需要转换
        //分配空间
        const uint8_t **in = (const uint8_t **)af->frame->extended_data;
        uint8_t **out = &is->audio_buf1;
        int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
        int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.ch_layout.nb_channels, out_count, is->audio_tgt.fmt, 0);
        int len2;
        if (out_size < 0) {
            av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
            return -1;
        }
        if (wanted_nb_samples != af->frame->nb_samples) {
            if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
                                        wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
                av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
                return -1;
            }
        }
        av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
        if (!is->audio_buf1)
            return AVERROR(ENOMEM);

        // 转换
        len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
        if (len2 < 0) {
            av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
            return -1;
        }
        if (len2 == out_count) {
            av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
            if (swr_init(is->swr_ctx) < 0)
                swr_free(&is->swr_ctx);
        }
        is->audio_buf = is->audio_buf1;
        resampled_data_size = len2 * is->audio_tgt.ch_layout.nb_channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
    } else {// 格式相同直接传递
        is->audio_buf = af->frame->data[0];
        resampled_data_size = data_size;
    }

    audio_clock0 = is->audio_clock;
    // af->pts不是无效值时,更新VideoState音频的PTS,让video同步用
    if (!isnan(af->pts))
        is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
    else
        is->audio_clock = NAN;
    is->audio_clock_serial = af->serial;
#ifdef DEBUG
    {
        static double last_clock;
        printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n", is->audio_clock - last_clock, is->audio_clock, audio_clock0);
        last_clock = is->audio_clock;
    }
#endif
    return resampled_data_size;
}
2、放到sdl缓存播放回调函数
static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
    VideoState *is = opaque;
    int audio_size, len1;
    audio_callback_time = av_gettime_relative();

    // 缓存区需要数据填充
    while (len > 0) {
        // audio_buf的播放数据已经消耗完了,需要重新获取数据
        if (is->audio_buf_index >= is->audio_buf_size) {
           audio_size = audio_decode_frame(is);
           if (audio_size < 0) {
                /* if error, just output silence */
               is->audio_buf = NULL;
               is->audio_buf_size = SDL_AUDIO_MIN_BUFFER_SIZE / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
           } 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;
        }

        // 可播放数据长度
        len1 = is->audio_buf_size - is->audio_buf_index;
        if (len1 > len)
            len1 = len;

        //拷贝数据
        if (!is->muted && is->audio_buf && is->audio_volume == SDL_MIX_MAXVOLUME) //音量最大时直接拷贝
            memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
        else { 
            //清空stream
            memset(stream, 0, len1);

            //不是静音时混音处理
            if (!is->muted && is->audio_buf)
                SDL_MixAudioFormat(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, AUDIO_S16SYS, len1, is->audio_volume);
        }

        // 移动位置
        len -= len1;
        stream += len1;
        is->audio_buf_index += len1;
    }

    // 音频播放数据
    is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;

    // 当audio时钟无效时,通过计算采样点获取,我们假设 SDL 使用的音频驱动程序有两个周期。
    if (!isnan(is->audio_clock)) {
        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
        sync_clock_to_slave(&is->extclk, &is->audclk);
    }
}
3、sdl音频播放初始化,在stream_component_open打开音频时打开
static int audio_open(void *opaque, AVChannelLayout *wanted_channel_layout, int wanted_sample_rate, struct AudioParams *audio_hw_params)
{
    SDL_AudioSpec wanted_spec, spec;
    const char *env;
    static const int next_nb_channels[] = {0, 0, 1, 6, 2, 6, 4, 6};
    static const int next_sample_rates[] = {0, 44100, 48000, 96000, 192000};
    int next_sample_rate_idx = FF_ARRAY_ELEMS(next_sample_rates) - 1;
    int wanted_nb_channels = wanted_channel_layout->nb_channels;

    // 获取SDL通道数
    env = SDL_getenv("SDL_AUDIO_CHANNELS");
    if (env) {
        wanted_nb_channels = atoi(env);
        av_channel_layout_uninit(wanted_channel_layout);
        av_channel_layout_default(wanted_channel_layout, wanted_nb_channels);
    }

    // 声道布局变声道数
    if (wanted_channel_layout->order != AV_CHANNEL_ORDER_NATIVE) {
        av_channel_layout_uninit(wanted_channel_layout);
        av_channel_layout_default(wanted_channel_layout, wanted_nb_channels);
    }
    wanted_nb_channels = wanted_channel_layout->nb_channels;

    // 初始化SDL播放参数
    wanted_spec.channels = wanted_nb_channels;
    wanted_spec.freq = wanted_sample_rate;
    if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
        av_log(NULL, AV_LOG_ERROR, "Invalid sample rate or channel count!\n");
        return -1;
    }
    while (next_sample_rate_idx && next_sample_rates[next_sample_rate_idx] >= wanted_spec.freq)
        next_sample_rate_idx--;
    wanted_spec.format = AUDIO_S16SYS;
    wanted_spec.silence = 0;
    wanted_spec.samples = FFMAX(SDL_AUDIO_MIN_BUFFER_SIZE, 2 << av_log2(wanted_spec.freq / SDL_AUDIO_MAX_CALLBACKS_PER_SEC));
    wanted_spec.callback = sdl_audio_callback;
    wanted_spec.userdata = opaque;

    // 打开SDL音频播放器
    while (!(audio_dev = SDL_OpenAudioDevice(NULL, 0, &wanted_spec, &spec, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE | SDL_AUDIO_ALLOW_CHANNELS_CHANGE))) {
        av_log(NULL, AV_LOG_WARNING, "SDL_OpenAudio (%d channels, %d Hz): %s\n", wanted_spec.channels, wanted_spec.freq, SDL_GetError());
        wanted_spec.channels = next_nb_channels[FFMIN(7, wanted_spec.channels)];
        if (!wanted_spec.channels) {
            wanted_spec.freq = next_sample_rates[next_sample_rate_idx--];
            wanted_spec.channels = wanted_nb_channels;
            if (!wanted_spec.freq) {
                av_log(NULL, AV_LOG_ERROR,  "No more combinations to try, audio open failed\n");
                return -1;
            }
        }
        av_channel_layout_default(wanted_channel_layout, wanted_spec.channels);
    }

    // 检测打开的格式是否正确
    if (spec.format != AUDIO_S16SYS) {
        av_log(NULL, AV_LOG_ERROR, "SDL advised audio format %d is not supported!\n", spec.format);
        return -1;
    }

    // 检测打开的通道数是否正确
    if (spec.channels != wanted_spec.channels) {
        av_channel_layout_uninit(wanted_channel_layout);
        av_channel_layout_default(wanted_channel_layout, spec.channels);
        if (wanted_channel_layout->order != AV_CHANNEL_ORDER_NATIVE) {
            av_log(NULL, AV_LOG_ERROR, "SDL advised channel count %d is not supported!\n", spec.channels);
            return -1;
        }
    }

    // 返回播放的音频参数,播放时是否需要转换
    audio_hw_params->fmt = AV_SAMPLE_FMT_S16;
    audio_hw_params->freq = spec.freq;
    if (av_channel_layout_copy(&audio_hw_params->ch_layout, wanted_channel_layout) < 0)
        return -1;
    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->ch_layout.nb_channels, 1, audio_hw_params->fmt, 1);
    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->ch_layout.nb_channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
    if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
        av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
        return -1;
    }
    return spec.size;
}


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