ffplay源码分析04 ---- 音频输出

简介:

ffplay的音频输出通过SDL实现,主要流程分为如下几步:

  • 打开音频设备,设置参数
  • 启动SDL音频设备播放
  • SDL音频回调函数读取数据

代码如下:

case AVMEDIA_TYPE_AUDIO:
        //从avctx(即AVCodecContext)中获取音频格式参数
        sample_rate    = avctx->sample_rate;
        nb_channels    = avctx->channels;
        channel_layout = avctx->channel_layout;

        /* prepare audio output 准备音频输出*/
        //调用audio_open打开sdl音频输出,实际打开的设备参数保存在audio_tgt,返回值表示输出设备的缓冲区大小
        if ((ret = audio_open(is, channel_layout, nb_channels, sample_rate, &is->audio_tgt)) < 0)
            goto fail;
        is->audio_hw_buf_size = ret;
        is->audio_src = is->audio_tgt;  //暂且将数据源参数等同于目标输出参数
        //初始化audio_buf相关参数
        is->audio_buf_size  = 0;
        is->audio_buf_index = 0;

        /* init averaging filter 初始化averaging滤镜, 非audio master时使用 */
        is->audio_diff_avg_coef  = exp(log(0.01) / AUDIO_DIFF_AVG_NB);
        is->audio_diff_avg_count = 0;
        /* 由于我们没有精确的音频数据填充FIFO,故只有在大于该阈值时才进行校正音频同步*/
        is->audio_diff_threshold = (double)(is->audio_hw_buf_size) / is->audio_tgt.bytes_per_sec;

        is->audio_stream = stream_index;    // 获取audio的stream索引
        is->audio_st = ic->streams[stream_index];  // 获取audio的stream指针
        // 初始化ffplay封装的音频解码器
        decoder_init(&is->auddec, avctx, &is->audioq, is->continue_read_thread);
        if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek) {
            is->auddec.start_pts = is->audio_st->start_time;
            is->auddec.start_pts_tb = is->audio_st->time_base;
        }
        // 启动音频解码线程
        if ((ret = decoder_start(&is->auddec, audio_thread, "audio_decoder", is)) < 0)
            goto out;
        // 0:播放 非0:暂停
        SDL_PauseAudioDevice(audio_dev, 0);
        break;

打开音频设备

static int audio_open(void *opaque, int64_t wanted_channel_layout,
                      int wanted_nb_channels, 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;

    env = SDL_getenv("SDL_AUDIO_CHANNELS");
    if (env) {  // 若环境变量有设置,优先从环境变量取得声道数和声道布局
        wanted_nb_channels = atoi(env);
        wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
    }
    if (!wanted_channel_layout || wanted_nb_channels != av_get_channel_layout_nb_channels(wanted_channel_layout)) {
        wanted_channel_layout = av_get_default_channel_layout(wanted_nb_channels);
        wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
    }
    // 根据channel_layout获取nb_channels,当传入参数wanted_nb_channels不匹配时,此处会作修正
    wanted_nb_channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
    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--;  // 从采样率数组中找到第一个不大于传入参数wanted_sample_rate的值
    // 音频采样格式有两大类型:planar和packed,假设一个双声道音频文件,一个左声道采样点记作L,一个右声道采样点记作R,则:
    // planar存储格式:(plane1)LLLLLLLL...LLLL (plane2)RRRRRRRR...RRRR
    // packed存储格式:(plane1)LRLRLRLR...........................LRLR
    // 在这两种采样类型下,又细分多种采样格式,如AV_SAMPLE_FMT_S16、AV_SAMPLE_FMT_S16P等,
    // 注意SDL2.0目前不支持planar格式
    // channel_layout是int64_t类型,表示音频声道布局,每bit代表一个特定的声道,参考channel_layout.h中的定义,一目了然
    // 数据量(bits/秒) = 采样率(Hz) * 采样深度(bit) * 声道数
    wanted_spec.format = AUDIO_S16SYS;
    wanted_spec.silence = 0;
    /*
     * 一次读取多长的数据
     * SDL_AUDIO_MAX_CALLBACKS_PER_SEC一秒最多回调次数,避免频繁的回调
     *  Audio buffer size in samples (power of 2)
     */
     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;
    // 打开音频设备并创建音频处理线程。期望的参数是wanted_spec,实际得到的硬件参数是spec
    // 1) SDL提供两种使音频设备取得音频数据方法:
    //    a. push,SDL以特定的频率调用回调函数,在回调函数中取得音频数据
    //    b. pull,用户程序以特定的频率调用SDL_QueueAudio(),向音频设备提供数据。此种情况wanted_spec.callback=NULL
    // 2) 音频设备打开后播放静音,不启动回调,调用SDL_PauseAudio(0)后启动回调,开始正常播放音频
    // SDL_OpenAudioDevice()第一个参数为NULL时,等价于SDL_OpenAudio()
    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;
            }
        }
        wanted_channel_layout = av_get_default_channel_layout(wanted_spec.channels);
    }
    // 检查打开音频设备的实际参数:采样格式
    if (spec.format != AUDIO_S16SYS) {
        av_log(NULL, AV_LOG_ERROR,
               "SDL advised audio format %d is not supported!\n", spec.format);
        return -1;
    }
    // 检查打开音频设备的实际参数:声道数
    if (spec.channels != wanted_spec.channels) {
        wanted_channel_layout = av_get_default_channel_layout(spec.channels);
        if (!wanted_channel_layout) {
            av_log(NULL, AV_LOG_ERROR,
                   "SDL advised channel count %d is not supported!\n", spec.channels);
            return -1;
        }
    }
    // wanted_spec是期望的参数,spec是实际的参数,wanted_spec和spec都是SDL中的结构。
    // 此处audio_hw_params是FFmpeg中的参数,输出参数供上级函数使用
    // audio_hw_params保存的参数,就是在做重采样的时候要转成的格式。
    audio_hw_params->fmt = AV_SAMPLE_FMT_S16;
    audio_hw_params->freq = spec.freq;
    audio_hw_params->channel_layout = wanted_channel_layout;
    audio_hw_params->channels =  spec.channels;
    /* audio_hw_params->frame_size这里只是计算一个采样点占用的字节数 */
    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels,
                                                             1, audio_hw_params->fmt, 1);
    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->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;
    }
    // 比如2帧数据,一帧就是1024个采样点, 1024*2*2 * 2 = 8192字节
    return spec.size;    /* SDL内部缓存的数据字节, samples * channels *byte_per_sample */
}

读取数据

static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
    VideoState *is = opaque;
    int audio_size, len1;

    audio_callback_time = av_gettime_relative();

    while (len > 0) {   // 循环读取,直到读取到足够的数据
        /* (1)如果is->audio_buf_index < is->audio_buf_size则说明上次拷贝还剩余一些数据,
         * 先拷贝到stream再调用audio_decode_frame
         * (2)如果audio_buf消耗完了,则调用audio_decode_frame重新填充audio_buf
         */
        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; // 讲字节 多少字节
            }
            is->audio_buf_index = 0;
        }
        //根据缓冲区剩余大小量力而行
        len1 = is->audio_buf_size - is->audio_buf_index;
        if (len1 > len)  // len = 3000 < len1 4096
            len1 = len;
        //根据audio_volume决定如何输出audio_buf
        /* 判断是否为静音,以及当前音量的大小,如果音量为最大则直接拷贝数据 */
        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 {
            memset(stream, 0, len1);
            // 3.调整音量
            /* 如果处于mute状态则直接使用stream填0数据, 暂停时is->audio_buf = NULL */
            if (!is->muted && is->audio_buf)
                SDL_MixAudioFormat(stream, (uint8_t *)is->audio_buf + is->audio_buf_index,
                                   AUDIO_S16SYS, len1, is->audio_volume);
        }
        len -= len1;
        stream += len1;
        /* 更新is->audio_buf_index,指向audio_buf中未被拷贝到stream的数据(剩余数据)的起始位置 */
        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. */
    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);
    }
}

 

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