FFMpeg笔记（三） 音频处理基本概念及音频重采样

Android放音的采样率固定为44.1KHz，录音的采样率固定为8KHz，因此底层的音频设备驱动需要设置好这两个固定的采样率。如果上层传过来的采样率不符的话，需要进行resample重采样处理。

几个名词：

1. 采样率

采样设备每秒抽取样本的次数

2. 音频格式及量化精度（位宽）

每种音频格式有不同的量化精度（位宽），位数越多，表示值就越精确，声音表现自然就越精准。FFMpeg中音频格式有以下几种，每种格式有其占用的字节数信息：

enum AVSampleFormat {
    AV_SAMPLE_FMT_NONE = -,
    AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
    AV_SAMPLE_FMT_S16,         ///< signed 16 bits
    AV_SAMPLE_FMT_S32,         ///< signed 32 bits
    AV_SAMPLE_FMT_FLT,         ///< float
    AV_SAMPLE_FMT_DBL,         ///< double

    AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
    AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
    AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
    AV_SAMPLE_FMT_FLTP,        ///< float, planar
    AV_SAMPLE_FMT_DBLP,        ///< double, planar
    AV_SAMPLE_FMT_S64,         ///< signed 64 bits
    AV_SAMPLE_FMT_S64P,        ///< signed 64 bits, planar

    AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
};

3. 分片（plane）和打包（packed）

以双声道为例，带P（plane）的数据格式在存储时，其左声道和右声道的数据是分开存储的，左声道的数据存储在data[0]，右声道的数据存储在data[1]，每个声道的所占用的字节数为linesize[0]和linesize[1]；

不带P（packed）的音频数据在存储时，是按照LRLRLR...的格式交替存储在data[0]中，linesize[0]表示总的数据量。

4. 声道分布（channel_layout)

声道分布在FFmpeg\libavutil\channel_layout.h中有定义，一般来说用的比较多的是AV_CH_LAYOUT_STEREO（双声道）和AV_CH_LAYOUT_SURROUND（三声道），这两者的定义如下：

#define AV_CH_LAYOUT_STEREO            (AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT)
#define AV_CH_LAYOUT_SURROUND          (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER)

5. 音频帧的数据量计算

一帧音频的数据量=channel数 * nb_samples样本数 * 每个样本占用的字节数

如果该音频帧是FLTP格式的PCM数据，包含1024个样本，双声道，那么该音频帧包含的音频数据量是2*1024*4=8192字节。

6. 音频播放时间计算

以采样率44100Hz来计算，每秒44100个sample，而正常一帧为1024个sample，可知每帧播放时间/1024=1000ms/44100，得到每帧播放时间=1024*1000/44100=23.2ms。

7. 音频重采样（resample）

FFMpeg自带的resample例子：FFmpeg\doc\examples\resampling_audio.c，这里把最核心的resample代码贴一下，在工程中使用时，注意设置的各种参数，给定的输入数据都不能错。

int main(int argc, char **argv)
{
    // 设置数据源src和dst声道布局
    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
    // 设置src和dst采样率
    int src_rate = , dst_rate = ;
    uint8_t **src_data = NULL, **dst_data = NULL;
    int src_nb_channels = , dst_nb_channels = ;
    int src_linesize, dst_linesize;
    int src_nb_samples = , dst_nb_samples, max_dst_nb_samples;
    // 设置src和dst音频格式
    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
    const char *dst_filename = NULL;
    FILE *dst_file;
    int dst_bufsize;
    const char *fmt;
    // 重采样上下文，包含resample信息
    struct SwrContext *swr_ctx;
    double t;
    int ret;

    if (argc != ) {
        fprintf(stderr, "Usage: %s output_file\n"
                "API example program to show how to resample an audio stream with libswresample.\n"
                "This program generates a series of audio frames, resamples them to a specified "
                "output format and rate and saves them to an output file named output_file.\n",
            argv[]);
        exit();
    }
    // resample后的数据保存到本地文件
    dst_filename = argv[];

    dst_file = fopen(dst_filename, "wb");
    if (!dst_file) {
        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
        exit();
    }

    /* create resampler context */
    swr_ctx = swr_alloc();
    if (!swr_ctx) {
        fprintf(stderr, "Could not allocate resampler context\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }

    /* set options */
    // 将resample信息写入resample上下文
    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, );
    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, );
    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, );

    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, );
    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, );
    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, );

    /* initialize the resampling context */
    if ((ret = swr_init(swr_ctx)) < ) {
        fprintf(stderr, "Failed to initialize the resampling context\n");
        goto end;
    }

    /* allocate source and destination samples buffers */

    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
                                             src_nb_samples, src_sample_fmt, );
    if (ret < ) {
        fprintf(stderr, "Could not allocate source samples\n");
        goto end;
    }

    /* compute the number of converted samples: buffering is avoided
     * ensuring that the output buffer will contain at least all the
     * converted input samples */
    max_dst_nb_samples = dst_nb_samples =
        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

    /* buffer is going to be directly written to a rawaudio file, no alignment */
    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
                                             dst_nb_samples, dst_sample_fmt, );
    if (ret < ) {
        fprintf(stderr, "Could not allocate destination samples\n");
        goto end;
    }

    t = ;
    do {
        /* generate synthetic audio */
        // 这里是自行生成源数据帧，实际工程中应该将解码后的PCM数据填入src_data中
        fill_samples((double *)src_data[], src_nb_samples, src_nb_channels, src_rate, &t);

        /* compute destination number of samples */
        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
        if (dst_nb_samples > max_dst_nb_samples) {
            av_freep(&dst_data[]);
            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
                                   dst_nb_samples, dst_sample_fmt, );
            if (ret < )
                break;
            max_dst_nb_samples = dst_nb_samples;
        }

        /* convert to destination format */
        // 重采样操作
        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
        if (ret < ) {
            fprintf(stderr, "Error while converting\n");
            goto end;
        }
        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                 ret, dst_sample_fmt, );
        if (dst_bufsize < ) {
            fprintf(stderr, "Could not get sample buffer size\n");
            goto end;
        }
        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
        fwrite(dst_data[], , dst_bufsize, dst_file);
    } while (t < );

    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < )
        goto end;
    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);

end:
    fclose(dst_file);

    if (src_data)
        av_freep(&src_data[]);
    av_freep(&src_data);

    if (dst_data)
        av_freep(&dst_data[]);
    av_freep(&dst_data);

    swr_free(&swr_ctx);
    return ret < ;
}