关于ffmpeg示例程序解码Mp3文件的header missing

17年看雷神的文章写（抄）出了第一个用ffmpeg做播放mp3音乐程序。然后就再没碰ffmpeg，因为当时觉得太难了，过了两年后再用ffmpeg4.2去编译当年的代码已经通不过了。后来就尝试用他的示例代码解码，一直失败，提示我missing header，我在整个源码里找这个提示的出处找不到，后来还加了好多ffmpeg群，问大神这个问题，不好意思，没一个人教我，习以为常，自己折腾吧。做为外行菜鸟的我只好下了最笨的功夫，一行一行，一个函数的查起，最后还没查出来。直到编译ShiftMedia,我才知道ffmpeg用的是libmad的库，于是我去研究libmad，也是笨笨的一步步走起，编译，断点，在回调函数里查，好在这源码不多，很容易编译，还有示例程序，最后绕了不少弯路，而且还去补了一下mp3头文件的知识，id3的内容。浪费N多时间后。终于明白了问题所在。libmad解码的是编码部分，去掉帧头和Mp3十个字节的文件头就Ok了。返回到ffmpeg，其实也是这样的办法运行示例程序。只要加一小段代码就可以解码mp3文件了。代码如下：

帧头大小的计算简单的就是mp3头文件十个字节里后四个四节，取后七位联在一起取值。

假设：1001-1000/1000-1000/0111-1111/1111-0000

转后：001-1000-000-1000-111-1111-111-0000

size[n]&0x7f,取后七位，然后再左移<<7，14，21位

 

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include <libavutil/frame.h>
#include <libavutil/mem.h>

#include <libavcodec/avcodec.h>

#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

static int get_format_from_sample_fmt(const char **fmt,
                                      enum AVSampleFormat sample_fmt)
{
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;

    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }

    fprintf(stderr,
            "sample format %s is not supported as output format\n",
            av_get_sample_fmt_name(sample_fmt));
    return -1;
}

static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
                   FILE *outfile)
{
    int i, ch;
    int ret, data_size;

    /* send the packet with the compressed data to the decoder */
    ret = avcodec_send_packet(dec_ctx, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting the packet to the decoder\n");
        exit(1);
    }

    /* read all the output frames (in general there may be any number of them */
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec_ctx, frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            return;
        else if (ret < 0) {
            fprintf(stderr, "Error during decoding\n");
            exit(1);
        }
        data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
        if (data_size < 0) {
            /* This should not occur, checking just for paranoia */
            fprintf(stderr, "Failed to calculate data size\n");
            exit(1);
        }
        for (i = 0; i < frame->nb_samples; i++)
            for (ch = 0; ch < dec_ctx->channels; ch++)
                fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
    }
}

int main(int argc, char **argv)
{
    const char *outfilename, *filename;
    const AVCodec *codec;
    AVCodecContext *c= NULL;
    AVCodecParserContext *parser = NULL;
    int len, ret;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    uint8_t *data;
    size_t   data_size;
    AVPacket *pkt;
    AVFrame *decoded_frame = NULL;
    enum AVSampleFormat sfmt;
    int n_channels = 0;
    const char *fmt;

    if (argc <= 2) {
        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
        exit(0);
    }
    filename    = argv[1];
    outfilename = argv[2];

    pkt = av_packet_alloc();

    /* find the MPEG audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    parser = av_parser_init(codec->id);
    if (!parser) {
        fprintf(stderr, "Parser not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);
    if (!c) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }

    /* open it */
    if (avcodec_open2(c, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "Could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }

    /* decode until eof */
  uint8_t mp3_header[10];
    fread(mp3_header, 10, 1, f);
    long frame_size = (mp3_header[6] & 0xff) << 21 | (mp3_header[7] & 0xff) << 14 | (mp3_header[8] & 0xff) << 7 | mp3_header[9] & 0xff;    data = inbuf;
    fseek(f, frame_size + 10, 0);
    data      = inbuf;
    data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);

    while (data_size > 0) {
        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "Could not allocate audio frame\n");
                exit(1);
            }
        }

        ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
                               data, data_size,
                               AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
        if (ret < 0) {
            fprintf(stderr, "Error while parsing\n");
            exit(1);
        }
        data      += ret;
        data_size -= ret;

        if (pkt->size)
            decode(c, pkt, decoded_frame, outfile);

        if (data_size < AUDIO_REFILL_THRESH) {
            memmove(inbuf, data, data_size);
            data = inbuf;
            len = fread(data + data_size, 1,
                        AUDIO_INBUF_SIZE - data_size, f);
            if (len > 0)
                data_size += len;
        }
    }

    /* flush the decoder */
    pkt->data = NULL;
    pkt->size = 0;
    decode(c, pkt, decoded_frame, outfile);

    /* print output pcm infomations, because there have no metadata of pcm */
    sfmt = c->sample_fmt;

    if (av_sample_fmt_is_planar(sfmt)) {
        const char *packed = av_get_sample_fmt_name(sfmt);
        printf("Warning: the sample format the decoder produced is planar "
               "(%s). This example will output the first channel only.\n",
               packed ? packed : "?");
        sfmt = av_get_packed_sample_fmt(sfmt);
    }

    n_channels = c->channels;
    if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
        goto end;

    printf("Play the output audio file with the command:\n"
           "ffplay -f %s -ac %d -ar %d %s\n",
           fmt, n_channels, c->sample_rate,
           outfilename);
end:
    fclose(outfile);
    fclose(f);

    avcodec_free_context(&c);
    av_parser_close(parser);
    av_frame_free(&decoded_frame);
    av_packet_free(&pkt);

    return 0;
}