Recherche avancée

Sur d’autres sites (4062)

• How can I extract "GROUP-ID", "LANGUAGE" and "INSTREAM-ID" from ts using ffmpeg

  21 novembre 2020, par user1163234

  I am trying to get the captions from a segment from a live feed. I am running the command

  


  ffmpeg -i seg-1077853030-v1-a1.ts

  


  Output

  


  `Input #0, mpegts, from 'seg-109853030-v1-a1.ts' :
Duration : 00:00:06.01, start : 57867.901133, bitrate : 2649 kb/s
Program 1

  


  Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 98 kb/s
Stream #0:2[0x102]: Data: timed_id3 (ID3  / 0x20334449)`


  


  My question is what command should I run to print out the caption file with the track metadata including label and language.

  


• How to Convert a binary(.bin) file to mp3 using libav in C language ?

  28 août 2018, par danglingptr

  I have created a bin file using my_audio_decode.c. How I can get back the original mp3 file by encoding ?

  P.S. I have gone through ffmpeg code, I modified as my requirements like codec to mp3 etc., You can see my modified code below.

  #include 
  
  #include 
  
  #include 
  
  
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  
  
  #include <libavutil></libavutil>channel_layout.h>
  
  #include <libavutil></libavutil>common.h>
  
  #include <libavutil></libavutil>frame.h>
  
  #include <libavutil></libavutil>samplefmt.h>
  
  
  
  /* check that a given sample format is supported by the encoder */
  
  static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
  
  {
  
      const enum AVSampleFormat *p = codec->sample_fmts;
  
  
  
      while (*p != AV_SAMPLE_FMT_NONE) {
  
          if (*p == sample_fmt)
  
              return 1;
  
          p++;
  
      }
  
      return 0;
  
  }
  
  
  
  /* just pick the highest supported samplerate */
  
  static int select_sample_rate(const AVCodec *codec)
  
  {
  
      const int *p;
  
      int best_samplerate = 0;
  
  
  
      if (!codec->supported_samplerates)
  
          return 44100;
  
  
  
      p = codec->supported_samplerates;
  
      while (*p) {
  
          if (!best_samplerate || abs(44100 - *p) &lt; abs(44100 - best_samplerate))
  
              best_samplerate = *p;
  
          p++;
  
      }
  
      return best_samplerate;
  
  }
  
  
  
  /* select layout with the highest channel count */
  
  static int select_channel_layout(const AVCodec *codec)
  
  {
  
      const uint64_t *p;
  
      uint64_t best_ch_layout = 0;
  
      int best_nb_channels   = 0;
  
  
  
      if (!codec->channel_layouts)
  
          return AV_CH_LAYOUT_STEREO;
  
  
  
      p = codec->channel_layouts;
  
      while (*p) {
  
          int nb_channels = av_get_channel_layout_nb_channels(*p);
  
  
  
          if (nb_channels > best_nb_channels) {
  
              best_ch_layout    = *p;
  
              best_nb_channels = nb_channels;
  
          }
  
          p++;
  
      }
  
      return best_ch_layout;
  
  }
  
  
  
  static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
  
                     FILE *output)
  
  {
  
      int ret;
  
  
  
     /* send the frame for encoding */
  
     ret = avcodec_send_frame(ctx, frame);
  
     if (ret &lt; 0) {
  
         fprintf(stderr, "Error sending the frame to the encoder\n");
  
         exit(1);
  
     }
  
  
  
     /* read all the available output packets (in general there may be any
  
      * number of them */
  
     while (ret >= 0) {
  
         ret = avcodec_receive_packet(ctx, pkt);
  
         if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
  
             return;
  
         else if (ret &lt; 0) {
  
             fprintf(stderr, "Error encoding audio frame\n");
  
             exit(1);
  
         }
  
  
  
         fwrite(pkt->data, 1, pkt->size, output);
  
         av_packet_unref(pkt);
  
     }
  
  }
  
  
  
  int main(int argc, char **argv)
  
  {
  
     const char *filename;
  
     const AVCodec *codec;
  
     AVCodecContext *c= NULL;
  
     AVFrame *frame;
  
     AVPacket *pkt;
  
     int i, j, k, ret;
  
     FILE *f;
  
     uint16_t *samples;
  
     float t, tincr;
  
  
  
     av_register_all();
  
     avcodec_register_all();
  
  
  
     if (argc &lt;= 1) {
  
         fprintf(stderr, "Usage: %s <output file="file">\n", argv[0]);
  
         return 0;
  
     }
  
     filename = argv[1];
  
  
  
     /* find the MP2 encoder */
  
     codec = avcodec_find_encoder(AV_CODEC_ID_MP3);
  
     if (!codec) {
  
         fprintf(stderr, "Codec not found\n");
  
         exit(1);
  
     }
  
  
  
     c = avcodec_alloc_context3(codec);
  
     if (!c) {
  
         fprintf(stderr, "Could not allocate audio codec context\n");
  
         exit(1);
  
     }
  
  
  
     /* put sample parameters */
  
     c->bit_rate = 64000;
  
  
  
     /* check that the encoder supports s16 pcm input */
  
     c->sample_fmt = AV_SAMPLE_FMT_S16P;
  
     if (!check_sample_fmt(codec, c->sample_fmt)) {
  
         fprintf(stderr, "Encoder does not support sample format %s",
  
                 av_get_sample_fmt_name(c->sample_fmt));
  
         exit(1);
  
     }
  
  
  
     /* select other audio parameters supported by the encoder */
  
     c->sample_rate    = select_sample_rate(codec);
  
     c->channel_layout = select_channel_layout(codec);
  
     c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);
  
  
  
     /* open it */
  
     if (avcodec_open2(c, codec, NULL) &lt; 0) {
  
         fprintf(stderr, "Could not open codec\n");
  
         exit(1);
  
     }
  
  
  
     f = fopen(filename, "wb");
  
     if (!f) {
  
         fprintf(stderr, "Could not open %s\n", filename);
  
         exit(1);
  
     }
  
  
  
     /* packet for holding encoded output */
  
     pkt = av_packet_alloc();
  
     if (!pkt) {
  
         fprintf(stderr, "could not allocate the packet\n");
  
         exit(1);
  
     }
  
  
  
     /* frame containing input raw audio */
  
     frame = av_frame_alloc();
  
     if (!frame) {
  
         fprintf(stderr, "Could not allocate audio frame\n");
  
         exit(1);
  
     }
  
  
  
     frame->nb_samples     = c->frame_size;
  
     frame->format         = c->sample_fmt;
  
     frame->channel_layout = c->channel_layout;
  
  
  
     /* allocate the data buffers */
  
     ret = av_frame_get_buffer(frame, 0);
  
     if (ret &lt; 0) {
  
         fprintf(stderr, "Could not allocate audio data buffers\n");
  
         exit(1);
  
     }
  
  
  
     /* encode a single tone sound */
  
     t = 0;
  
     tincr = 2 * M_PI * 440.0 / c->sample_rate;
  
     for (i = 0; i &lt; 200; i++) {
  
         /* make sure the frame is writable -- makes a copy if the encoder
  
          * kept a reference internally */
  
         ret = av_frame_make_writable(frame);
  
         if (ret &lt; 0)
  
             exit(1);
  
         samples = (uint16_t*)frame->data[0];
  
  
  
         for (j = 0; j &lt; c->frame_size; j++) {
  
             samples[2*j] = (int)(sin(t) * 10000);
  
  
  
             for (k = 1; k &lt; c->channels; k++)
  
                 samples[2*j + k] = samples[2*j];
  
             t += tincr;
  
         }
  
         encode(c, frame, pkt, f);
  
     }
  
  
  
     /* flush the encoder */
  
     encode(c, NULL, pkt, f);
  
  
  
     fclose(f);
  
  
  
     av_frame_free(&amp;frame);
  
     av_packet_free(&amp;pkt);
  
     avcodec_free_context(&amp;c);
  
  
  
     return 0;
  
  }
  
  </output>

  But the code is confusing and it is taking only one input as argument, I need a clear working example c code like it has to take 2 args, like input file and output file.

• Revert "lavc/utils : Do not require dimensions for PNG."

  13 juillet 2014, par Derek Buitenhuis

  Revert "lavc/utils : Do not require dimensions for PNG."
  mplayer-specifc hacks should not be in our codebase. mplayer should fix
  
  its own code. It is not our responsibility to work around their broken
  
  code.
  This reverts commit e8e575633faf19711910cf9caf59f7db300a9ccd.
  Signed-off-by : Derek Buitenhuis <derek.buitenhuis@gmail.com>
  

  • [DH] libavcodec/utils.c