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• Http Streaming from ffmpeg, How to get a sequenced packet ?

  10 septembre 2011, par manutd

  I am trying to make http streaming program.
  So I follow this code at this.
  However, when i decode, only one frame is decoded.
  I think I need call back function.
  Do you know how to make a call back function ?
  I know 'asf' packet's call back function is like int read_data(void *opaque, char *buf, int buf_size)

  But the other formats(mp3, ogg, aac, ..) doesn't work..

  Please help me.

  Any advice or comment are very appreciated.

  #include 
  
  #include 
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libavdevice></libavdevice>avdevice.h>
  
  
  
  int main(int argc, char **argv)
  
  {
  
          static AVInputFormat *file_iformat;
  
          static AVFormatContext *pFormatCtx;
  
          AVFormatParameters params;
  
  
  
          AVCodecContext *pCodecCtx;
  
          AVCodec *pCodec;
  
  
  
          const char url[] = "http://listen.radionomy.com/feelingfloyd";
  
  
  
          avcodec_register_all();
  
          avdevice_register_all();
  
          av_register_all();
  
  
  
          av_log_set_level(AV_LOG_VERBOSE);
  
  
  
          file_iformat = av_find_input_format("mp3"); /* mp3 demuxer */
  
          if (!file_iformat)
  
          {
  
                  fprintf(stderr, "Unknown input format: %s\n", &amp;url[0]);
  
                  exit(1);
  
          }
  
  
  
          //file_iformat->flags |= AVFMT_NOFILE; /* ??? */
  
          params.prealloced_context = 0;
  
  
  
          if (av_open_input_file(&amp;pFormatCtx, &amp;url[0], file_iformat, 0, &amp;params) != 0)
  
          {
  
                  fprintf(stderr, "err 1\n");
  
                  exit(2);
  
          }
  
  
  
          /* poulates AVFormatContex structure */
  
          if (av_find_stream_info(pFormatCtx) &lt; 0)
  
          {
  
                  fprintf(stderr, "err 2\n");
  
          }
  
  
  
          /* sanity check (1 stream) */
  
          if (pFormatCtx->nb_streams != 1 &amp;&amp;
  
                          pFormatCtx->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO)
  
          {
  
                  fprintf(stderr, "err 3\n");
  
          }
  
  
  
          pCodecCtx = pFormatCtx->streams[0]->codec;
  
  
  
          /* find decoder for input audio stream */
  
          pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
  
          if (pCodec == NULL)
  
          {
  
                  fprintf(stderr, "err 4: unsupported codec\n");
  
          }
  
  
  
          if (pCodec->capabilities &amp; CODEC_CAP_TRUNCATED)
  
                  pCodecCtx->flags |= CODEC_FLAG_TRUNCATED;
  
  
  
          if (avcodec_open(pCodecCtx, pCodec) &lt; 0)
  
          {
  
                  fprintf(stderr, "err 5\n");
  
          }
  
  
  
          {
  
                  uint8_t *pAudioBuffer;
  
                  AVPacket pkt;
  
  
  
                  int ret;
  
                  int data_size = 2 * AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
  
  
                  av_init_packet(&amp;pkt);
  
                  //pkt.data=NULL;
  
                  //pkt.size=0;
  
                  //pkt.stream_index = 0;
  
  
  
                  pAudioBuffer = av_malloc(data_size * sizeof(int16_t));
  
  
  
                  while (av_read_frame(pFormatCtx, &amp;pkt) == 0) {
  
                          //data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
                          ret = avcodec_decode_audio3(pFormatCtx->streams[pkt.stream_index]->codec,
  
                                          (int16_t *)pAudioBuffer, &amp;data_size, &amp;pkt);
  
  
  
                          /* got an error (-32) here */
  
                          if (ret &lt; 0) {
  
                                  av_strerror(ret, (char *)pAudioBuffer, data_size);
  
                                  fprintf(stderr, "err 6 (%s)\n", pAudioBuffer);
  
                                  break;
  
                          }
  
  
  
                          printf("size=%d, stream_index=%d |ret=%d data_size=%d\n",
  
                                          pkt.size, pkt.stream_index, ret, data_size);
  
                          av_free_packet(&amp;pkt);
  
                  }
  
  
  
                  av_free(pAudioBuffer);
  
          }
  
  
  
          avcodec_close(pCodecCtx);
  
          av_close_input_file(pFormatCtx);
  
  
  
          return 0;
  
  }

• Node.js Stream Mp3 to http without having to save file

  10 janvier 2019, par user2758113

  I am trying to stream just audio from a youtube link straight to http with node.js.

  My code looks like this, I am using express 4.0.

  var express = require('express');
  
  var router = express.Router();
  
  var ytdl = require('ytdl');
  
  var ffmpeg = require('fluent-ffmpeg');
  
  var fs = require('fs');
  
  
  
  router.get('/', function(req, res) {
  
  
  
    var url = 'https://www.youtube.com/watch?v=GgcHlZsOgQo';
  
    var video = ytdl(url)
  
  
  
    res.set({
  
        "Content-Type": "audio/mpeg"
  
    })
  
  
  
    new ffmpeg({source: video})
  
        .toFormat('mp3')
  
        .writeToStream(res, function(data, err) {
  
          if (err) console.log(err)
  
        })
  
  
  
  });
  
  
  
  module.exports = router;

  Now, I’m able to stream the video’s audio to the response if I save the file then pipe it to the response, but I’d rather try to figure out some way to go from downloading to ffmpeg to response.

  Not sure if this is possible. The main goal is to keep it as light weight as possible, and not have to read from files.

  I’ve seen this code which is essentially what I’d like to do minus the saving to a file part.

  

• avformat/dashenc : Support HTTP Persistent for master.m3u8 as well

  18 octobre 2018, par kjeyapal@akamai.com

  avformat/dashenc : Support HTTP Persistent for m3u8 as well
  

  • [DH] libavformat/dashenc.c