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• ffmpeg : Audio/Video Fade in/out

  31 mai 2017, par dazzafact

  I have this working Script with Audio fading. How can i input also a Fading for the video in and out. It always gives me an error :

  "Option filter:v (set stream filtergraph) cannot be applied to input url ./mp3/conv/1.m4a -- you are trying to apply an input option to an output file or vice versa. Move this option before the file it belongs to."

  This Works with audio-fading:

   ffmpeg  -ss 00:00:00 -t 90 -i "concat:intermediate0.ts|concat:intermediate1.ts"  
  -i "./mp3/conv/1.m4a" -af "afade=t=out:st=84:d=6"  -map 0:v:0 -map 1:a:0 
  video/out515.mp4 -y
  

  This doesn't Work with Audio+Vic-^Fading:

  ffmpeg  -ss 00:00:00 -t 90-i  "concat:intermediate0.ts|intermediate1.ts" 
  -filter:v 'fade=in:0:30,fade=out:250:30' -i "./mp3/conv/1.m4a" 
  -af "afade=t=out:st=84:d=6" -map 0:v:0 -map 1:a:0  video/out515.mp4 -y
  

• Using videoshow (npm module) with ffmpeg to conver audio+image into video

  31 mai 2017, par delesslin

  I'm trying to use the videoshow utility to combine a short audio clip with an image on my ubuntu system. I installed ffmpeg globally while in the root directory using:

  sudo apt-get install ffmpeg
  

  I then installed videoshow inside the project folder using:

  sudo npm install videoshow
  

  The project folder contains 3 files plus the node_modules folder: an image (wolf.jpg), an audio clip (wolf.mp3), and a js file (audio.js). I derived audio.js from an example script on the videoshow github page. Here is my script:

      var videoshow = require('videoshow')
  
  var images = [
    "wolf.jpg"
  ]
  
  var videoOptions = {
    fps: 25,
    loop: 5, // seconds
    transition: true,
    transitionDuration: 1, // seconds
    videoBitrate: 1024,
    videoCodec: 'libx264',
    size: '640x?',
    audioBitrate: '128k',
    audioChannels: 2,
    format: 'mp4',
    pixelFormat: 'yuv420p'
  }
  
  videoshow(images, videoOptions)
    .audio('wolf.mp3')
    .save('wolf.mp4')
    .on('start', function (command) {
      console.log('ffmpeg process started:', command)
    })
    .on('error', function (err, stdout, stderr) {
      console.error('Error:', err)
      console.error('ffmpeg stderr:', stderr)
    })
    .on('end', function (output) {
      console.error('Video created in:', output)
    })
  

  In the terminal, inside the project folder I then call:

  node audio.js
  

  The terminal is silent for a moment followed by:

      ffmpeg process started: ffmpeg -i /tmp/videoshow-db63732f-7376-4663-a7bc-c061091e579a -y -filter_complex concat=n=1:v=1:a=0 wolf.mp4
  ffmpeg process started: ffmpeg -i /tmp/videoshow-1f8851b4-c297-4070-a249-3624970dbb85 -i wolf.mp3 -y -b:a 128k -ac 2 -r 25 -b:v 1024k -vcodec libx264 -filter:v scale=w=640:h=trunc(ow/a/2)*2 -f mp4 -map 0:0 -map 1:0 -t 5 -af afade=t=in:ss=0:st=0:d=3 -af afade=t=out:st=2:d=3 -pix_fmt yuv420p wolf.mp4
  Error: [Error: ffmpeg exited with code 1: ]
  ffmpeg stderr: undefined
  

  I'm not sure why this isn't working, but any/all assistance would be deeply appreciated...

  Hawu'h (thanks), Roo

• how to make video module with nodejs like facebook did with photos

  31 mai 2017, par aharit

  Just wondering how to start ? The needs :

  With around 10 photos, being able to produce a small video of 5-10sec, with animations for example (transition ?), i want to reproduce the facebook videos process if anybody know about that, which technical stack is the best, modules (ffmpeg, wrapper ffmpeg)(pyhton, nodejs).

  Thx

• FFMPEG trim by remux doesn't write a keyframe

  31 mai 2017, par DweebsUnited

  I am using the FFMPEG libraries to trim video files. I do this all as a remux, with no encoding or decoding.

  Trimming currently works correctly with audio, but the trimmed video data appears as a solid color, with small squares of pixels changing. I believe this is because I am not catching/writing a keyframe. It is my understanding that av_seek_frame will seek to a keyframe, which does not seem to be the case..

  If need be, can I decode and then reencode just the first video frame I read after seeking? This will probably be more code than reencoding every frame, but speed is the primary issue here, not complexity.

  Thank you for any help. Also I apologize if I am misunderstanding something to do with video files, I'm still new to this.

  Example output frame:

  Code, adapted from the remux example provided with ffmpeg:

  const char *out_filename = "aaa.mp4";
  
  FILE *fp = fdopen(fd, "r");
  fseek(fp, 0, SEEK_SET);
  
  if ( fp ) {
  
      // Build an ffmpeg file
      char path[512];
      sprintf(path, "pipe:%d", fileno(fp));
  
      // Turn on verbosity
      av_log_set_level( AV_LOG_DEBUG );
      av_log_set_callback( avLogCallback );
  
  
      av_register_all();
      avcodec_register_all();
  
  
      AVOutputFormat *ofmt = NULL;
      AVFormatContext *ifmt_ctx = avformat_alloc_context(), *ofmt_ctx = NULL;
      AVPacket pkt;
      int ret, i;
  
  
      if ((ret = avformat_open_input(&ifmt_ctx, path, av_find_input_format("mp4"), NULL)) < 0) {
          LOG("Could not open input file '%s'", path);
          goto end;
      }
  
      if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
          LOG("Failed to retrieve input stream information", "");
          goto end;
      }
  
  
      avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
      if (!ofmt_ctx) {
          LOG("Could not create output context\n");
          ret = AVERROR_UNKNOWN;
          goto end;
      }
  
      ofmt = ofmt_ctx->oformat;
  
  
      for (i = 0; i < ifmt_ctx->nb_streams; i++) {
          AVStream *in_stream = ifmt_ctx->streams[i];
          AVStream *out_stream = avformat_new_stream(ofmt_ctx, NULL);
  
          if (!out_stream) {
              LOG("Failed allocating output stream\n");
              goto end;
          }
  
          ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
          if (ret < 0) {
              LOG("Failed to copy context from input to output stream codec context\n");
              goto end;
          }
          out_stream->codecpar->codec_tag = 0;
      }
  
      if (!(ofmt->flags & AVFMT_NOFILE)) {
          ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
          if (ret < 0) {
              LOG("Could not open output file '%s'", out_filename);
              goto end;
          }
      }
  
      ret = avformat_write_header(ofmt_ctx, NULL);
      if (ret < 0) {
          LOG("Error occurred when writing headers\n");
          goto end;
      }
  
      ret = av_seek_frame(ifmt_ctx, -1, from_seconds * AV_TIME_BASE, AVSEEK_FLAG_ANY);
      if (ret < 0) {
          LOG("Error seek\n");
          goto end;
      }
  
      int64_t *dts_start_from;
      int64_t *pts_start_from;
      dts_start_from = (int64_t *) malloc(sizeof(int64_t) * ifmt_ctx->nb_streams);
      memset(dts_start_from, 0, sizeof(int64_t) * ifmt_ctx->nb_streams);
      pts_start_from = (int64_t *) malloc(sizeof(int64_t) * ifmt_ctx->nb_streams);
      memset(pts_start_from, 0, sizeof(int64_t) * ifmt_ctx->nb_streams);
  
      while (1) {
          AVStream *in_stream, *out_stream;
  
          ret = av_read_frame(ifmt_ctx, &pkt);
          LOG("while %d", ret);
          LOG("Packet size: %d", pkt.size);
          LOG("Packet stream: %d", pkt.stream_index);
          if (ret < 0)
              break;
  
          in_stream = ifmt_ctx->streams[pkt.stream_index];
          out_stream = ofmt_ctx->streams[pkt.stream_index];
  
          if (av_q2d(in_stream->time_base) * pkt.pts > end_seconds) {
              av_packet_unref(&pkt);
              break;
          }
  
          if (dts_start_from[pkt.stream_index] == 0) {
              dts_start_from[pkt.stream_index] = pkt.dts;
              printf("dts_start_from: %s\n", av_ts_make_string((char[AV_TS_MAX_STRING_SIZE]){0},dts_start_from[pkt.stream_index]));
          }
          if (pts_start_from[pkt.stream_index] == 0) {
              pts_start_from[pkt.stream_index] = pkt.pts;
              printf("pts_start_from: %s\n", av_ts_make_string((char[AV_TS_MAX_STRING_SIZE]){0},pts_start_from[pkt.stream_index]));
          }
  
          /* copy packet */
          pkt.pts = ::av_rescale_q_rnd(pkt.pts - pts_start_from[pkt.stream_index], in_stream->time_base, out_stream->time_base, (AVRounding) (AV_ROUND_NEAR_INF |
                                                                                                                                              AV_ROUND_PASS_MINMAX));
          pkt.dts = ::av_rescale_q_rnd(pkt.dts - dts_start_from[pkt.stream_index], in_stream->time_base, out_stream->time_base, (AVRounding) (AV_ROUND_NEAR_INF |
                                                                                                                                              AV_ROUND_PASS_MINMAX));
          if (pkt.pts < 0) {
              pkt.pts = 0;
          }
          if (pkt.dts < 0) {
              pkt.dts = 0;
          }
          pkt.duration = (int) av_rescale_q((int64_t) pkt.duration, in_stream->time_base, out_stream->time_base);
          pkt.pos = -1;
          printf("\n");
  
          ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
          if (ret < 0) {
              LOG("Error muxing packet\n");
              break;
          }
          av_packet_unref(&pkt);
      }
      free(dts_start_from);
      free(pts_start_from);
  
      av_write_trailer(ofmt_ctx);
  
      end:
      LOG("END");
  
      avformat_close_input(&ifmt_ctx);
  
      // Close output
      if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
          avio_closep(&ofmt_ctx->pb);
      avformat_free_context(ofmt_ctx);
  
      if (ret < 0 && ret != AVERROR_EOF) {
          LOG("-- Error occurred: %s\n", av_err2str(ret));
          return 1;
      }
  }
  

• Audio broadcasting using ffmpeg and rtmp protocol

  31 mai 2017, par Sachin Jose

  I have an application to broadcast video(YouTube). I am trying to implement a separate audio broadcasting feature. How do we audio broadcast using ffmpeg in rtmp protocol. I need help on ffmpeg command line arguments.

  I have something like:

  ffmpeg -i input.mp3 -re -acodec libmp3lame -ab 64k -ac 1 -ar 44100 -g 75 -qscale 21 -f flv rtmplink and key