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• Can't put another gif aside on video using ffmpeg

  5 mars 2023, par Peter.k

  I want to put 2 gifs side by side onto a video. I use code :

  


  ffmpeg -i tt108_5.mp4 -ignore_loop 0 -i e:\gals\flags\Russie.gif -ignore_loop 0 -i e:\gals\flags\Ukraine.gif -filter_complex \
"[1:v][0:v]scale2ref=oh*mdar:ih/5[ua][b];[ua]setsar=1,format=yuva420p,colorchannelmixer=aa=0.75[u];[b][u]overlay=shortest=1;\
 [2:v][0:v]scale2ref=oh*mdar:ih/5[ua2][b2];[ua2]setsar=1,format=yuva420p,colorchannelmixer=aa=0.75[u2];[b2][u2]overlay=0:200:shortest=1"\
-c:a copy -y -r 30 -b:v 2000k tt108_5f.mp4


  


  It works only for 1st gif, but the second is missing in the result. I also would like to know how to address the first gif's size to use some variables instead of arbitrary number overlay=0:200, to put them in vstack mode.

  


• FFMPEG : How to output image2 format into a tcp/udp socket ?

  5 août 2019, par Sagi Mann

  I’ve got ffmpeg to read some RTSP stream and output image2 format to stdout like so :

  ffmpeg -rtsp_transport tcp -i  "rtsp:xxxxx" -f image2 -update 1 -

  But stdout is not good enough for me.. I am trying to pass it to "push" it to some other process that I cannot "pipe" to ffmpeg due to some architecture constraints. I am running on Linux so I was hoping to simulate some tcp/udp socket via the file system e.g. /dev/somthing or similar. Alternatively, maybe it’s possible to get ffmpeg to send the image directly to a given tcp/udp address ? This didn’t work though (ffmpeg expects a file output) :

  ffmpeg -rtsp_transport tcp -i  "rtsp:xxxxx" -f image2 -update 1 "udp://localhost:3333"

  Any ideas ?
  Thanks

• FFmpeg Decklink playout audio PCM

  28 décembre 2019, par Rick Jansen

  I’m working on a ffmpeg playout application for Decklink but I’m facing some audio issues. I’ve seen other questions about this topic but none of them are currently helping.

  I’ve tried Reubens code (https://stackoverflow.com/a/15372417/12610231) with the swr_convert for playing out ffmpeg/libav frames to a Decklink board (this needs to be 16 bits PCM interleaved) but the audio sounds wrong. It sounds like it’s missing samples/ only getting half of the required samples).

  When I record the samples in a raw audio file and play it out with Audacity the timeline is half the length of the actual recording and playing the samples on double speed.

  I also tried the ’manual’ conversion (https://stackoverflow.com/a/15372417/12610231) but unfortunately, not the result I was hoping for.

  Here are some snippets of my code

  swr_ctx = swr_alloc();
  
  av_opt_set_int(swr_ctx, "in_channel_count", pAudioCodecCtx->channels, 0);
  
  av_opt_set_int(swr_ctx, "in_sample_rate", pAudioCodecCtx->sample_rate, 0);
  
  av_opt_set_int(swr_ctx, "in_channel_layout", pAudioCodecCtx->channel_layout, 0);
  
  av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", pAudioCodecCtx->sample_fmt, 0);
  
  av_opt_set_int(swr_ctx, "out_channel_count", 2, 0);
  
  av_opt_set_int(swr_ctx, "out_sample_rate", 48000, 0);
  
  av_opt_set_int(swr_ctx, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
  
  av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
  
  
  
  if (swr_init(swr_ctx))
  
  {
  
      printf("Error SWR");
  
  }

  ///

  ret = avcodec_decode_audio4(pAudioCodecCtx, pFrame, &frameFinished, &packet);
  
  
  
  if (ret < 0) {
  
      printf("Error in decoding audio frame.\n");
  
  }
  
  
  
  swr_convert(swr_ctx, (uint8_t**)&m_audioBuffer, pFrame->nb_samples, (const uint8_t *)pFrame->extended_data, pFrame->nb_samples);

  It also looks like that the FFmpeg packet contains out of 1 video packet en 2 audio packets, not sure what to do with the second audio packet, I already tried to combine the first and second audio package without any good result on the audio side.

  Any help is appreciated.