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• FFmpeg C Api - Reduce fps but maintain video duration

  25 mars 2015, par Justin Bradley

  Using the FFmpeg C API I’m trying to convert an input video into a video that looks like an animated gif - meaning no audio stream and a video stream of 4/fps.

  I have the decode/encode part working. I can drop the audio stream from the output file, but I’m having trouble reducing the fps. I can change the output video stream’s time_base to 4/fps, but it increases the video’s duration - basically playing it in slow mo.

  I think I need to drop the extra frames before I write them to the output container.

  Below is the loop where I read the input frames, and then write them to output container.

  Is this where I’d drop the extra frames ? How do I determine which frames to drop (I,P,B frames) ?

  while(av_read_frame(input_container, &decoded_packet)>=0) {
  
  
  
      if (decoded_packet.stream_index == video_stream_index) {
  
          len = avcodec_decode_video2(input_stream->codec, decoded_frame, &got_frame, &decoded_packet);
  
          if(len < 0) {
  
              exit(1);
  
          }
  
  
  
          if(got_frame) {
  
              av_init_packet(&encoded_packet);
  
              encoded_packet.data =  NULL;
  
              encoded_packet.size =  0;
  
  
  
              if(avcodec_encode_video2(output_stream->codec, &encoded_packet, decoded_frame, &got_frame) < 0) {
  
                  exit(1);
  
              }
  
              if(got_frame) {
  
                  if (output_stream->codec->coded_frame->key_frame) {
  
                      encoded_packet.flags |= AV_PKT_FLAG_KEY;
  
                  }
  
  
  
                  encoded_packet.stream_index = output_stream->index;
  
                  encoded_packet.pts = av_rescale_q(current_frame_num, output_stream->codec->time_base, output_stream->time_base);
  
                  encoded_packet.dts = av_rescale_q(current_frame_num, output_stream->codec->time_base, output_stream->time_base);
  
  
  
                  if(av_interleaved_write_frame(output_container, &encoded_packet) < 0) {
  
                      exit(1);
  
                  }
  
                  else {
  
                      current_frame_num +=1;
  
                  }
  
              }
  
              frame_count+=1;
  
              av_free_packet(&encoded_packet);
  
          }
  
      }
  
  }

• ffmpeg quality conversion options (video compression)

  25 septembre 2020, par Jason Hunter

  Can you provide a link, or an explanation, to the -q:v 1 argument that deals with video/image quality, and compression, in ffmpeg.

  


  Let me explain...

  


  for f in *
   do 
     extension="${f##*.}"
     filename="${f%.*}"
     ffmpeg -i "$f" -q:v 1 "$filename"_lq."$extension"
     rm -f "$f"
   done


  


  The ffmpeg for loop above compresses all images and videos in your working directory, it basically lowers the quality which results in smaller file sizes (the desired outcome).

  


  I'm most interested in the -q:v 1 argument of this for loop. The 1 in the -q:v 1 argument is what controls the amount of compression. But I can't find any documentation describing how to change this value of 1, and describing what it does. Is it a percentage ? Multiplier ? How do I adjust this knob ? Can/should I use negative values ? Integers only ? Min/max values ? etc.

  


  I started with the official documentation but the best I could find was a section on video quality, and the -q flag description is sparse.

  


  


  -frames[:stream_specifier] framecount (output,per-stream)
  
Stop writing to the stream after framecount frames.
  
.
  
-q[:stream_specifier] q (output,per-stream)
  

  
-qscale[:stream_specifier] q (output,per-stream)
  
Use fixed quality scale (VBR). The meaning of q/qscale is codec-dependent. If qscale is used without a stream_specifier then it applies only to the video stream, this is to maintain compatibility with previous behavior and as specifying the same codec specific value to 2 different codecs that is audio and video generally is not what is intended when no stream_specifier is used.

  


  


• FFMPEG Multiple outputs from BlackMagic input

  24 septembre 2018, par leo_dragons

  I’m currently needing help to achieve multiple outputs from one input.
  Right now, the outputs are set like this :

  ffmpeg -re -f decklink -i "DeckLink Mini Recorder" -y -pix_fmt yuv420p -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 128k -ar 44100 -async 1 -b:v 2300k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 1280x720 -bufsize 1000k -maxrate 3072k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_720p"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 900k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 854x480 -bufsize 400k -maxrate 1000k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_480p_hq"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 550k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 854x480 -bufsize 400k -maxrate 500k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_480p_lq"
  
  ffmpeg -re -i "rtmp://10.0.0.172:1935/Testing/live_720p" -c:v h264 -preset fast -tune zerolatency -c:a aac -ac 2 -b:a 114k -ar 44100 -async 1 -b:v 450k -g 5 -probesize 32 -framerate 30 -movflags +faststart -s 640x360 -bufsize 400k -maxrate 500k -shortest -f flv "rtmp://10.0.0.172:1935/Testing/live_360p" 

  This uses quite a lot of processing power, and also generates unnecessary latency (since I have to stream to WOWZA first, then back to FFMPEG and then back to WOWZA).

  And I want to optimize this.

  I’ve been trying several methods, but I only managed to overflow the decklink buffer. How could I solve this ?