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• Optimizing FFmpeg/x264 Configuration for Ultra-Low Latency UDP Streaming [closed]

  17 décembre 2024, par pourjour

  I'm implementing a real-time screen streaming application using FFmpeg's libraries (libavcodec, libavformat) with H.264 encoding and UDP transport. While I have basic low-latency settings in place, I'm looking to minimize latency as much as possible while maintaining reasonable quality.
Here's my current encoder configuration :

  


  // Configure codec for low latency
codecContext->width = WIDTH;
codecContext->height = HEIGHT;
codecContext->time_base = AVRational{1, FPS};
codecContext->framerate = AVRational{FPS, 1};
codecContext->pix_fmt = AV_PIX_FMT_YUV420P;
codecContext->gop_size = 10;
codecContext->max_b_frames = 0;
codecContext->refs = 1;
codecContext->flags |= AV_CODEC_FLAG_LOW_DELAY;
codecContext->bit_rate = 3000000;

// x264 specific settings
av_opt_set(codecContext->priv_data, "preset", "ultrafast", 0);
av_opt_set(codecContext->priv_data, "tune", "zerolatency", 0);
av_opt_set(codecContext->priv_data, "delay", "0", 0);
av_opt_set(codecContext->priv_data, "profile", "baseline", 0);
av_opt_set(codecContext->priv_data, "x264opts",
           "no-mbtree:sliced-threads:sync-lookahead=0:rc-lookahead=0:"
           "no-scenecut:no-cabac:force-cfr", 0);


  


  For network transport, I'm using MPEGTS over UDP :

  


  QString url = QString("udp://%1:%2?pkt_size=1316").arg(targetAddress).arg(targetPort);


  


  Current issues :

  


  

  • Still experiencing 200-300ms latency Some quality degradation with
fast motion Occasional frame drops

  


  


  Questions :

  


  

  • Are there additional x264 options or FFmpeg settings I should
consider for reducing latency ?

  


  • What are the optimal GOP and bitrate
settings for balancing latency vs quality ?

  


  • Are there better muxer settings or alternative container formats I should consider ?

  


  • How can I optimize the network transport settings (packet size, buffering,
etc.) ?

  


  • Are there any tradeoffs I should be aware of with my current
configuration ?

  


  


• yt-dlp how to download a specific video using timestamps, and limiting the download resolution ?

  30 septembre 2023, par ignacM

  I am running a python program to download a video, this is the command I use :

  


  command = ['powershell.exe ffmpeg',
                   '-ss', str(start),
                   '-i', '$(yt-dlp',
                   '-f', 'bestvideo[ext=webm]',

                   '-g', '"%s")' % (url_base + video_identifier),
                   '-t', str(end - start),
                   '-c:v', 'libx264', '-c:a', 'copy', '%s' % output_filename]

command = ' '.join(command)
output = subprocess.check_output(command, shell=True, stderr=subprocess.STDOUT)


  


  This prints something like
powershell.exe ffmpeg -ss 10 -i $(yt-dlp -f bestvideo[ext=webm] -g "https://www.youtube.com/watch?v=Z15erfLqNKo") -t 10 -c:v libx264 -c:a copy C :\Users...\video.mp4

  


  This works perfectly fine. However, I tried to download a video that had very high resolution (4K) and I want to limit the resolution when downloading. I have found that placing [height<=1080] as so should work :

  &#xA;

  powershell.exe ffmpeg -ss 10 -i $(yt-dlp -f bestvideo[ext=webm][height<=1080] -g "https://www.youtube.com/watch?v=Z15erfLqNKo") -t 10 -c:v libx264 -c:a copy C :\Users...\video.mp4

  &#xA;

  However this does not work for me and gives me the error :&#xA;Error command "" returned non-zero exit status 1.&#xA;Output : b'The system cannot find the file specified.\r\n'

  &#xA;

  I also have noticed that only limiting the resolution does not work for me either :

  &#xA;

  powershell.exe ffmpeg -ss 10 -i $(yt-dlp -f bestvideo[height<=1080] -g "https://www.youtube.com/watch?v=Z15erfLqNKo") -t 10 -c:v libx264 -c:a copy C :\Users...\video.mp4

  &#xA;

  Actually, placing [height<=1080] after bestvideo never works, no matter what I do.

  &#xA;

  What could be the problem ? Or what command can I run to achieve both tasks (limit resolution and download specific timeframe (not whole video and cutting it)) ?

  &#xA;

• aac encoding with ffmpeg result in super short file

  5 janvier 2017, par Mockarutan

  So I’m having problem with AAC encoding. I’m trying to encode some synthetic sound waves, but it does not work as expected. The file I get gives just a super short sound when played in VLC. When I play is in ffplay is plays like I expect, but it says "duration : 00:00:00.05" which I suppose says 5 ms. But I encode a lot more than that, and it played more. So VLC plays a super short sound, ffplay plays a longer file (the expected length), but displays it with super short duration, what’s going on ?

  Source : http://pastebin.com/M5MKkEL3

  One of the things that looks wrong to me is this :
  If you look for the variable "audio_time", if you breakpoint it and read it every encode frame, you will get this :

  ..
  Encode frame 8 : 0.00010416666666666666
  Encode frame 9 : 0.00012500000000000000
  (and so on)

  The diff is : 0.00002085, which is a 1/1000 of a the diff I expected from 47 samples frames per second, which is what the encoder wants with 48k sample rate (48k / 1024 = 47).

  So why do I get a thousand of the expected data encoded ?

  Feel free to point anything suspicious out !

  Thanks in advance !