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• How to improve quality and latency in FFmpeg

  3 août 2020, par Andrew

  I'm trying to stream FFmpeg over the new SRT protocol that's supported as an output. For now though, I'm sending it in the form of FFmpeg -> UDP -> SRT -> SRT -> UDP -> MPV/FFmpeg.

  


  Somewhere along the line, the quality degrades sharply, and latency increases by quite a bit. This seemed to happen when adding audio. Meaning, if streaming just video, quality is decent and latency is low. If streaming just audio, quality is great but latency is high.

  


  Not sure where I'm going wrong with this, so any help would be appreciated. The main focus points is high quality low latency.

  


  ---

  


  Recording video through :

  


  ./ffmpeg.exe -rtbufsize 2147M -f dshow -i video="":audio="" -flush_packets 0 -preset medium -tune zerolatency -f mp4 -b:v 6M -g 30 -f mpegts udp://127.0.0.1:9001?pkt_size=1316


  


  or

  


  ./ffmpeg.exe -f dshow -analyzeduration 200k -probesize 6M -i video="":audio="" -flush_packets 0 -preset fast -tune zerolatency -b:v 5M -b:a 384K -c:a libopus -g 25 -f mpegts udp://127.0.0.1:9001?pkt_size=1316


  


  ---

  


  Then when taking it back at the receiver side, I use one of :

  


  ./ffmpeg.exe -i udp://127.0.0.1:9001 -c copy -bufsize 32M -f mpegts udp://127.0.0.1:9002?bitrate=26214400
./mpv.exe udp://127.0.0.1:9002


  


  or

  


  ./mpv.exe --no-cache --untimed --no-demuxer-thread --video-sync=audio --vd-lavc-threads=1 udp://127.0.0.1:9002


  


  ---

  


  What am I missing ? As is quality is generally terrible and latency can get up to 10 seconds. Is it just not possible with this set up ?

  


• Merge commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’

  23 mars 2017, par Clément Bœsch

  Merge commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’
  * commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’ : (23 commits)
  
    vp9lpf/x86 : make filter_16_h work on 32-bit.
  
    vp9lpf/x86 : make filter_48/84/88_h work on 32-bit.
  
    vp9lpf/x86 : make filter_44_h work on 32-bit.
  
    vp9lpf/x86 : make filter_16_v work on 32-bit.
  
    vp9lpf/x86 : make filter_48/84_v work on 32-bit.
  
    vp9lpf/x86 : make filter_88_v work on 32-bit.
  
    vp9lpf/x86 : make filter_44_v work on 32-bit.
  
    vp9lpf/x86 : save one register in SIGN_ADD/SUB.
  
    vp9lpf/x86 : store unpacked intermediates for filter6/14 on stack.
  
    vp9lpf/x86 : move variable assigned inside macro branch.
  
    vp9lpf/x86 : simplify ABSSUM_CMP by inverting the comparison meaning.
  
    vp9lpf/x86 : remove unused register from ABSSUB_CMP macro.
  
    vp9lpf/x86 : slightly simplify 44/48/84/88 h stores.
  
    vp9lpf/x86 : make cglobal statement more conservative in register allocation.
  
    vp9lpf/x86 : save one register in loopfilter surface coverage.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_44_16_sse2,ssse3,avx.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_h_48,84_16_sse2,ssse3,avx().
  
    vp9lpf/x86 : add an SSE2 version of vp9_loop_filter_[vh]_88_16
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_88_16_ssse3,avx.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_16_16_sse2().
  
    ...
  All these commits are cherry-picks from FFmpeg. Maybe some slight
  
  differences sneaked in but the Libav codebase still differs too much
  
  with our own to make a proper diff. This merge is a noop.
  Merged-by : Clément Bœsch <u@pkh.me>
  

• Convert encoded audio file to text with signal values

  15 août 2017, par Niccolò Cirone

  I’ve been programming in c for the first time with audio files. I found this code which supposedly should read an audio file and then write a csv file containing several info in order to analyse the audio waves,that in case will be a simple voice : i’m interested in amplitude of the waves, in the timbre of the voice and its hight and extension.

             main () {   
  
             // Create a 20 ms audio buffer (assuming Fs = 44.1 kHz)
  
             int16_t buf[N] = {0}; // buffer
  
             int n;                // buffer index
  
  
  
            // Open WAV file with FFmpeg and read raw samples via the pipe.
  
            FILE *pipein;
  
            pipein = popen("ffmpeg -i whistle.wav -f s16le -ac 1 -", "r");
  
            fread(buf, 2, N, pipein);
  
            pclose(pipein);
  
  
  
            // Print the sample values in the buffer to a CSV file
  
            FILE *csvfile;
  
            csvfile = fopen("samples.csv", "w");
  
            for (n=0 ; n<n></n>code>

  Could someone explain me in detail how can I read an audio file so that I could extract from it the info I need ? Referring to this code, could someone explain me the meaning of the pipe at line 8

  pipein = popen("ffmpeg -i whistle.wav -f s16le -ac 1 -", "r");

  p.s. I already know how to read the header of the audio file, which contains a lot of useful info, but I also want to analyse the entire audio file, sample by sample.