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• How to increase speed of ffmpeg stream when running in cli assuming streaming over network

  28 juin 2020, par VRu11ra

  I'd like to increase the speed of playback so that I can catch up to whatever the newest available audio packet is. Using PulseAudio on archlinux for server, client uses windows although that really shouldn't matter.

  


  Server commands issued :

  


  pactl load-module module-null-sink sink_name=remote
ffmpeg -f pulse -i "remote.monitor" -ac 2 -acodec pcm_s16le -ar 48000 -f s16le "udp://{LAN_IP_OF_CLIENT}:{PORT}"


  


  Client command issued :

  


  ffplay.exe -nodisp -ac 2 -acodec pcm_s16le -ar 48000 -analyzeduration 0 -probesize 32 -f u8 -i udp://0.0.0.0:{PORT}


  


  Current setup is using pavucontrol to put the audio output to the pactl sink from firefox and just keeping the cli application running somewhere. Often times the network is slow, and the audio will grow an increasingly noticable lag behind whatever is onscreen. When I re-execute the commands on both server and client it catches up. If possible I'd like to keep up with whatever's being broadcast- I figure the simplest solution is to nudge the playback speed a little faster than audio is being sent over so that in the mid-long term it will fix itself.

  


  If there's just a way to discard audio packets that aren't the newest ones and jump ahead when possible I'd prefer that as a solution- I know too little about ffmpeg to know if that's possible to do easily.

  


• avdevice/pulse_audio_dec : reduce default fragment size

  11 juin 2022, par Marton Balint

  avdevice/pulse_audio_dec : reduce default fragment size
  Reduces default fragment size from the pulse audio default of 2 sec to 50 ms.
  
  This also has an effect on the size of the returned frames, which will be
  
  around 50 ms as well, making timestamps more accurate.
  This should fix the regression in ticket #9776.
  Pulseaudio timestamps for monitor sources are still pretty inaccurate for me,
  
  but I don't see how else should we query latencies from the library.
  Signed-off-by : Marton Balint <cus@passwd.hu>
  

  • [DH] doc/indevs.texi
  • [DH] libavdevice/pulse_audio_dec.c

• Desktop audio falls behind when recording microphone + desktop audio + screen using ffmpeg

  15 septembre 2013, par madr

  I have put together this script for recording the microphone, the desktop audio and the screen using ffmpeg :

  DATE=`which date`
  
  RESO=2560x1440
  
  FPS=30
  
  PRESET=ultrafast
  
  DIRECTORY=$HOME/Video/
  
  FILENAME=videocast`$DATE +%d%m%Y_%H.%M.%S`.mkv
  
  
  
  ffmpeg -y -vsync 1 \
  
  -f pulse -ac 2 -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor \
  
  -f pulse -ac 1 -ar 25000 -i alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono \
  
  -filter_complex aresample=async=1,amix=duration=shortest,apad \
  
  -f x11grab -r $FPS -s $RESO -i :0.0 \
  
  -acodec libvorbis \
  
  -vcodec libx264 -pix_fmt yuv420p -preset $PRESET -threads 0 \
  
  $DIRECTORY$FILENAME

  Everything is recorded and between the screen and the microphone sound there are no issues what so ever, however the desktop audio falls behind badly.

  It begins in sync but gets worse over time during playback, also in ffplay. It does not matter what application playing sound : both Youtube-videos in the browser, desktop sounds and Rhythmbox (playing a couple of seconds of song then stops, wait and repeat) gets out of sync.

  The terminal output complain about

  "ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred" 

  and similar but I do not know what that means.

  Full terminal output here :

  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Aug 11 2013 14:52:28 with gcc 4.8.1 (GCC) 20130725 (prerelease)
  
    configuration: --prefix=/usr --disable-debug --disable-static --enable-avresample --enable-dxva2 --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libv4l2 --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-pic --enable-postproc --enable-runtime-cpudetect --enable-shared --enable-swresample --enable-vdpau --enable-version3 --enable-x11grab
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libavresample   1.  1.  0 /  1.  1.  0
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  
  Input #0, pulse, from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor':
  
    Duration: N/A, start: 0.014093, bitrate: 1536 kb/s
  
      Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
  
  Guessed Channel Layout for  Input Stream #1.0 : mono
  
  Input #1, pulse, from 'alsa_input.usb-0d8c_C-Media_USB_Headphone_Set-00-Set.analog-mono':
  
    Duration: N/A, start: 0.006172, bitrate: 400 kb/s
  
      Stream #1:0: Audio: pcm_s16le, 25000 Hz, mono, s16, 400 kb/s
  
  [x11grab @ 0x218a6e0] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 2560 height: 1440
  
  [x11grab @ 0x218a6e0] shared memory extension found
  
  Input #2, x11grab, from ':0.0':
  
    Duration: N/A, start: 1379021580.184321, bitrate: N/A
  
      Stream #2:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 2560x1440, -2147483 kb/s, 30 tbr, 1000k tbn, 30 tbc
  
  [libx264 @ 0x21ae560] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
  
  [libx264 @ 0x21ae560] profile Constrained Baseline, level 5.0
  
  [libx264 @ 0x21ae560] 264 - core 133 r2339 585324f - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0 me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
  
  Output #0, matroska, to '/home/anders/Video/videocast12092013_23.33.00.mkv':
  
    Metadata:
  
      encoder         : Lavf55.12.100
  
      Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 25000 Hz, mono, fltp
  
      Stream #0:1: Video: h264 (libx264) (H264 / 0x34363248), yuv420p, 2560x1440, q=-1--1, 1k tbn, 30 tbc
  
  Stream mapping:
  
    Stream #0:0 (pcm_s16le) -> aresample (graph 0)
  
    Stream #1:0 (pcm_s16le) -> amix:input1 (graph 0)
  
    amix (graph 0) -> Stream #0:0 (libvorbis)
  
    Stream #2:0 -> #0:1 (rawvideo -> libx264)
  
  Press [q] to stop, [?] for help
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred22.73 bitrate=10384.5kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred3.22 bitrate=10423.3kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred25.25 bitrate=11011.0kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred5.76 bitrate=11013.7kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred27.25 bitrate=11175.4kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred7.76 bitrate=11168.7kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred8.24 bitrate=11176.4kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) overrun occurred55.48 bitrate=11243.8kbits/s    
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  ALSA lib pcm.c:7843:(snd_pcm_recover) underrun occurred
  
  frame=12871 fps= 30 q=-1.0 Lsize=  542369kB time=00:07:09.31 bitrate=10349.3kbits/s    
  
  video:539762kB audio:2363kB subtitle:0 global headers:3kB muxing overhead 0.044476%
  
  [libx264 @ 0x21ae560] frame I:52    Avg QP:15.46  size:725888
  
  [libx264 @ 0x21ae560] frame P:12819 Avg QP:18.26  size: 40172
  
  [libx264 @ 0x21ae560] mb I  I16..4: 100.0%  0.0%  0.0%
  
  [libx264 @ 0x21ae560] mb P  I16..4:  2.6%  0.0%  0.0%  P16..4: 18.1%  0.0%  0.0%  0.0%  0.0%    skip:79.3%
  
  [libx264 @ 0x21ae560] coded y,uvDC,uvAC intra: 57.8% 49.8% 25.3% inter: 8.9% 8.7% 2.2%
  
  [libx264 @ 0x21ae560] i16 v,h,dc,p: 23% 29% 32% 16%
  
  [libx264 @ 0x21ae560] i8c dc,h,v,p: 45% 28% 18%  9%
  
  [libx264 @ 0x21ae560] kb/s:10306.26

  Please help me, I am really close to get this working !

  UPDATE : The desktop audio is out of sync when skipping filter_complex and microphone also, bit in a smaller amount. Using copy instead of libvorbis does not change anything either.