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• Latency and DAF in RTP transmissions

  24 février 2023, par jfernandz

  I'm trying to perform some tests for audio RTP transmissions to know their technical limitations. The idea is to prevent DAF effect in this kind of transmissions, I'm assuming a latency lower than 50ms will prevent it. But there is another handicap in my analysis, the RTP transmission must be over WiFi.

  


  For this tests I'm trying to transmit raw audio (not sure if skipping the encoding stage will improve latency) through ffmpeg between two different laptops, so I'm running ffmpeg in the first laptop (172.20.1.2) as :

  


  $ ffmpeg -f pulse -i 56 -c copy -f rtp rtp://172.20.1.5:10000

  


  which produces the following output :

  


  ffmpeg version n5.1.2 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from '56':
  Duration: N/A, start: 1677234050.938677, bitrate: 1536 kb/s
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Output #0, rtp, to 'rtp://172.20.1.5:10000':
  Metadata:
    encoder         : Lavf59.27.100
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536

Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=     322kB time=00:00:01.67 bitrate=1573.6kbits/s speed=1.06x


  


  I'm assuming the shown SDP is a valid one :

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:1536


  


  So I saved it in a file called ccopy.sdp on the second laptop (172.20.1.5). However, when I run ffplay in this other laptop as :

  


  $ ffplay -protocol_whitelist file,rtp,udp -i ccopy.sdp

  


  I can see there is problems with this SDP :

  


  ffplay version n5.1.2 Copyright (c) 2003-2022 the FFmpeg developers
  built with gcc 12.2.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-amf --enable-avisynth --enable-cuda-llvm --enable-lto --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-libzimg --enable-nvdec --enable-nvenc --enable-opencl --enable-opengl --enable-shared --enable-version3 --enable-vulkan
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
[sdp @ 0x7f8eec000c80] Could not find codec parameters for stream 0 (Audio: none, 0 channels): unknown codec
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Input #0, sdp, from 'ccopy.sdp':
  Metadata:
    title           : No Name
  Duration: N/A, bitrate: N/A
  Stream #0:0: Audio: none, 0 channels
Failed to open file 'ccopy.sdp' or configure filtergraph
    nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   


  


  Not sure if I'm doing something wrong or this is because of I cannot actually use pcm_s16le for an RTP transmission. Moreover ... Is there some argument for ffmpeg that I can use to improve this RTP transmission and reduce latency under 50ms.

  


  Thank you all :-)

  


  PS : When I don't use -c copy argument for ffmpeg and therefore I have this SDP

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 172.20.1.5
t=0 0
a=tool:libavformat LIBAVFORMAT_VERSION
m=audio 10000 RTP/AVP 97
b=AS:768
a=rtpmap:97 PCMU/48000/2


  


  The RTP transmission works as I expect, but with a significant DAF.

  


• Record sound with ffmpeg on ubuntu 12.04 [closed]

  27 juin 2012, par vzybilly

  I have been working for a few days on trying to get ffmpeg to record sound, a short list of what I've tried :

  #Crappy screen grab
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -f mp4 ./out
  
  #awesome screen grab, grabbing sound but non out.
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -f alsa -ac 2 -i pulse -vcodec libx264 -s "1366x768" -acodec libmp3lame -ab 128k -threads 0 -f mp4 ~/Desktop/vid
  
  #audio test, no audio in file.
  
  #ffmpeg -f alsa -ac 2 -i pulse -acodec libmp3lame -ab 128k -threads 0 -f mp3 ./test.mp3
  
  #awesome screen grab.
  
  #ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -threads 0 -sameq -an -f mp4 ~/Desktop/vid[/CODE]I'm running ubuntu 12.04 from beta(ish)

  it would be awesome if someone could help me get this to work all in one line or (the way i'm going) multiple instances of ffmpeg (screen grab, microphone, program)

  I have also tried the pavucontrol with doing the monitoring of when recording audio, but that does not help either.

  Thanks for all of your help, vzybilly 

  EDIT :
  This one crashed.

  $ ffmpeg -f alsa -ac 2 -i plughw:0,0 -f x11grab -r 100 -s 1366x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -preset ultrafast -threads 3 testVid.mkv
  
  ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers
  
    built on Jun 12 2012 16:37:58 with gcc 4.6.3
  
  *** THIS PROGRAM IS DEPRECATED ***
  
  This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
  
  [alsa @ 0x8fce240] capture with some ALSA plugins, especially dsnoop, may hang.
  
  [alsa @ 0x8fce240] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, alsa, from 'plughw:0,0':
  
    Duration: N/A, start: 433.999945, bitrate: N/A
  
      Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  [x11grab @ 0x8fde820] device: :0.0 -> display: :0.0 x: 0 y: 0 width: 1366 height: 768
  
  [x11grab @ 0x8fde820] shared memory extension  found
  
  [x11grab @ 0x8fde820] Estimating duration from bitrate, this may be inaccurate
  
  Input #1, x11grab, from ':0.0':
  
    Duration: N/A, start: 1340805516.368518, bitrate: N/A
  
      Stream #1.0: Video: rawvideo, bgra, 1366x768, -2147483 kb/s, 100 tbr, 1000k tbn, 100 tbc
  
  File 'testVid.mkv' already exists. Overwrite ? [y/N] y
  
  Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p'
  
  [buffer @ 0x8fde700] w:1366 h:768 pixfmt:bgra
  
  [avsink @ 0x8fcdf20] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out'
  
  [scale @ 0x8ff3ce0] w:1366 h:768 fmt:bgra -> w:1366 h:768 fmt:yuv420p flags:0x4
  
  [libx264 @ 0x8fdd920] lookaheadless mb-tree requires intra refresh or infinite keyint
  
  [libx264 @ 0x8fdd920] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2
  
  [libx264 @ 0x8fdd920] profile Constrained Baseline, level 4.2
  
  [libx264 @ 0x8fdd920] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - 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=3 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=1.25 aq=0
  
  Output #0, matroska, to 'archinstall4.mkv':
  
    Metadata:
  
      encoder         : Lavf53.21.0
  
      Stream #0.0: Video: libx264, yuv420p, 1366x768, q=-1--1, 1k tbn, 100 tbc
  
      Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Stream mapping:
  
    Stream #1.0 -> #0.0
  
    Stream #0.0 -> #0.1
  
  Press ctrl-c to stop encoding
  
  [alsa @ 0x8fce240] ALSA buffer xrun.
  
  [matroska @ 0x8fcd980] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 213 >= 213
  
  av_interleaved_write_frame(): Invalid argument

  Any thoughts ?

  EDIT & ANSWER :
  Got it all working with a script :

  #!/bin/bash
  
  #vzybilly
  
  #these are temp files
  
  aud="aud.mp3"
  
  vid="vid.mp4"
  
  #grab audio & pid
  
  ffmpeg -f alsa -ac 2 -i plughw:0,0 $aud &
  
  audPID=$!
  
  #grab screen & pid
  
  ffmpeg -f x11grab -s "1366x768" -r "24" -i :0.0 -threads 0 -sameq -an -f mp4 $vid &
  
  vidPID=$!
  
  #wait, till name given (that means stop)
  
  read -p "Stop by giving an Output video name?" out
  
  #stop audio and video with pids
  
  kill -n 2 $audPID
  
  kill -n 2 $vidPID
  
  echo "$out"
  
  #combine to the target output file
  
  ffmpeg -i $aud -i $vid -acodec copy -vcodec copy "$out"
  
  #purge the temp files
  
  rm $aud
  
  rm $vid

• FFMPEG parameter -sseof position is not “cutting” as expected

  9 mai 2020, par Vitalii

  I have a regular recording of the desktop screen, indefinite in time. I want to keep in memory, only the last 30 seconds. My command looks like this, where I try to cut the video of the last 30 seconds, using -sizeof -30. But the entire length of the video is saved.
  
avfoundation -i '1' it's my macbook screen

  



  ffmpeg -rtbufsize 300M -sseof -30 -f avfoundation -i '1' -c:v libx264 -preset ultrafast -tune zerolatency -crf 25 ~/Desktop/output.mkv


  



  System Macos 10.15
  
ffmpeg version 4.2.2

  



  My full log

  



  ffmpeg -rtbufsize 300M -sseof -30  -f avfoundation -i '1' -c:v libx264 -preset ultrafast -tune zerolatency -crf 25 ~/Desktop/output.mkv
ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.3 (clang-1103.0.32.59)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_3 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags=-fno-stack-check --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[AVFoundation input device @ 0x7fec8551cec0] Configuration of video device failed, falling back to default.
[avfoundation @ 0x7fec8600a200] Selected pixel format (yuv420p) is not supported by the input device.
[avfoundation @ 0x7fec8600a200] Supported pixel formats:
[avfoundation @ 0x7fec8600a200]   uyvy422
[avfoundation @ 0x7fec8600a200]   yuyv422
[avfoundation @ 0x7fec8600a200]   nv12
[avfoundation @ 0x7fec8600a200]   0rgb
[avfoundation @ 0x7fec8600a200]   bgr0
[avfoundation @ 0x7fec8600a200] Overriding selected pixel format to use uyvy422 instead.
[avfoundation @ 0x7fec8600a200] Stream #0: not enough frames to estimate rate; consider increasing probesize
Cannot use -sseof, duration of 1 not known
Input #0, avfoundation, from '1':
  Duration: N/A, start: 107388.118500, bitrate: N/A
    Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 2560x1600, 1000k tbr, 1000k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
Press [q] to stop, [?] for help
[libx264 @ 0x7fec869bee00] MB rate (16000000000) > level limit (16711680)
[libx264 @ 0x7fec869bee00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0x7fec869bee00] profile High 4:2:2, level 6.2, 4:2:2 8-bit
[libx264 @ 0x7fec869bee00] 264 - core 155 r2917 0a84d98 - H.264/MPEG-4 AVC codec - Copyleft 2003-2018 - 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=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=4 lookahead_threads=4 sliced_threads=1 slices=4 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=25.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=0
Output #0, matroska, to '/Users/vitalii-artec3d/Desktop/output.mkv':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv422p, 2560x1600, q=-1--1, 1000k fps, 1k tbn, 1000k tbc
    Metadata:
      encoder         : Lavc58.54.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
frame=  322 fps= 30 q=19.0 Lsize=     947kB time=00:00:10.80 bitrate= 718.0kbits/s speed=0.998x    
video:943kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.382723%
[libx264 @ 0x7fec869bee00] frame I:2     Avg QP:19.00  size:465715
[libx264 @ 0x7fec869bee00] frame P:320   Avg QP:19.30  size:   105
[libx264 @ 0x7fec869bee00] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 0x7fec869bee00] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.1%  0.0%  0.0%  0.0%  0.0%    skip:99.9%
[libx264 @ 0x7fec869bee00] coded y,uvDC,uvAC intra: 36.9% 24.9% 21.0% inter: 0.0% 0.1% 0.0%
[libx264 @ 0x7fec869bee00] i16 v,h,dc,p: 66% 28%  4%  2%
[libx264 @ 0x7fec869bee00] i8c dc,h,v,p: 70% 16% 12%  1%
[libx264 @ 0x7fec869bee00] kb/s:23977740.00