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• Convert an RTSP/RTMP-Livestream with G.711 audio into RTMP/RTSP with aac-audio

  31 août 2018, par Alex Fuhr

  im new at this forum and my english skills are not the best !

  I have a website where i publish the videostreams of the cameras to show what happens inside during the nesting-time live ! An guy with high IT-skills has build me a little Server for Restream it (Datarhei-Restreamer) But this guy has still no time and worse response-times...

  To my Problem : The Restreamer dont support the "G.711" Audio-Codec from the cameras and the Livestream are still without audio at the website. So, i need to convert the Livestreams (RTSP and RTMP- in H.264) so that the audio changes to "aac" or something other supported. But i have no plan how to do this. I tried it with FFMPEG but i dont find the correct commands to get the my result. There is something with an Streaming-server to send the new created stream to - i dont get it into my head to do this (i need just a stream that are viewable with VLC player and then as input for my restreamer-server, jsut the same like ca

  I want to change the source-stream into the correct codec (audio from G.711 to AAC, the rest like source) and then, put this "new" stream into my Restreamer-Server and it will work fine ! (Tested with XSplitbroadcaster, but dont runs on Raspberry, only 1 instance runable but 2 livestreams needs to be encoded at same time) And this programm has annoying bugs (endless and not removeable error-messages, but running stream)

  I have a new second raspberry that are planned as "live-encoder" for the restreamer-raspberry were the "new" streams are are going in (rtmp/rtsp-input on a graphical ui) I try it still with FFMPEG but still no result...

  Sorry about this long text with all the language-issues but im really frustrated with it because i have purchased 2 new cameras with total 450 euros just to get the livestream with sound now :(

• 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.

• FFMPEG Directshow Multiple Audio Capture [on hold]

  17 mai 2017, par putuyuwono

  Is it possible to capture multiple audio devices using ffmpeg dshow ?
  I am trying to capture my desktop along with mic and speaker audio using FFMPEG dshow. I have tried using the following command but it doesn’t work :

  ffmpeg -f dshow -i audio="Stereo Mix (Realtek High Definition Audio)" -f dshow -i audio="Microphone Array (Creative VF0800)" -f gdigrab -framerate 10 -video_size 1920x1080 -draw_mouse 1 -i desktop screen.avi

  It only captures audio from the first mentioned audio device. Am I missing some options in the above command ?