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• Processing WebRTC RTC stream in node js server with ffmpeg

  23 juillet 2020, par Dave B

  I am trying to build a video chat app where I am creating an RTCPeerConnection and creating an offer and saving the SDP in a file. I want the SDP I sent to server should be sent to RTMP server like Nginx RTMP. Here is the offer SDP I am getting

  


  v=0
o=- 3688975056307578818 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1 2
a=msid-semantic: WMS AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 284f316d-6c5d-4283-af4d-86d44803807d
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2317617486 cname:fpKjO3/hiHYYBw7w
a=ssrc:2317617486 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 284f316d-6c5d-4283-af4d-86d44803807d
a=ssrc:2317617486 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:2317617486 label:284f316d-6c5d-4283-af4d-86d44803807d
m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 122 127 121 125 107 108 109 124 120 123 119 114 115 116
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:122 rtx/90000
a=fmtp:122 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 967462980 1884395933
a=ssrc:967462980 cname:fpKjO3/hiHYYBw7w
a=ssrc:967462980 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:967462980 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:967462980 label:7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:1884395933 cname:fpKjO3/hiHYYBw7w
a=ssrc:1884395933 msid:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq 7106bf9f-3f79-4f24-959b-b82b603a7acc
a=ssrc:1884395933 mslabel:AqaeHB0L8drTd5r0qQnzCSeYVf4bHFaVqfrq
a=ssrc:1884395933 label:7106bf9f-3f79-4f24-959b-b82b603a7acc
m=application 9 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 0.0.0.0
a=ice-ufrag:Xtid
a=ice-pwd:jE3iBRpWqFaIN3UJVOAh0G/1
a=ice-options:trickle
a=fingerprint:sha-256 48:E4:36:A6:24:66:F6:40:0F:93:9C:AB:C9:93:DF:C7:0F:D1:21:F5:9E:F7:FA:A8:58:84:1F:68:A1:61:B6:0F
a=setup:actpass
a=mid:2
a=sctp-port:5000
a=max-message-size:262144


  


  this is the ffmpeg command

  


  ffmpeg -protocol_whitelist rtp,udp,file -loglevel trace -analyzeduration 300M -probesize 300M -i test.sdp -c:v copy -c:a aac -ar 16k -ac 1 -preset ultrafast -tune zerolatency rtmp://127.0.0.1/live/1234


  


  Also trying this

  


  ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4


  


  FFMpeg gives an error like this

  


  ffmpeg version 4.2.2 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.2.2_2 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --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
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp'.
Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libvpx'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'opus'.
Reading option '-i' ... matched as input url with argument 'test.sdp'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'output.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url test.sdp.
Applying option re (read input at native frame rate) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument libvpx.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument opus.
Successfully parsed a group of options.
Opening an input file: test.sdp.
[NULL @ 0x7f949000ac00] Opening 'test.sdp' for reading
[sdp @ 0x7f949000ac00] Format sdp probed with size=2048 and score=50
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 48000
[sdp @ 0x7f949000ac00] audio channels set to: 2
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: opus
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: adpcm_g722
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_mulaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: pcm_alaw
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 48000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 32000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 16000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] audio codec set to: (null)
[sdp @ 0x7f949000ac00] audio samplerate set to: 8000
[sdp @ 0x7f949000ac00] audio channels set to: 1
[sdp @ 0x7f949000ac00] video codec set to: vp8
    Last message repeated 20 times
[udp @ 0x7f948fc07200] bind failed: Address already in use
[AVIOContext @ 0x7f948fe1c4c0] Statistics: 5985 bytes read, 0 seeks
test.sdp: Invalid data found when processing input


  


  Can anyone please point out what I am doing wrong here ? or I am in the right path ?
Please help !

  


• How do you make a modern bleep censor ?

  2 août 2023, par nick carraway

  I have always been a fan of live caller radio shows. These (sports) shows allow callers to call a hotline and talk to the radio host directly.

  


  One of the oldest problems with this show format is what if the caller curses, or says something highly inappropriate ? To keep the show clean (and legal), the radio shows broadcast with a 7 second delay. They also use a "bleep" censor, which historically allowed them to wipe the incriminating phrase with a "Beeeeeep" sound. These days, however, they completely cut out the caller's sentence before it even begins. ("Ah, we had to let you go there pal. Can't say that on the radio"). In the modern method, the transition is seamless, almost like they shorten the 7 second delay to a 4 second delay as they remove the start of the caller's reply entirely, and overwrite it with the host's explanations. The caller does not appear to be "interrupted" at all, the start of their sentence leading to the bad phrase is never even broadcasted.

  


  I've been thinking about how to do this in software. I found a project that looked promising. It adds a 7 second delay to your streams, and allows you to convert X amount of those seconds into silence assuming a caller says something inappropriate. While not ideal (since it's a few seconds of dead silence and would interrupt the caller mid-sentence), how can you do something like this in ffmpeg ? It is a good starting point before implementing the more modern features.

  


  

  1. How do you use ffmpeg to livestream a video/audio stream with a delay ?

  


  2. How do you overwrite the last 3 seconds of that stream with silence or a "bleep", when you need to ?

  


  3. Are you able to easily switch your stream to overwrite those 3 seconds with new audio (e.g. the host's explanation for why the caller was hung-up on) ? And how can you go back from a 4 second delay to a 7 second delay ?

  


  


  OR, is there a ready-made software to get flawless "radio-like" hang-ups on bad callers ?

  


• FFmpeg autogen in combination with openh264

  23 juin 2018, par grill2010

  I’m using FFmpeg autogen to stream images captured from the PC in realtime (60 or 30 fps) to an Android application. Everything is working perfectly as I used the h.264 encoding example provided by FFmpeg autogen.

  As far as I have understood FFmpeg autogen is under the LGPL licence and if you are using custom FFmpeg libs compiled under LGPL licence like for example ffmpeg-win32 you can dynamically link to the FFmpeg autogen library without violating the LGPL licence if your project is for example closed source.

  So I thought everything would be okay but then I figured out the the h.264 coded is patented in the US and MPEG LA represents the patent holders of AVC/H.264 technologies. If you are using it for free videos you can use h.264 codecs without paying royalties. Nevertheless, my Windows program which is using the encoder is free but my Android application is not free so I guess I have to pay royalties.

  I have a few questions regarding this issue :

  1. Which encoder does the FFmpeg autogen use, because as far as I know x264 is under GPL licence. I assume that it will use some internal FFmpeg software encoder ?

  2. You can’t use any built in standard encoder from FFmpeg (x264 etc.) in projects which don’t offer their encoded video stream for free without the need of paying royalties, right ?

  3. After some researches I found openh264. This library is built by cisco and they will pay all the royalties but you are not allowed to bundle it with your application, it must be downloaded during the installation process. It also works with FFmpeg but how to I get it to work with FFmpeg autogen ? Do I need a custom FFmpeg build which are using the openh264 lib from cisco ?

  I know this is not a lawyer forum but maybe someone has already dealt with all this issues or could at least answer the questions about FFmpeg autogen. I’ve searched a while now on the internet but unfortunately it seems that the licencing mechanism with h.264 is not so easy to understand.