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• Stream RTP packets to FFMPEG [duplicate]

  21 mars 2017, par Johnathan Kanarek

  This question already has an answer here :

  • Stream RTP to FFMPEG using SDP
    
    1 answer
    

  I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream. I want to forward this RTP data to ffmpeg. I create SDP file that describes both the audio and video streams and send the packets through UDP.
  The SDP :

  v=0
  
  o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
  
  s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
  
  c=IN IP4 192.168.193.182
  
  t=0 0
  
  m=audio 33400 RTP/AVP 111
  
  a=rtpmap:111 /opus/48000
  
  a=fmtp:111 minptime=10;useinbandfec=1
  
  a=rtcp-fb:111 transport-cc
  
  a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  
  a=mid:audio
  
  a=recvonly
  
  m=video 33402 RTP/AVP 100
  
  a=rtpmap:100 /VP8/90000
  
  a=rtcp-fb:100 ccm fir
  
  a=rtcp-fb:100 nack
  
  a=rtcp-fb:100 nack pli
  
  a=rtcp-fb:100 goog-remb
  
  a=rtcp-fb:100 transport-cc
  
  a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
  
  a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  
  a=extmap:4 urn:3gpp:video-orientation
  
  a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
  
  a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
  
  a=mid:video
  
  a=recvonly
  
  a=rtcp-mux

  The command :
  ffmpeg -loglevel debug -analyzeduration 2147483647 -probesize 2147483647 -protocol_whitelist file,crypto,udp,rtp -re -vcodec vp8 -acodec opus -i test.sdp -vcodec h264 -acodec aac -y output.mp4

  The log :

  ffmpeg version 3.2
  
   Copyright (c) 2000-2016 the FFmpeg developers
  
  
  
  
  
    built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-11)
  
  
  
    configuration: --prefix=/opt/kaltura/ffmpeg-3.2 --libdir=/opt/kaltura/ffmpeg-3.2/lib --shlibdir=/opt/kaltura/ffmpeg-3.2/lib --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC -I/opt/kaltura/include' --extra-ldflags=-L/opt/kaltura/lib --disable-devices --enable-bzlib --enable-libgsm --enable-libmp3lame --enable-libschroedinger --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libx265 --enable-avisynth --enable-libxvid --enable-filter=movie --enable-avfilter --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libvpx --enable-libspeex --enable-libass --enable-postproc --enable-pthreads --enable-static --enable-shared --enable-gpl --disable-debug --disable-optimizations --enable-gpl --enable-pthreads --enable-swscale --enable-vdpau --enable-bzlib --disable-devices --enable-filter=movie --enable-version3 --enable-indev=lavfi --enable-x11grab
  
  
  
    libavutil      55. 34.100 / 55. 34.100
  
  
  
    libavcodec     57. 64.100 / 57. 64.100
  
  
  
    libavformat    57. 56.100 / 57. 56.100
  
  
  
    libavdevice    57.  1.100 / 57.  1.100
  
  
  
    libavfilter     6. 65.100 /  6. 65.100
  
  
  
    libswscale      4.  2.100 /  4.  2.100
  
  
  
    libswresample   2.  3.100 /  2.  3.100
  
  
  
    libpostproc    54.  1.100 / 54.  1.100
  
  
  
  Splitting the commandline.
  
  
  
  Reading option '-loglevel' ...
  
   matched as option 'loglevel' (set logging level) with argument 'debug'.
  
  
  
  Reading option '-analyzeduration' ...
  
   matched as AVOption 'analyzeduration' with argument '2147483647'.
  
  
  
  Reading option '-probesize' ...
  
   matched as AVOption 'probesize' with argument '2147483647'.
  
  
  
  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 'vp8'.
  
  
  
  Reading option '-acodec' ...
  
   matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'opus'.
  
  Reading option '-i' ... matched as input file with argument 'test.sdp'.
  
  Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'h264'.
  
  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 file.
  
  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 file 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 vp8.
  
  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.
  
  [sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
  
  [sdp @ 0xb1ef00] audio codec set to: (null)
  
  [sdp @ 0xb1ef00] audio samplerate set to: 44100
  
  [sdp @ 0xb1ef00] audio channels set to: 1
  
  [sdp @ 0xb1ef00] video codec set to: (null)
  
  [udp @ 0xb21940] end receive buffer size reported is 131072
  
  [udp @ 0xb21660] end receive buffer size reported is 131072
  
  [sdp @ 0xb1ef00] setting jitter buffer size to 500
  
  [udp @ 0xb21da0] end receive buffer size reported is 131072
  
  [udp @ 0xb22060] end receive buffer size reported is 131072
  
  [sdp @ 0xb1ef00] setting jitter buffer size to 500
  
  
  
  [sdp @ 0xb1ef00] Before avformat_find_stream_info() pos: 889 bytes read:889 seeks:0 nb_streams:2
  
  
  
  [vp8 @ 0xb27600] Header size larger than data provided
  
  
  
      Last message repeated 2 times
  
  [sdp @ 0xb1ef00] Non-increasing DTS in stream 1: packet 2 with DTS 0, packet 3 with DTS 0
  
  [vp8 @ 0xb27600] Header size larger than data provided
  
  
  
  ... repeats many times until I kill the socket ...
  
  
  
      Last message repeated 1 times
  
  [sdp @ 0xb1ef00] Non-increasing DTS in stream 1: packet 273 with DTS 553050, packet 274 with DTS 553050
  
  [vp8 @ 0xb27600] Header size larger than data provided
  
  
  
  received id=7199daf55e496b370e36cd1d25b1ef5b9dff6858 type=bye
  
  PeerConnection close. id=7199daf55e496b370e36cd1d25b1ef5b9dff6858
  
  -- PeerConnection.closed,  err: undefined
  
  -- peers in the room = 0
  
  [sdp @ 0xb1ef00] decoding for stream 1 failed
  
  [sdp @ 0xb1ef00] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, yuv420p): unspecified size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [sdp @ 0xb1ef00] After avformat_find_stream_info() pos: 889 bytes read:889 seeks:0 frames:584
  
  Input #0, sdp, from 'test.sdp':
  
    Metadata:
  
      title           : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0, 309, 1/90000: Audio: opus, 48000 Hz, mono, fltp
  
      Stream #0:1, 275, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc
  
  Successfully opened the file.
  
  Parsing a group of options: output file output.mp4.
  
  Applying option vcodec (force video codec ('copy' to copy stream)) with argument h264.
  
  Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac.
  
  Successfully parsed a group of options.
  
  Opening an output file: output.mp4.
  
  Matched encoder 'libx264' for codec 'h264'.
  
  
  
  [file @ 0xbc56e0]
  
  Setting default whitelist 'file,crypto'
  
  
  
  Successfully opened the file.
  
  
  
  detected 1 logical cores
  
  
  
  [graph 0 input from stream 0:1 @ 0xb1eca0]
  
  Setting 'video_size' to value '0x0'
  
  
  
  [buffer @ 0xbc54e0]
  
  Unable to parse option value "0x0" as image size
  
  
  
  [graph 0 input from stream 0:1 @ 0xb1eca0]
  
  Setting 'pix_fmt' to value '0'
  
  
  
  [graph 0 input from stream 0:1 @ 0xb1eca0]
  
  Setting 'time_base' to value '1/90000'
  
  
  
  [graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'pixel_aspect' to value '0/1'
  
  [graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'sws_param' to value 'flags=2'
  
  [graph 0 input from stream 0:1 @ 0xb1eca0] Setting 'frame_rate' to value '90000/1'
  
  [buffer @ 0xbc54e0] Unable to parse option value "0x0" as image size
  
  [buffer @ 0xbc54e0] Error setting option video_size to value 0x0.
  
  [graph 0 input from stream 0:1 @ 0xb1eca0] Error applying options to the filter.
  
  Error opening filters!
  
  [AVIOContext @ 0xbc57c0] Statistics: 0 seeks, 0 writeouts
  
  
  
  [AVIOContext @ 0xb1f8c0]
  
  Statistics: 889 bytes read, 0 seeks

  As you can see, at the beginning of the log the SDP parsed without recognizing the codecs :

  Opening an input file: test.sdp.
  
  [sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
  
  [sdp @ 0xb1ef00] audio codec set to: (null)
  
  [sdp @ 0xb1ef00] audio samplerate set to: 44100
  
  [sdp @ 0xb1ef00] audio channels set to: 1
  
  [sdp @ 0xb1ef00] video codec set to: (null)

  Then it’s trying to read the packets from the sockets.
  Only when I close the socket, ffmpeg continues to parse the SDP, this time finding the correct codec :

  Opening an input file: test.sdp.
  
  [sdp @ 0xb1ef00] Format sdp probed with size=2048 and score=50
  
  [sdp @ 0xb1ef00] audio codec set to: (null)
  
  [sdp @ 0xb1ef00] audio samplerate set to: 44100
  
  [sdp @ 0xb1ef00] audio channels set to: 1
  
  [sdp @ 0xb1ef00] video codec set to: (null)

  I suspect that the "Non-increasing DTS" and "Header size larger than data provided" errors are caused by wrong parsing of the packets due to usage with the wrong codec.

  I checked the SDP order and it seems the same as in other examples I have.

  Can someone suggest an explanation ?

  BTW, audio only works fine, but I guess it’s because of the simplicity of OPUS.

  Thanks.

• Concatenate two mp4 files using FFmpeg with Android

  4 juillet 2013, par vijay

  I need to merge video files in Android using FFmpeg.I used the code from halfninja and passed the following parameters to run method in Videokit.c.

  ffmpeg -i "concat:input1.mpg|input2.mpg" -c copy output.mpg

  It causes error when these parameters are passed from Android through JNI. Then i replaced -c copy with -vcodec copy -acodec copy.It worked fine.

  Now i tried to concat two .mp4 files / two .3gp files by using the above method and my output format is mp4.The output file is generated but it is empty(0kb).

  When i googled it out, i was asked to convert all mp4 or 3gp files into mpg and then concat it and again convert back to mp4 in the following post :

  Concatenate mp4 files in Android using halfninja ffmpeg (Refer Steven Penny's answer).

  But i have problems(only first input file is converted and the app exits) when converting and storing the intermediate file(i am calling run method thrice).Also this method stores all the input files in the device sdcard and this should not be done.

  -c copy works fine when passing parameters for ffmpeg executable(generated when ndk-build is made) through terminal but it throws an error "unrecognized option c" when called through JNI from android device.Can anyone let me know why this happens ?

  When we look at Steven's answer, we have to call the run method multiple times to covert all the input files and one more time to concat all these.Is it possible to it in a single call ?

  Is it possible to concat video files with different formats without storing the input files in device(mp4+mp4, mp4+3gp,3gp+3gp are my possible combinations of input) ?

  Any help is appreciated.

• Revision 26b6318de8 : Make get_coef_context() branchless. This should significantly speedup cost_coef

  1er juillet 2013, par Ronald S. Bultje

  Changed Paths :
   Modify /vp9/common/vp9_entropy.c

  
   Modify /vp9/common/vp9_entropy.h

  
   Modify /vp9/decoder/vp9_detokenize.c

  
   Modify /vp9/encoder/vp9_encodemb.c

  
   Modify /vp9/encoder/vp9_rdopt.c

  
   Modify /vp9/encoder/vp9_tokenize.c

  
  
  Make get_coef_context() branchless.

  This should significantly speedup cost_coeffs(). Basically what the
  patch does is to make the neighbour arrays padded by one item to
  prevent an eob check in get_coef_context(), then it populates each
  col/row scan and left/top edge coefficient with two times the same
  neighbour - this prevents a single/double context branch in
  get_coef_context(). Lastly, it populates neighbour arrays in pixel
  order (rather than scan order), so we don't have to dereference the
  scantable to get the correct neighbours.

  Total encoding time of first 50 frames of bus (speed 0) at 1500kbps
  goes from 2min10.1 to 2min5.3, i.e. a 2.6% overall speed increase.

  Change-Id : I42bcd2210fd7bec03767ef0e2945a665b851df56