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• pipe:0 : could not find codec parameters

  10 avril 2018, par Harish

  I am trying to store streaming video as a segments of 15 min,it is working fine with gst-launch-1.0 my command is

  gst-launch-1.0 -e --gst-debug-level=2 v4l2src device=/dev/video0 ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 ! \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \
  
  | /usr/bin/ffmpeg -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt
  
  -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  I changed same command to gstd,my command is

  gstd-client pipeline_create p v4l2src device=/dev/video0 ! ${textoverlay} text=${dev0} ! ${clockoverlay} ! vspmfilter outbuf-alloc=true ! \
  
  video/x-raw,width=640,height=480,framerate=15/2 ! omxh264enc target-bitrate=10485760 num-p-frames=0 \
  
  mpegtsmux name="muxer" ! fdsink fd=1 \
  
  alsasrc device=hw:0,0 latency-time=30000 buffer-time=800000 ! audioconvert ! audio/x-raw,format=S16LE,channel=2,rate=44100 ! \
  
  faac bitrate=128000 ! muxer. \ 
  
  gstd-client pipeline_play p | /usr/bin/ffmpeg  -v info -i pipe:0 -map 0 -c copy -f segment \
  
  -reset_timestamps 1 -segment_atclocktime 1 -segment_time 900 \
  
  -segment_list_type csv -segment_list_size 0 \
  
  -segment_list /home/root/ContinuousCapture/CamFiles.txt -segment_list_entry_prefix /home/root/ContinuousCapture/ \
  
  -segment_format mpegts -strftime 1 /home/root/ContinuousCapture/video_%g%m%d_%H%M%S.mpegts

  its giving me the following error

  libdir=/usr/lib --shlibdir=/usr/lib --datadir=/usr/share/ffmpeg --disable-mipsdsp --disable-mipsdspr2 --cpu=generic --pkg-config=pkg-config --enable-avcodec --enable-avdevice --enable-avfilter --enable-avformat --disable-avresample --enable-bzlib --enable-gpl --disable-libgsm --disable-indev=jack --disable-libvorbis --disable-libmp3lame --disable-openssl --enable-postproc --disable-libschroedinger --disable-sdl2 --disable-libspeex --enable-swresample --enable-swscale --enable-libtheora --disable-vaapi --disable-vdpau --disable-libvpx --enable-libx264 --enable-outdev=xv
  
  libavutil      55. 58.100 / 55. 58.100
  
  libavcodec     57. 89.100 / 57. 89.100
  
  libavformat    57. 71.100 / 57. 71.100
  
  libavdevice    57.  6.100 / 57.  6.100
  
  libavfilter     6. 82.100 /  6. 82.100
  
  libswscale      4.  6.100 /  4.  6.100
  
  libswresample   2.  7.100 /  2.  7.100
  
  libpostproc    54.  5.100 / 54.  5.100
  
  [mpegts @ 0x49450] Format mpegts detected only with low score of 2, misdetection possible!
  
  [mpegts @ 0x49450] Could not detect TS packet size, defaulting to non-FEC/DVHS
  
  pipe:0: could not find codec parameters

  Thank you

• How to tell ffmpeg to encode only the latest frames from input pipe (live streaming) ?

  12 mars 2018, par Valdir

  Similar question on another language : How to get the latest frames in ffmpeg, not the next frame

  Well, basically I receive a vp8 webm chunks through sockets and send it to ffmpeg stdin. My problem is that when the network slows down or temporarily stops, ffmpeg output freezes until the expected chunks arrives. When that happens I can see the output "as expected".

  The real problem is that for every glitch, ffmpeg starts to transcoding from the point it stopped, so let’s say I’m streaming my webcam and at 01:01 I have a network glitch. My chunks are still sent from browser, but never received. At this point the video live streaming (on client’s browser) is at let’s say 01:10, and after the network starts to work as expected, ffmpeg starts receiving all chunks at once since 01:01 and encodes it from that point.

  What I want is ffmpeg to encode only the latest chunks in order to keep in sync with the original video being broadcasted. So I want it to discard that part that lagged.

  It seems like the original problem comes from the socket connection, which sends packets in order and can delay depending on the connection.

• ffmpeg - how to use pipe with amovie parameter ?

  19 février 2018, par user1347009

  I am looking for a solution to overlay to mp3 files in which the longer one will be played in the background and the smaller one will continue to play like a loop with that.
  My current code is working without pipe :

  ffmpeg -i 'longer.mp3'  -filter_complex "amovie='small.mp3:loop=999'[s];[0][s]amix=duration=shortest" new.mp3

  This code works like a charm but what i want to achieve is this :

  curl "any-url-to-api-to-audio-file" | ffmpeg -i 'longer.mp3'  -filter_complex "amovie='pipe\:0:loop=999'[s];[0][s]amix=duration=shortest" new.mp3

  This code is not working and its giving me error as well,

  ffmpeg version N-89955-g118e1b0b33 Copyright (c) 2000-2018 the FFmpeg 
  
  developers
  
  built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-18)
  
  configuration: --enable-openssl --enable-libmp3lame
  
  libavutil      56.  7.100 / 56.  7.100
  
  libavcodec     58.  9.100 / 58.  9.100
  
  libavformat    58.  7.100 / 58.  7.100
  
  libavdevice    58.  0.101 / 58.  0.101
  
  libavfilter     7. 11.101 /  7. 11.101
  
  libswscale      5.  0.101 /  5.  0.101
  
  libswresample   3.  0.101 /  3.  0.101
  
  Input #0, mp3, from 'morning_alarm.mp3':
  
  Duration: 00:00:28.06, start: 0.025057, bitrate: 128 kb/s
  
  Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
  
  Metadata:
  
  encoder         : LAME3.97 
  
  Side data:
  
  replaygain: track gain - -9.200000, track peak - unknown, album gain - unknown, album peak - unknown, 
  
  % Total    % Received % Xferd  Average Speed   Time    Time     Time  Current
  
                               Dload  Upload   Total   Spent    Left  Speed
  
  0     0    0     0    0     0      0      0 --:--:-- --:--:-- --:--:--     
  
  0101  5088  101  5088    0     0  28080      0 --:--:-- --:--:-- --:--:--  160k
  
  Stream mapping:
  
  Stream #0:0 (mp3) -> amix:input0
  
  amix -> Stream #0:0 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  [Parsed_amovie_0 @ 0x26c76c0] Failed to avformat_open_input 'pipe:0'
  
  [AVFilterGraph @ 0x26c3d80] Error initializing filter 'amovie' with args 
  
  'pipe\:0:loop=999'
  
  Error reinitializing filters!
  
  Failed to inject frame into filter network: Invalid data found when 
  
  processing input
  
  Error while processing the decoded data for stream #0:0
  
  Conversion failed!

  All i want to do is to overlay two audio files via command line in which 1st audio file will be longer and the second one will be smaller.
  The smaller one will be repeated like a loop till the longer one play.

  Please tell me how can i do that ?