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• FFMPEG incorrect image extension fails coding [closed]

  14 novembre 2016, par G.T.

  I have a problem using ffmpeg on images. My problem is that quite a lot of images on the internet has incorrect extension ( png files names .jpg and jpg files named .png) which makes the ffmpeg fail.

  I couldn’t find any documentation on the internet how can this be fixed ? Can I force somehow ffmpeg to try to look in other codec types too not just the ones the extension suggests ?

  To reproduce this you just need to download a png from the internet and rename it’s extension to .jpg

  And even if you use the simplest command like ffmpeg -i image.jpg it fails with :

  ffprobe.exe -i tux.jpg -report
  
  ffprobe version N-61663-g19139d8 Copyright (c) 2007-2014 the FFmpeg developers
  
    built on Mar 20 2014 22:06:17 with gcc 4.8.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      52. 67.100 / 52. 67.100
  
    libavcodec     55. 52.102 / 55. 52.102
  
    libavformat    55. 34.101 / 55. 34.101
  
    libavdevice    55. 11.100 / 55. 11.100
  
    libavfilter     4.  3.100 /  4.  3.100
  
    libswscale      2.  5.102 /  2.  5.102
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  [AVIOContext @ 00000000041614e0] Statistics: 41236 bytes read, 0 seeks
  
  [mjpeg @ 0000000004160d60] marker=db avail_size_in_buf=41068
  
  [mjpeg @ 0000000004160d60] dqt: invalid precision
  
  [mjpeg @ 0000000004160d60] marker parser used 3 bytes (20 bits)
  
  [mjpeg @ 0000000004160d60] marker=fe avail_size_in_buf=40036
  
  [mjpeg @ 0000000004160d60] marker parser used 2 bytes (16 bits)
  
  [mjpeg @ 0000000004160d60] marker=cf avail_size_in_buf=39899
  
  [mjpeg @ 0000000004160d60] mjpeg: unsupported coding type (cf)
  
  [mjpeg @ 0000000004160d60] marker parser used 0 bytes (0 bits)
  
  [mjpeg @ 0000000004160d60] marker=c3 avail_size_in_buf=39465
  
  [mjpeg @ 0000000004160d60] sof0: picture: 4697x50895
  
  [image2 @ 000000000415fee0] decoding for stream 0 failed
  
  [image2 @ 000000000415fee0] Could not find codec parameters for stream 0 (Video: mjpeg): unspecified size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  tux.jpg: End of file

  PS : I tried to increase analyzeduration and probsize if that’s not obvious but same thing happens.

• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

• FFMPEG error while receiving rtsp stream

  8 avril 2014, par Pawel Rutka

  I got error with receiving rtsp stream from IPCam Edimax IC-3030 and I don't know what to do. Can anyone help me or show me a way to solution ?

  /home/prog12# ffplay "rtsp://192.168.1.7/ipcam_h264.sdp"
  
  ffplay version 2.1.4 Copyright (c) 2003-2014 the FFmpeg developers
  
  built on Mar 22 2014 18:16:53 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
  configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-      libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable- libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame
  
  
  
  ALSA lib pulse.c:243:(pulse_connect) PulseAudio: Unable to connect: Access denied
  
  
  
   [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
   [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
   [rtsp @ 0xb0c006c0] Could not find codec parameters for stream 0 (Video: h264): unspecified size
  
   Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
   Input #0, rtsp, from 'rtsp://192.168.1.7/ipcam_h264.sdp':
  
   Metadata:
  
  title           : IPCam
  
   Duration: N/A, bitrate: 64 kb/s
  
  Stream #0:0: Video: h264, 90k tbr, 90k tbn, 180k tbc
  
  Stream #0:1: Audio: pcm_alaw, 8000 Hz, 1 channels, s16, 64 kb/s
  
  SDL_OpenAudio (1 channels): 
  
  No more channel combinations to try, audio open failed
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implementedf=0/0   
  
  [rtsp @ 0xb0c006c0] UDP timeout, retrying with TCPsq=    0B f=0/0   
  
  [rtsp @ 0xb0c006c0] method PAUSE failed: 501 Not Implemented
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   
  
  nan M-V:    nan fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0   

  Thats the output.

  EDIT

  ps ax| grep -i pulse 
  
  8904 ?        S<l></l>usr/bin/pulseaudio --start --log-target=syslog
  
  10369 pts/1    S+     0:00 grep --color=auto -i pulse

  And for new command :

  sudo ffplay -video_size 640x480 "rtsp://192.168.1.7/ipcam_h264.sdp"
  
  ffplay version 2.1.4 Copyright (c) 2003-2014 the FFmpeg developers
  
  built on Mar 22 2014 18:16:53 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
  configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-libvpx --enable-libmp3lame
  
  
  
  Option video_size not found.aq=    0KB vq=    0KB sq=    0B f=0/0