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• Pushing data to web-browser while processing input from ffmpeg

  6 septembre 2013, par Stacked

  I want to push output of this directly to the user using PHP/Python :

  wget -qO- "http://my-video-stream-input.url/here" | ffmpeg -i pipe:0 -ab 192000 -acodec libmp3lame -map_metadata -1 -vn 1378457645_myfile.mp3

  The above command takes the input stream and converts on-the-fly it to audio without waiting for full-file to download, this works perfectly at terminal. Now, I need to push the ffmpeg processed output audio to the web-browser, once again on-the-fly without completing the full transcode, like :

  Wget->ffmpeg->Web-browser in real-time

  I tried the below in PHP but this results in 0 byte file-downloads :

  $cmd = "wget -qO- "http://my-video-stream-input.url/here" | ffmpeg -i pipe:0 -ab 192000 -acodec libmp3lame -map_metadata -1 -vn 1378457645_myfile.mp3";
  
  header('Content-type: audio/mpeg');
  
  header("Content-Type: application/octet-stream");
  
  header("Content-Disposition: attachment; filename=\"1378457645_myfile.mp3\"");
  
  passthru($cmd);

  Adding 2>&1 to the $cmd shows downloads a 3.6 KB file with following

    ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Feb 22 2013 07:22:31 with gcc 4.4.5
  
    configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --extra-libs=-lgcrypt --disable-altivec --disable-armv5te --disable-armv6 --disable-vis
  
    libavutil    50. 43. 0 / 50. 43. 0
  
    libavcodec   52.123. 0 / 52.123. 0
  
    libavformat  52.111. 0 / 52.111. 0
  
    libavdevice  52.  5. 0 / 52.  5. 0
  
    libavfilter   1. 80. 0 /  1. 80. 0
  
    libswscale    0. 14. 1 /  0. 14. 1
  
    libpostproc  51.  2. 0 / 51.  2. 0
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:0':
  
    Duration: 00:02:54.75, start: 164.745578, bitrate: N/A
  
      Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16
  
      Metadata:
  
        creation_time   : 2013-08-30 05:36:13
  
  Output #0, mp3, to '1378458744_myfile.mp3':
  
    Metadata:
  
      TSSE            : Lavf52.111.0
  
      Stream #0.0(und): Audio: libmp3lame, 44100 Hz, stereo, s16, 192 kb/s
  
      Metadata:
  
        creation_time   : 2013-08-30 05:36:13
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  size=     134kB time=00:00:05.69 bitrate= 192.3kbits/s    
  
  size=     263kB time=00:00:11.23 bitrate= 192.1kbits/s    
  
  size=     386kB time=00:00:16.45 bitrate= 192.1kbits/s    
  
  size=     515kB time=00:00:21.96 bitrate= 192.1kbits/s    
  
  size=     637kB time=00:00:27.16 bitrate= 192.1kbits/s    
  
  size=     765kB time=00:00:32.62 bitrate= 192.0kbits/s    
  
  size=     884kB time=00:00:37.69 bitrate= 192.0kbits/s    
  
  size=    1011kB time=00:00:43.12 bitrate= 192.0kbits/s    
  
  size=    1134kB time=00:00:48.37 bitrate= 192.0kbits/s    
  
  size=    1253kB time=00:00:53.47 bitrate= 192.0kbits/s    
  
  size=    1379kB time=00:00:58.82 bitrate= 192.0kbits/s    
  
  size=    1508kB time=00:01:04.31 bitrate= 192.0kbits/s    
  
  size=    1632kB time=00:01:09.64 bitrate= 192.0kbits/s    
  
  size=    1758kB time=00:01:14.99 bitrate= 192.0kbits/s    
  
  size=    1883kB time=00:01:20.35 bitrate= 192.0kbits/s    
  
  size=    2010kB time=00:01:25.76 bitrate= 192.0kbits/s    
  
  size=    2141kB time=00:01:31.35 bitrate= 192.0kbits/s    
  
  size=    2265kB time=00:01:36.65 bitrate= 192.0kbits/s    
  
  size=    2389kB time=00:01:41.92 bitrate= 192.0kbits/s    
  
  size=    2515kB time=00:01:47.31 bitrate= 192.0kbits/s    
  
  size=    2637kB time=00:01:52.50 bitrate= 192.0kbits/s    
  
  size=    2767kB time=00:01:58.04 bitrate= 192.0kbits/s    
  
  size=    2888kB time=00:02:03.21 bitrate= 192.0kbits/s    
  
  size=    3017kB time=00:02:08.70 bitrate= 192.0kbits/s    
  
  size=    3142kB time=00:02:14.06 bitrate= 192.0kbits/s    
  
  size=    3266kB time=00:02:19.33 bitrate= 192.0kbits/s    
  
  size=    3391kB time=00:02:24.66 bitrate= 192.0kbits/s    
  
  size=    3518kB time=00:02:30.07 bitrate= 192.0kbits/s    
  
  size=    3650kB time=00:02:35.71 bitrate= 192.0kbits/s    
  
  size=    3778kB time=00:02:41.20 bitrate= 192.0kbits/s    
  
  size=    3862kB time=00:02:44.78 bitrate= 192.0kbits/s    
  
  video:0kB audio:3862kB global headers:0kB muxing overhead 0.004804%

• FFMpeg - Merge multiple rtmp stream inputs to a single rtmp output

  5 septembre 2013, par Paulo Miguel Almeida

  I'm trying to combine/merge two rtmp streams and then publish 'em to another stream

  Ex. :

  ffmpeg -i rtmp://ip:1935/live/micMyStream7 -i rtmp://ip:1935/live/MyStream7  -strict -2  -f flv rtmp://ip:1935/live/bcove7

  The scenario is the following, I got a stream which comes from an user's microphone that
  is the first one (micMyStream7) and I also got a stream from another user but this one has audio and video(MyStream7).

  As they are talking to each other when a user is speaking, the other one would only be listening to and vice versa.

  My idea is to set up a third stream called (bcove) which would "merge" both of them so that I could have spectators who would only be listening to the entire conversation between them.

  This is the log that ffmpeg printed although I couldn't recognize any message which helped me out.

  paulo@paulo-desktop:~$ ffmpeg -re -i rtmp://ip:1935/live/micMyStream7 -i rtmp://ip:1935/live/MyStream7  -strict -2  -f flv rtmp://ip:1935/live/bcove7
  
  ffmpeg version N-56029-g2ffead9 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Sep  4 2013 11:05:57 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1)
  
    configuration: 
  
    libavutil      52. 43.100 / 52. 43.100
  
    libavcodec     55. 31.100 / 55. 31.100
  
    libavformat    55. 16.100 / 55. 16.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 83.102 /  3. 83.102
  
    libswscale      2.  5.100 /  2.  5.100
  
    libswresample   0. 17.103 /  0. 17.103
  
  Input #0, flv, from 'rtmp://ip:1935/live/micMyStream7':
  
    Metadata:
  
      author          : 
  
      copyright       : 
  
      description     : 
  
      keywords        : 
  
      rating          : 
  
      title           : 
  
      presetname      : Medium Bandwidth (300 Kbps) - VP6
  
      creationdate    : Wed Sep  4 16:41:52 2013
  
                      : 
  
      videodevice     : Built-in iSight
  
      videokeyframe_frequency: 5
  
      audiodevice     : External microphone
  
      audiochannels   : 1
  
      audioinputvolume: 75
  
    Duration: N/A, start: 0.000000, bitrate: 253 kb/s
  
      Stream #0:0: Video: vp6f, yuv420p, 320x240, 204 kb/s, 44.83 tbr, 1k tbn, 1k tbc
  
      Stream #0:1: Audio: mp3, 22050 Hz, mono, s16p, 49 kb/s
  
  Input #1, flv, from 'rtmp://ip:1935/live/MyStream7':
  
    Metadata:
  
      author          : 
  
      copyright       : 
  
      description     : 
  
      keywords        : 
  
      rating          : 
  
      title           : 
  
      presetname      : Custom
  
      creationdate    : Wed Sep  4 12:02:24 2013
  
                      : 
  
      videodevice     : FaceTime HD Camera (Built-in)
  
      videokeyframe_frequency: 5
  
      audiodevice     : Internal microphone
  
      audiochannels   : 1
  
      audioinputvolume: 75
  
    Duration: N/A, start: 0.000000, bitrate: 253 kb/s
  
      Stream #1:0: Video: vp6f, yuv420p, 320x240, 204 kb/s, 45.08 tbr, 1k tbn, 1k tbc
  
      Stream #1:1: Audio: mp3, 22050 Hz, mono, s16p, 49 kb/s
  
  Output #0, flv, to 'rtmp://ip:1935/live/bcove7':
  
    Metadata:
  
      author          : 
  
      copyright       : 
  
      description     : 
  
      keywords        : 
  
      rating          : 
  
      title           : 
  
      presetname      : Medium Bandwidth (300 Kbps) - VP6
  
      creationdate    : Wed Sep  4 16:41:52 2013
  
                      : 
  
      videodevice     : Built-in iSight
  
      videokeyframe_frequency: 5
  
      audiodevice     : External microphone
  
      audiochannels   : 1
  
      audioinputvolume: 75
  
      encoder         : Lavf55.16.100
  
      Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 320x240, q=2-31, 200 kb/s, 1k tbn, 44.83 tbc
  
      Stream #0:1: Audio: adpcm_swf ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 88 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (vp6f -> flv)
  
    Stream #0:1 -> #0:1 (mp3 -> adpcm_swf)
  
  Press [q] to stop, [?] for help
  
  [mp3 @ 0x3625ec0] overread, skip -9 enddists: -3 -300:14.44 bitrate= 224.0kbits/s    
  
  [mp3 @ 0x3625ec0] overread, skip -7 enddists: -3 -30:26.39 bitrate= 203.5kbits/s  

  Thanks in advance

• RED5 1.0.2 recorded FLV convert to Mobile/HTML5 format with FFMPEG av out of sync

  14 août 2014, par Daew dawe

  I have problem with recorded video in Red5 v1.0.2 (i had issues with previous v1, it does not recorded any video, with 1.0.2 it works). When I record flv I want to convert it to some mp4. But I have problem with settings, because every time there is some issue with quality/audio sync. Can u please help me how to convert with ffmpeg (in future automatic process on server).

  Second problem is that in flash client buffer length is always 0, but in v0.8 it was filled and on end I waited until empty, here I’m not sure how long should I wait. I founded this url http://code.google.com/p/red5/issues/detail?id=312 where they said to wait until i get UnPublish.Success, but that event I got only after ns.close()

  My flash client record settings is (FP10) :

  video :

  • resolution = 640x360
  • fps = 30
  • keyframeinterval = 15
  • video quality = 90
  • bandwidth = 0

  audio :

  • microphone codec = SPEEX
  • encodeQuality = 9

  • silencelevel = 0

  • bufferTime = 15

  recorded video parameters in VLC (translated from czech to english) :

  video

  • Codec : Flash Video (FLV1)
  • Resolution : 640x360
  • format : Planar 4:2:0 YUV

  audio

  • codec : Speex Audio (spx )
  • frequency : 16000Hz
  • bits per sample : 16
  • data flow : 16 kb/s

  FFMEPG info about video :

   Metadata:
  
     server          : Red5 Server 1.0.2 Rev: 4616
  
     creationdate    : Mon Sep 02 23:17:08 CEST 2013
  
     canSeekToEnd    : true
  
   Duration: 00:00:33.24, start: 0.000000, bitrate: 645 kb/s
  
     Stream #0:0: Video: flv1, yuv420p, 640x360, 625 kb/s, 1k tbr, 1k tbn, 1k tbc
  
  
  
     Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 16 kb/s

  bsplayer showing 25fps - but I recorded 30fps, I dont understand this so much.

  what I tried with ffmpeg (I’m ffmpeg newbie).

  First I recorded 33sec long video

  when I convert audio with command : ffmpeg -i test.flv -ar 44100 -ab 160k -ac 1 output.mp3 , then the audio have only 30sec

  I tried this commands, but no one with good solution

  ffmpeg -i test.flv -vcodec mpeg4 -acodec libvo_aacenc output.mp4
  
  
  
  ffmpeg -i test.flv -acodec libvo_aacenc -aq 200 outputsss.mp4
  
  
  
  ffmpeg -i test.flv -c:v libvpx -c:a libvorbis output.webm // here is sound synced good - but sound have repeating silence lags (every 1-2s)

  really thank you for your help, I’m fighting with conversion many days :(