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  • Not converting flv file to mp4 file using ffmpeg

    4 octobre 2015, par Sandeep Nambiar

    I installled ffmpeg for my centos server.then i converted my flv files to mp4 by using ffmpeg.But its creating 0 byte error mp4 file.
    Conversion code

    shell_exec('/usr/local/bin/ffmpeg -i ffmpeg/test1.flv-c copy -copyts ffmpeg/test.mp4');

    ffmpeg version is

    ffmpeg version N-75682-g5fa5e73 Copyright (c) 2000-2015 the FFmpeg developers
    
  built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
    
  configuration:
    
  libavutil      55.  2.100 / 55.  2.100
    
  libavcodec     57.  3.100 / 57.  3.100
    
  libavformat    57.  2.102 / 57.  2.102
    
  libavdevice    57.  0.100 / 57.  0.100
    
  libavfilter     6.  9.101 /  6.  9.101
    
  libswscale      4.  0.100 /  4.  0.100
    
  libswresample   2.  0.100 /  2.  0.100
    
Hyper fast Audio and Video encoder
    
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfi     
    
le}...

    After run this i am getting the below output

    ffmpeg version N-75682-g5fa5e73 Copyright (c) 2000-2015 the FFmpeg developers
    
  built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
    
  configuration:
    
  libavutil      55.  2.100 / 55.  2.100
    
  libavcodec     57.  3.100 / 57.  3.100
    
  libavformat    57.  2.102 / 57.  2.102
    
  libavdevice    57.  0.100 / 57.  0.100
    
  libavfilter     6.  9.101 /  6.  9.101
    
  libswscale      4.  0.100 /  4.  0.100
    
  libswresample   2.  0.100 /  2.  0.100
    
Input #0, flv, from 'ffmpeg/test1.flv':
    
  Metadata:
    
    starttime       : 0
    
    totalduration   : 67
    
    totaldatarate   : 354
    
    bytelength      : 2954448
    
    canseekontime   : true
    
    sourcedata      : BD075F34AHM1443878448533564
    
    purl            :
    
    pmsg            :
    
    httphostheader  : r7---sn-xupn5a5u-4wge7.googlevideo.com
    
  Duration: 00:01:06.63, start: 0.000000, bitrate: 354 kb/s
    
    Stream #0:0: Video: flv1, yuv420p, 426x240, 288 kb/s, 30 fps, 30 tbr, 1k tbn                                                                                                                                                             , 1k tbc
    
    Stream #0:1: Audio: mp3, 22050 Hz, mono, s16p, 65 kb/s
    
[aac @ 0x2b4b640] The encoder 'aac' is experimental but experimental codecs are                                                                                                                                                              not enabled, add '-strict -2' if you want to use it.

  • FFmpeg buildpack for Heroku does not support ogg. Unknown encoder 'libvorbis'

    3 octobre 2015, par Matthew Slaton

    After some work, I was finally able to get the buildpack installed on Heroku. I ran into this error :

    Stream #0:0: Audio: pcm_u8 ([1][0][0][0] / 0x0001), 22050 Hz, mono, u8, 176 kb/s
    

    
2015-10-03T01:15:59.635036+00:00 app[web.1]: -t is not an input option, 
    
    keeping it for the next output; consider fixing your command line.
    
2015-10-03T01:15:59.635055+00:00 app[web.1]: Please use -q:a or -q:v, -qscale is ambiguous
    
2015-10-03T01:15:59.635479+00:00 app[web.1]: Unknown encoder 'libvorbis'

    My code is :

    ffmpeg -t 15 -i #{current_path} -acodec libvorbis -qscale 1 #{temp_path}

    The -t option is supposed to be for seconds and libvorbis is for ogg conversion.

    ffmpeg -version returns this :

    ffmpeg version git-2013-06-02-5711e4f
    
built on Jun 2 2013 07:38:40 with gcc 4.4.3 (Ubuntu 4.4.3-4ubuntu5.1)
    
configuration: --enable-shared --disable-asm --prefix=/app/vendor/ffmpeg
    
libavutil 52. 34.100 / 52. 34.100 libavcodec 55. 13.100 / 55. 13.100
    
libavformat 55. 8.102 / 55. 8.102 libavdevice 55. 2.100 / 55. 2.100
    
libavfilter 3. 74.101 / 3. 74.101 libswscale 2. 3.100 / 2. 3.100
    
libswresample 0. 17.102 / 0. 17.102

    What am I doing wrong ?

  • Audio is lost on transcoding from DTS to AAC

    18 octobre 2015, par cirko

    On a Synology drive, I’m trying to transcode audio in an .MKV file from DTS to AAC using the native ffmpeg encoder. But without any error message (at least I don’t recognize any here), the audio stream is just missing in the output files. I tried it on several different files, all with the same result, leading to the conclusion that either I’m unable to see the error or something’s wrong with FFMPEG. I used the standard FFMPEG package provided with OPKG.
    This is the output I usually get :

    CirkosDaten> ffmpeg -i encodertest.mkv -map 0 -c:v copy -c:s copy -c:a aac 
    
-strict experimental output.mkv
    
ffmpeg version 2.0.2 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jun  3 2015 06:25:48 with gcc 4.6.4 (Marvell GCC release 
    
20150204-c4af733b 64K MAXPAGESIZE ALIGN CVE-2015-0235)
    
  configuration: --prefix=/usr/syno --incdir='${prefix}/include/ffmpeg' 
    
--arch=arm --target-os=linux --cross-prefix=/usr/local/arm-marvell-linux-
    
gnueabi/bin/arm-marvell-linux-gnueabi- --enable-cross-compile --enable-
    
optimizations --enable-pic --enable-gpl --enable-shared --disable-static 
    
--enable-version3 --enable-nonfree --enable-libfaac --enable-encoders 
    
--enable-pthreads --disable-bzlib --disable-protocol=rtp --disable-
    
muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-
    
ffserver --disable-ffplay --disable-devices --disable-bzlib --disable-
    
altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-
    
libmp3lame --disable-vaapi --disable-decoder=amrnb --disable-encoder=zmbv 
    
--disable-encoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed 
    
--disable-encoder=eac3 --disable-decoder=dca --disable-decoder=eac3 
    
--disable-decoder=truehd --cc=/usr/local/arm-marvell-linux-gnueabi/bin/arm-
    
marvell-linux-gnueabi-ccache-gcc
    

    
  libavutil      52. 38.100 / 52. 38.100
    
  libavcodec     55. 18.102 / 55. 18.102
    
  libavformat    55. 12.100 / 55. 12.100
    
  libavdevice    55.  3.100 / 55.  3.100
    
  libavfilter     3. 79.101 /  3. 79.101
    
  libswscale      2.  3.100 /  2.  3.100
    
  libswresample   0. 17.102 /  0. 17.102
    
  libpostproc    52.  3.100 / 52.  3.100
    
Guessed Channel Layout for  Input Stream #0.1 : 5.1
    
Guessed Channel Layout for  Input Stream #0.2 : 5.1
    
Input #0, matroska,webm, from 'encodertest.mkv':
    
  Metadata:
    
    title           : encodingtest
    
    creation_time   : 2012-09-23 09:38:19
    
  Duration: 00:01:00.94, start: 0.000000, bitrate: 9624 kb/s
    
    Chapter #0.0: start 0.000000, end 55.347000
    
    Metadata:
    
      title           : 00:00:00.000
    
    Chapter #0.1: start 55.347000, end 60.936000
    
    Metadata:
    
      title           : 00:04:56.171
    
    Stream #0:0: Video: h264 (High), yuv420p, 1920x816 [SAR 1:1 DAR 40:17], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
    
    Metadata:
    
      title           : encodingtest
    
    Stream #0:1(ger): Audio: dts, 48000 Hz, 5.1 (default)
    
    Metadata:
    
      title           : DTS
    
    Stream #0:2(eng): Audio: dts, 48000 Hz, 5.1
    
    Metadata:
    
      title           : DTS
    
    Stream #0:3(ger): Subtitle: subrip (default)
    
    Metadata:
    
      title           : Forced Subs
    
Output #0, matroska, to 'output.mkv':
    
  Metadata:
    
    title           : encodingtest
    
    encoder         : Lavf55.12.100
    
    Chapter #0.0: start 0.000000, end 55.347000
    
    Metadata:
    
      title           : 00:00:00.000
    
    Chapter #0.1: start 55.347000, end 60.936000
    
    Metadata:
    
      title           : 00:04:56.171
    
    Stream #0:0: Video: h264 (H264 / 0x34363248), yuv420p, 1920x816 [SAR 1:1 DAR 40:17], q=2-31, 23.98 fps, 1k tbn, 1k tbc (default)
    
    Metadata:
    
      title           : encodingtest
    
    Stream #0:1(ger): Subtitle: subrip (default)
    
    Metadata:
    
      title           : Forced Subs
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (copy)
    
  Stream #0:3 -> #0:1 (copy)
    
Press [q] to stop, [?] for help
    
frame= 1461 fps=528 q=-1.0 Lsize=   54636kB time=00:01:00.81 bitrate=7360.1kbits/s    
    
video:54624kB audio:0kB subtitle:0 global headers:0kB muxing overhead 
    
0.023249%encodingtest

    When I additionally set the bitrate, the output says

    Codec AVOption b (set bitrate (in bits/s)) specified for output file #0 
    
(output.mkv) has not been used for any stream. The most likely reason is 
    
either wrong type (e.g. a video option with no video streams) or that it is 
    
a private option of some encoder which was not actually used for any stream.

    This indicates that somehow the 5.1, 48kHz DTS stream isn’t considered as an input. Why ?

    EDIT : Is it possible that DTS decoding is turned off somewhere here ? I reckoned maybe that’s why I get no errors. But I didn’t see any obvious option for that in the output.

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