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    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

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    18 février 2011, par

    Multilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
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    5 septembre 2013, par

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Sur d’autres sites (9266)

  • Encode x264 video with ffmpeg for Android with starting offset

    4 août 2013, par scubed

    I'm trying to convert a video to play on an Android device.
    The video is from a big movie. I am chopping it back into pieces
    to correspond with the actual segments of the movie using -ss and -t.

    The input is mp4 with H.264 and AAC.
    The output is mkv using H.264 and Vorbis.

    Specifically, the input is :

    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 320x240, 2240 kb/s, 29.97 fps, 60 tbr, 90k tbn, 180k tbc
    
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 162 kb/s

    I'm using : ffmpeg version 1.0.7

    The command I'm trying is something like :

    ffmpeg -ss 00:03:52.000 -i in.mp4 -t 00:01:00.000 -c:v libx264 -preset medium -crf 20 -maxrate 400k -bufsize 1835k -c:a libvorbis -sn out.mkv

    However, while the resulting video works fine on my computer, when I click on
    my phone, it says : Can't play video
    and checking the Android log, it has :

    E/SoftAVC (24319): Decoder failed: -2
    
E/OMXCodec(24319): [OMX.google.h264.decoder] ERROR(0x80001001, -1007)

    It is still able to make a thumbnail for the movie, but not play it.

    Interestingly, some simple variations of that command do work :
    Remove -ss to start at the beginning of the video
    Use -an to disable audio

    These variations still failed :
    Copying the original audio with -c:a copy, or other audio codecs like vorbis, mp3
    Using mp4 instead of mkv
    Using baseline H.264 profile, including restricting level to 1.2.

    Running through mkvmerge first not only fails, but makes Android not able to even make a thumbnail.

    I don't know if it is related, but another small thing I noticed is that for
    starting transcoding later in the movie, the audio starts out slightly out-of-sync.
    After several seconds, it gets back in sync. The audio is in sync in the original.

    Robert Rowntree :

    -vcodec libx264 -b:v 200k -bt 50k -threads 0 -b_strategy 1 -acodec copy -f mp4 -strict -2

    Interesting. Your command almost works. The video actually plays on Android. The one problem is that the audio is out-of-sync and stays out-of-sync throughout the whole clip. But, that's much closer than I've been. I'll search around there and see if I can find the right combination.

    I tried combinations of it. It appears that using both mp4 and copying the audio is what allows it to work. Using libvorbis or going to mkv breaks it again. But, I would like to transcode the audio, and I suspect to keep it in sync, I might have to transcode it anyways. Note that even with transcoding, when I play it back on the computer, I still don't have sync between audio and video.

    LordNeckbeard :
    Here is the complete log.

    ffmpeg version 1.0.7 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on Jul 27 2013 13:01:19 with gcc 4.4.5 (Gentoo 4.4.5 p1.2, pie-0.4.5)
    
  configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-mtune=athlon64 -O2 -pipe -fomit-frame-pointer -fstack-protector' --extra-cflags='-mtune=athlon64 -O2 -pipe -fomit-frame-pointer -fstack-protector' --extra-cxxflags='-mtune=athlon64 -O2 -pipe -fomit-frame-pointer -fstack-protector' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-avresample --disable-stripping --disable-debug --disable-doc --disable-vaapi --disable-runtime-cpudetect --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libx264 --enable-libxvid --enable-libcaca --enable-openal --disable-indev=v4l2 --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-libfreetype --enable-pthreads --enable-libspeex --enable-libvorbis --disable-altivec --disable-avx --disable-vis --disable-neon --cpu=athlon64 -  libavutil      51. 73.101 / 51. 73.101
    
  libavcodec     54. 59.100 / 54. 59.100
    
  libavformat    54. 29.104 / 54. 29.104
    
  libavdevice    54.  2.101 / 54.  2.101
    
  libavfilter     3. 17.100 /  3. 17.100
    
  libswscale      2.  1.101 /  2.  1.101
    
  libswresample   0. 15.100 /  0. 15.100
    
  libpostproc    52.  0.100 / 52.  0.100
    
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'in.mp4':
    
  Metadata:
    
    major_brand     : mp42
    
    minor_version   : 0
    
    compatible_brands: mp42isomavc1
    
    creation_time   : 2013-07-13 02:23:51
    
    encoder         : HandBrake 0.9.6 2012022800
    
  Duration: 03:14:01.41, start: 0.000000, bitrate: 2408 kb/s
    
    Chapter #0.0: start -0.133467, end 648.697411
    
    Metadata:
    
      title           : Chapter  1
    
    Chapter #0.1: start 648.697411, end 1297.345411
    
    Metadata:
    
      title           : Chapter  2
    
    Chapter #0.2: start 1297.345411, end 1729.777411
    
    Metadata:
    
      title           : Chapter  3
    
    Chapter #0.3: start 1729.777411, end 2378.425411
    
    Metadata:
    
      title           : Chapter  4
    
    Chapter #0.4: start 2378.425411, end 3027.073411
    
    Metadata:
    
      title           : Chapter  5
    
    Chapter #0.5: start 3027.073411, end 3675.721411
    
    Metadata:
    
      title           : Chapter  6
    
    Chapter #0.6: start 3675.721411, end 4108.153411
    
    Metadata:
    
      title           : Chapter  7
    
    Chapter #0.7: start 4108.153411, end 4756.801411
    
    Metadata:
    
      title           : Chapter  8
    
    Chapter #0.8: start 4756.801411, end 5405.449411
    
    Metadata:
    
      title           : Chapter  9
    
    Chapter #0.9: start 5405.449411, end 6054.097411
    
    Metadata:
    
      title           : Chapter 10
    
    Chapter #0.10: start 6054.097411, end 6702.745411
    
    Metadata:
    
      title           : Chapter 11
    
    Chapter #0.11: start 6702.745411, end 7135.177411
    
    Metadata:
    
      title           : Chapter 12
    
    Chapter #0.12: start 7135.177411, end 7783.825411
    
    Metadata:
    
      title           : Chapter 13
    
    Chapter #0.13: start 7783.825411, end 8432.473411
    
    Metadata:
    
      title           : Chapter 14
    
    Chapter #0.14: start 8432.473411, end 9081.121411
    
    Metadata:
    
      title           : Chapter 15
    
    Chapter #0.15: start 9081.121411, end 9513.553411
    
    Metadata:
    
      title           : Chapter 16
    
    Chapter #0.16: start 9513.553411, end 10162.201411
    
    Metadata:
    
      title           : Chapter 17
    
    Chapter #0.17: start 10162.201411, end 10810.849411
    
    Metadata:
    
      title           : Chapter 18
    
    Chapter #0.18: start 10810.849411, end 11459.497411
    
    Metadata:
    
      title           : Chapter 19
    
    Chapter #0.19: start 11459.497411, end 11641.412478
    
    Metadata:
    
      title           : Chapter 20
    
    Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 320x240, 2240 kb/s, 29.97 fps, 60 tbr, 90k tbn, 180k tbc
    
    Metadata:
    
      creation_time   : 2013-07-13 02:23:51
    
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 162 kb/s
    
    Metadata:
    
      creation_time   : 2013-07-13 02:23:51
    
    Stream #0:2(und): Subtitle: mov_text (text / 0x74786574)
    
    Metadata:
    
      creation_time   : 2013-07-13 02:23:51
    
[libx264 @ 0x14ea220] using cpu capabilities: MMX2 SSE2Slow SlowCTZ
    
[libx264 @ 0x14ea220] profile High, level 2.1
    
[libx264 @ 0x14ea220] 264 - core 120 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=60 keyint_min=6 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=20.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=400 vbv_bufsize=1835 crf_max=0.0 nal_hrd=none ip_ratio=1.40 aq=1:1.00
    
Output #0, matroska, to 'out.mkv':
    
  Metadata:
    
    major_brand     : mp42
    
    minor_version   : 0
    
    compatible_brands: mp42isomavc1
    
    encoder         : Lavf54.29.104
    
    Chapter #0.0: start 0.000000, end 60.000000
    
    Metadata:
    
      title           : Chapter  1
    
    Stream #0:0(und): Video: h264 (H264 / 0x34363248), yuv420p, 320x240, q=-1--1, 1k tbn, 60 tbc
    
    Metadata:
    
      creation_time   : 2013-07-13 02:23:51
    
    Stream #0:1(und): Audio: vorbis (oV[0][0] / 0x566F), 48000 Hz, stereo, flt
    
    Metadata:
    
      creation_time   : 2013-07-13 02:23:51
    
Stream mapping:
    
  Stream #0:0 -> #0:0 (h264 -> libx264)
    
  Stream #0:1 -> #0:1 (aac -> libvorbis)
    
Press [q] to stop, [?] for help
    
frame= 1799 fps= 92 q=-1.0 Lsize=    3738kB time=00:00:59.98 bitrate= 510.5kbits/s dup=0 drop=51    =51    
    
video:3016kB audio:683kB subtitle:0 global headers:4kB muxing overhead 0.939943%
    
[libx264 @ 0x14ea220] frame I:31    Avg QP:20.23  size: 14126
    
[libx264 @ 0x14ea220] frame P:634   Avg QP:23.03  size:  3317
    
[libx264 @ 0x14ea220] frame B:1134  Avg QP:27.71  size:   482
    
[libx264 @ 0x14ea220] consecutive B-frames:  2.3% 12.8% 84.7%  0.2%
    
[libx264 @ 0x14ea220] mb I  I16..4:  3.8% 63.8% 32.4%
    
[libx264 @ 0x14ea220] mb P  I16..4:  0.1%  0.3%  0.1%  P16..4: 47.4% 30.2% 19.5%  0.0%  0.0%    skip: 2.4%
    
[libx264 @ 0x14ea220] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8: 35.2%  3.0%  0.6%  direct: 8.8%  skip:52.3%  L0:28.7% L1:63.9% BI: 7.4%
    
[libx264 @ 0x14ea220] 8x8 transform intra:64.0% inter:59.5%
    
[libx264 @ 0x14ea220] coded y,uvDC,uvAC intra: 94.2% 99.5% 95.5% inter: 23.3% 55.5% 14.0%
    
[libx264 @ 0x14ea220] i16 v,h,dc,p: 75% 10%  5% 10%
    
[libx264 @ 0x14ea220] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 16% 12%  8%  7%  8%  8% 11% 11%
    
[libx264 @ 0x14ea220] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 20%  7%  8%  9%  9% 10% 10% 11%
    
[libx264 @ 0x14ea220] i8c dc,h,v,p: 38% 31% 14% 17%
    
[libx264 @ 0x14ea220] Weighted P-Frames: Y:7.3% UV:4.4%
    
[libx264 @ 0x14ea220] ref P L0: 48.8% 14.2% 29.1%  7.5%  0.4%
    
[libx264 @ 0x14ea220] ref B L0: 65.4% 30.8%  3.7%
    
[libx264 @ 0x14ea220] ref B L1: 89.0% 11.0%
    
[libx264 @ 0x14ea220] kb/s:411.70

  • Reduce HLS latency from +30 seconds

    4 juin 2014, par Rick

    Ubuntu 12.04

    nginx 1.2.4

    avconv -version

    avconv version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
    
  built on Feb  6 2014 20:56:59 with gcc 4.6.3
    
avconv 0.8.10-4:0.8.10-0ubuntu0.12.04.1
    
libavutil    51. 22. 2 / 51. 22. 2
    
libavcodec   53. 35. 0 / 53. 35. 0
    
libavformat  53. 21. 1 / 53. 21. 1
    
libavdevice  53.  2. 0 / 53.  2. 0
    
libavfilter   2. 15. 0 /  2. 15. 0
    
libswscale    2.  1. 0 /  2.  1. 0
    
libpostproc  52.  0. 0 / 52.  0. 0

    I’m using avconv and nginx to create an HLS stream but right now my latency is regularly well over 30 seconds. After much reading I am aware that HLS has built in latency and that 10s is expected and even preferred but 30s seems quite extreme.

    I’ve seen a lot of discussion on the nginx-rtmp google group, this thread in particular had a lot of suggestions. I have attempted to reduce solve my issue by reducing the hls_fragment and the hls_playlist_length but they haven’t had a significant effect.

    nginx.conf :

    #user  nobody;
    
worker_processes  1;
    

    
error_log  logs/error.log debug;
    

    
events {
    
    worker_connections  1024;
    
}
    

    
http {
    
    include       mime.types;
    
    default_type  application/octet-stream;
    

    
    sendfile        on;
    
    keepalive_timeout  65;
    

    
    server {
    
        listen       8888;
    
        server_name  localhost;
    

    
        add_header 'Access-Control-Allow-Origin' "*";
    

    
        location /hls {
    
            types {
    
                application/vnd.apple.mpegurl m3u8;
    
                video/mp2t ts;
    
            }
    
            root /tmp;
    
        }
    

    
        # rtmp stat
    
        location /stat {
    
            rtmp_stat all;
    
            rtmp_stat_stylesheet stat.xsl;
    
        }
    
        location /stat.xsl {
    
            # you can move stat.xsl to a different location
    
            root /usr/build/nginx-rtmp-module;
    
        }
    

    
        # rtmp control
    
        location /control {
    
            rtmp_control all;
    
        }
    

    
        error_page   500 502 503 504  /50x.html;
    
        location = /50x.html {
    
            root   html;
    
        }
    
    }
    
}
    

    
rtmp {
    
    server {
    
        listen 1935;
    
        ping 30s;
    
        notify_method get;
    

    
        application myapp {
    
            live on;
    

    
            hls on;
    
            hls_path /tmp/hls;
    
            hls_base_url http://x.x.x.x:8888/hls/;
    
            hls_sync 2ms;
    
            hls_fragment 2s;
    

    
            #hls_variant _low BANDWIDTH=160000;
    
            #hls_variant _mid BANDWIDTH=320000;
    
            #hls_variant _hi BANDWIDTH=640000;
    
        }
    
    }
    
}

    avconv command :

    avconv -r 30 -y -f image2pipe -codec:v mjpeg -i - -f flv -codec:v libx264 -profile:v baseline -preset ultrafast -tune zerolatency -an -f flv rtmp://127.0.0.1:1935/myapp/mystream

  • Error in converting audio file format from ogg to wav [on hold]

    9 juin 2014, par Sumit Bisht

    I am trying to convert an ogg format file that was created using webrtc (html5 usermedia content generated on firefox) and transferred and decoded on the server into a wav file through ffmpeg but am getting this error on cmmand line while trying to convert :

    $ ffmpeg -i 2014-6-5_16-17-54.ogg res1.wav
    
ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
    
  built on May  1 2014 13:12:12 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
    
  configuration: --enable-gpl --enable-version3 --enable-shared --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
    
  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 #0.0 : mono
    
Input #0, ogg, from '2014-6-5_16-17-54.ogg':
    
  Duration: 00:00:01.84, start: 0.000000, bitrate: 18 kb/s
    
    Stream #0:0: Audio: opus, 48000 Hz, mono
    
    Metadata:
    
      ENCODER         : Mozilla29.0.1
    
[graph 0 input from stream 0:0 @ 0x18dca20] Invalid sample format (null)
    
Error opening filters!

    Although, I am able to play the file on server and using the same command, am able to convert .ogg files generated somewhere else. What might be I missing ?

    Edit :
    Here’s the source code that is used to write to the file :

    1) During startup - use the methods of getUserMedia API.

    navigator.getUserMedia({
    
        audio: true,
    
        video: false
    
    }, function(stream) {
    
        audioStream = RecordRTC(stream, {
    
            bufferSize: 16384
    
        });
    
        audioStream.startRecording();

    2) During stopping of the recording - extracting the recorded information.

    function(audioDataURL) {
    
     var audioFile = {};
    
     audioFile = {
    
       contents: audioDataURL
    
     **strong text**};

    On server end, the following code is creating a file from this data :

    dataURL = dataURL.split(',').pop(); // dataURL is the audioDataURL as defined above
    
fileBuffer = new Buffer(dataURL, 'base64');
    
fs.writeFileSync(filePath, fileBuffer);

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