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Autres articles (23)

  • Qu’est ce qu’un éditorial

    21 juin 2013, par

    Ecrivez votre de point de vue dans un article. Celui-ci sera rangé dans une rubrique prévue à cet effet.
    Un éditorial est un article de type texte uniquement. Il a pour objectif de ranger les points de vue dans une rubrique dédiée. Un seul éditorial est placé à la une en page d’accueil. Pour consulter les précédents, consultez la rubrique dédiée.
    Vous pouvez personnaliser le formulaire de création d’un éditorial.
    Formulaire de création d’un éditorial Dans le cas d’un document de type éditorial, les (...)

  • Contribute to translation

    13 avril 2011

    You can help us to improve the language used in the software interface to make MediaSPIP more accessible and user-friendly. You can also translate the interface into any language that allows it to spread to new linguistic communities.
    To do this, we use the translation interface of SPIP where the all the language modules of MediaSPIP are available. Just subscribe to the mailing list and request further informantion on translation.
    MediaSPIP is currently available in French and English (...)

  • HTML5 audio and video support

    13 avril 2011, par

    MediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
    The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
    For older browsers the Flowplayer flash fallback is used.
    MediaSPIP allows for media playback on major mobile platforms with the above (...)

Sur d’autres sites (5444)

  • IE11 not playing mp4 file

    11 juin 2014, par John Qualis

    I am using ffmpeg to convert a freely available public RTSP stream to a mp4 file. I can play the file quite well in Chrome using a standard HTML5 video client on a windows 7 machine but not in IE11. Any ideas why the mp4 will not play in IE11 or WMP ?

    ffmpeg -i rtsp://dmzosx001.dpa.act.gov.au/medium -acodec copy
    -vcodec copy -f mp4 -movflags frag_keyframe+empty_moov
    -min_frag_duration 1000 -reset_timestamps 1 -vsync 1
    -flags global_header -bsf:v dump_extra -y output.mp4
  • Correct command to transmit audio to ip camera using ffmpeg ?

    4 novembre 2016, par the_naive

    So I found some hints in this discussion on the correct command to transmit audio to Axis IP camera through using ffmpeg in windows, but still I have not managed to successfully transmit audio to the camera.

    The command I’m using is the following :

    ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://oper
    ator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -multiple_requests 1 -reconnect_at_eof 1 -reconnect_streamed 1 -content_type "audio/basic" -report

    The ouput I get following this command is the following :

    ffmpeg started on 2016-11-04 at 17:32:13
    Report written to "ffmpeg-20161104-173213.log"
    Command line:
    ffmpeg -v debug -y -re -f dshow -i "audio=Microphone (2- High Definition Audio Device)" -c:a pcm_mulaw -ac 1 -ar 16000 -b:a 128k -f flv http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi -content_type audio/basic -multiple_requests 1 -reconnect 1 -reconnect_at_eof 1 -reconnect_streamed 1 -report
    ffmpeg version N-82225-gb4e9252 Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 5.4.0 (GCC)
     configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
     libavutil      55. 35.100 / 55. 35.100
     libavcodec     57. 66.101 / 57. 66.101
     libavformat    57. 57.100 / 57. 57.100
     libavdevice    57.  2.100 / 57.  2.100
     libavfilter     6. 66.100 /  6. 66.100
     libswscale      4.  3.100 /  4.  3.100
     libswresample   2.  4.100 /  2.  4.100
     libpostproc    54.  2.100 / 54.  2.100
    Splitting the commandline.
    Reading option '-v' ... matched as option 'v' (set logging level) with argument 'debug'.
    Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
    Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
    Reading option '-f' ... matched as option 'f' (force format) with argument 'dshow'.
    Reading option '-i' ... matched as input file with argument 'audio=Microphone (2- High Definition Audio Device)'.
    Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'pcm_mulaw'.
    Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
    Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
    Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '128k'.
    Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
    Reading option 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi' ... matched as output file.
    Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/basic'.
    Reading option '-multiple_requests' ... matched as AVOption 'multiple_requests' with argument '1'.
    Reading option '-reconnect' ... matched as AVOption 'reconnect' with argument '1'.
    Reading option '-reconnect_at_eof' ... matched as AVOption 'reconnect_at_eof' with argument '1'.
    Reading option '-reconnect_streamed' ... matched as AVOption 'reconnect_streamed' with argument '1'.
    Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
    Trailing options were found on the commandline.
    Finished splitting the commandline.
    Parsing a group of options: global .
    Applying option v (set logging level) with argument debug.
    Applying option y (overwrite output files) with argument 1.
    Applying option report (generate a report) with argument 1.
    Successfully parsed a group of options.
    Parsing a group of options: input file audio=Microphone (2- High Definition Audio Device).
    Applying option re (read input at native frame rate) with argument 1.
    Applying option f (force format) with argument dshow.
    Successfully parsed a group of options.
    Opening an input file: audio=Microphone (2- High Definition Audio Device).
    [dshow @ 00000000000279e0] Selecting pin Capture on audio only
    dshow passing through packet of type audio size    88200 timestamp 310221040000 orig timestamp 310221040000 graph timestamp 310226130000 diff 5090000 Microphone (2- High Definition Audio Device)
    [dshow @ 00000000000279e0] All info found
    Guessed Channel Layout for Input Stream #0.0 : stereo
    Input #0, dshow, from 'audio=Microphone (2- High Definition Audio Device)':
     Duration: N/A, start: 31022.104000, bitrate: 1411 kb/s
       Stream #0:0, 1, 1/10000000: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
    Successfully opened the file.
    Parsing a group of options: output file http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
    Applying option c:a (codec name) with argument pcm_mulaw.
    Applying option ac (set number of audio channels) with argument 1.
    Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
    Applying option b:a (video bitrate (please use -b:v)) with argument 128k.
    Applying option f (force format) with argument flv.
    Successfully parsed a group of options.
    Opening an output file: http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi.
    [http @ 0000000001c94040] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
    [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1

    Transfer-Encoding: chunked

    User-Agent: Lavf/57.57.100

    Accept: */*

    Expect: 100-continue

    Connection: close

    Host: 10.10.210.2

    Icy-MetaData: 1




    [http @ 0000000001c94040] request: POST /axis-cgi/audio/transmit.cgi HTTP/1.1

    Transfer-Encoding: chunked

    User-Agent: Lavf/57.57.100

    Accept: */*

    Connection: close

    Host: 10.10.210.2

    Icy-MetaData: 1

    Authorization: Digest username="operator", realm="AXIS_ACCC8E027F47", nonce="0EcsO3xABQA=ab5efc4740a6c625ecf6a6729d0d67d2b62b615a", uri="/axis-cgi/audio/transmit.cgi", response="4bd3a627b20d6bcaba9e2f595ef6cd2a", algorithm="MD5", qop="auth", cnonce="6a579dd6664b57eb", nc=00000001




    Successfully opened the file.
    detected 8 logical cores
    [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'time_base' to value '1/44100'
    [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_rate' to value '44100'
    [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'sample_fmt' to value 's16'
    [graph 0 input from stream 0:0 @ 0000000001c9f6e0] Setting 'channel_layout' to value '0x3'
    [graph 0 input from stream 0:0 @ 0000000001c9f6e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
    [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_fmts' to value 's16'
    [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'sample_rates' to value '16000'
    [audio format for output stream 0:0 @ 0000000001c9fa20] Setting 'channel_layouts' to value '0x4'
    [audio format for output stream 0:0 @ 0000000001c9fa20] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
    [AVFilterGraph @ 000000000002ab20] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
    [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Using s16p internally between filters
    [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] Matrix coefficients:
    [auto-inserted resampler 0 @ 0000000001ca4060] [SWR @ 0000000001ca4a80] FC: FL:0.500000 FR:0.500000
    [auto-inserted resampler 0 @ 0000000001ca4060] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
    Output #0, flv, to 'http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi':
     Metadata:
       encoder         : Lavf57.57.100
       Stream #0:0, 0, 1/1000: Audio: pcm_mulaw ([8][0][0][0] / 0x0008), 16000 Hz, mono, s16, 128 kb/s
       Metadata:
         encoder         : Lavc57.66.101 pcm_mulaw
    Stream mapping:
     Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
    Press [q] to stop, [?] for help
    cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    av_interleaved_write_frame(): Unknown error
    No more output streams to write to, finishing.
    Error writing trailer of http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi: Error number -10053 occurredsize=       8kB time=00:00:00.49 bitrate= 131.2kbits/s speed=79.6x    
    video:0kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.492485%
    Input file #0 (audio=Microphone (2- High Definition Audio Device)):
     Input stream #0:0 (audio): 1 packets read (88200 bytes); 1 frames decoded (22050 samples);
     Total: 1 packets (88200 bytes) demuxed
    Output file #0 (http://operator:operator@10.10.210.2/axis-cgi/audio/transmit.cgi):
     Output stream #0:0 (audio): 1 frames encoded (7984 samples); 1 packets muxed (7984 bytes);
     Total: 1 packets (7984 bytes) muxed
    1 frames successfully decoded, 0 decoding errors
    [AVIOContext @ 0000000001c9e4c0] Statistics: 0 seeks, 2 writeouts
    dshow passing through packet of type audio size    12152 timestamp 310226130000 orig timestamp 310226130000 graph timestamp 310226820000 diff 690000 Microphone (2- High Definition Audio Device)
    Conversion failed!

    For some reason, despite setting multiple_requests, reconnect_eof, reconnect_streamed all to true, connection becomes closed.

    Could you please tell me what I’m doing wrong ?

  • Qt Video Recorder

    11 mai 2014, par Davlog

    I am trying to create a video recorder with Qt. What I did so far was taking a screenshot of a rectangle on the screen and save it. At the end I use ffmpeg to get a video file out of the images.

    I connected a timer’s signal timeout() to my custom slot which takes the snapshot and saves it to my tmp folder. The timer has an intervall of 1000 / 30. That should be 30 times per second. But 1000 / 30 is a little bit more than 33 milliseconds so I cannot really get 30 fps. It’s a bit more.

    I recorded a youtube video with my recorder and everything was smooth but a little bit faster / slower depending on the intervall.

    So my question basically is how do I get really 30 / 40 / 50 / ... fps ?