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  • La sauvegarde automatique de canaux SPIP

    1er avril 2010, par

    Dans le cadre de la mise en place d’une plateforme ouverte, il est important pour les hébergeurs de pouvoir disposer de sauvegardes assez régulières pour parer à tout problème éventuel.
    Pour réaliser cette tâche on se base sur deux plugins SPIP : Saveauto qui permet une sauvegarde régulière de la base de donnée sous la forme d’un dump mysql (utilisable dans phpmyadmin) mes_fichiers_2 qui permet de réaliser une archive au format zip des données importantes du site (les documents, les éléments (...)

  • 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 (...)

  • De l’upload à la vidéo finale [version standalone]

    31 janvier 2010, par

    Le chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
    Upload et récupération d’informations de la vidéo source
    Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
    Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)

Sur d’autres sites (3565)

  • Remove characters from string using cmd.exe batch file and ffmpeg

    26 décembre 2023, par user2498772

    Here is the code...

    


    @echo off
:: set paths (no quotes)
set ffmpeg=F:\Video\Ffmpeg\ffmpeg.exe
set infolder=F:\Downloads
set outfolder=F:\Video\Finished



@echo.
@echo on
for %%f in (%infolder%\*.mp4) do "%ffmpeg%" -i "%%~dpnf.mp4" -c copy "%outfolder%\%%~nf.mp4"
@echo off
if errorlevel 1 pause
@echo.


    


    This code converts an mpeg-dash file to mp4.
An mpeg-dash is a streaming file format.

    


    The input and output file looks like this : "filename - DASH_V.mp4". (no quotes)

    


    I would like the output file to have the " - DASH_V" removed, so it looks like "filename.mp4". (no quotes)

    


    Any help would be appreciated. Could be a quick fix like changing "%% dpnf -13.0.mp4" to last command, but I am clueless on syntax as you can see.

    


    Tried a million things and got nowhere. Clueless on syntax.

    


  • FFPlay : How to manually select video quality during playing the mpd stream ?

    19 août 2023, par Brian_wu

    I used ffplay to play mpd stream and successed,

    


    Here the MPD file :

    


    &lt;?xml version="1.0" encoding="utf-8"?>&#xA;<mpd xmlns="urn:mpeg:dash:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-live:2011" type="static" mediapresentationduration="PT1M8.7S" maxsegmentduration="PT5.0S" minbuffertime="PT12.5S">&#xA;    <programinformation>&#xA;    </programinformation>&#xA;    <servicedescription>&#xA;    </servicedescription>&#xA;    <period start="PT0.0S">&#xA;        <adaptationset contenttype="video" startwithsap="1" segmentalignment="true" bitstreamswitching="true" framerate="24000/1001" maxwidth="1280" maxheight="720" par="16:9">&#xA;        <representation mimetype="video/mp4" codecs="avc1.4d401f" bandwidth="10237" width="480" height="270" sar="1:1">&#xA;            <segmenttemplate timescale="24000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">&#xA;                <segmenttimeline>&#xA;                    <s t="0" d="150150" r="9"></s>&#xA;                    <s d="149149"></s>&#xA;                </segmenttimeline>&#xA;            </segmenttemplate>&#xA;        </representation>&#xA;        <representation mimetype="video/mp4" codecs="avc1.4d401f" bandwidth="60882" width="1280" height="720" sar="1:1">&#xA;            <segmenttemplate timescale="24000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">&#xA;                <segmenttimeline>&#xA;                    <s t="0" d="150150" r="9"></s>&#xA;                    <s d="149149"></s>&#xA;                </segmenttimeline>&#xA;            </segmenttemplate>&#xA;        </representation>&#xA;    </adaptationset>&#xA;    <adaptationset contenttype="audio" startwithsap="1" segmentalignment="true" bitstreamswitching="true" lang="und">&#xA;        <representation mimetype="audio/mp4" codecs="mp4a.40.2" bandwidth="128000" audiosamplingrate="32000">&#xA;            <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>&#xA;            <segmenttemplate timescale="32000" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">&#xA;                <segmenttimeline>&#xA;                    <s t="0" d="159744"></s>&#xA;                    <s d="160768" r="11"></s>&#xA;                    <s d="111915"></s>&#xA;                </segmenttimeline>&#xA;            </segmenttemplate>&#xA;        </representation>&#xA;        <representation mimetype="audio/mp4" codecs="mp4a.40.2" bandwidth="128000" audiosamplingrate="44100">&#xA;            <audiochannelconfiguration schemeiduri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"></audiochannelconfiguration>&#xA;            <segmenttemplate timescale="44100" initialization="init-stream$RepresentationID$.m4s" media="chunk-stream$RepresentationID$-$Number%05d$.m4s" startnumber="1">&#xA;                <segmenttimeline>&#xA;                    <s t="0" d="220160"></s>&#xA;                    <s d="221184" r="11"></s>&#xA;                    <s d="158713"></s>&#xA;                </segmenttimeline>&#xA;            </segmenttemplate>&#xA;        </representation>&#xA;    </adaptationset>&#xA;</period>&#xA;</mpd>

    &#xA;&#xA;

    this dash stream contains 2 vindeos(480P and 720P), and 2 audios.The videos always starts low quality(480P), I want to change the video resolution to the high quality(720P) during playing process, what should i do ?

    &#xA;

  • youtube stream from ffmpeg is buffering

    1er juin 2020, par Bartonsen

    I'm using ffmpeg running on a Raspberry Pi 3B with 1GB RAM to stream live video on youtube.&#xA;In the beginning the audio+video stream is excellent, but after some minutes I see error messages in YT studio, and video starts buffering.&#xA;After some more time (could be 30 mins or 1 hr) the youtube stream is gone, although ffmpeg is still running.

    &#xA;&#xA;

    ffmpeg is configured like this :

    &#xA;&#xA;

    ./configure --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi&#xA;

    &#xA;&#xA;

    Running ffmpeg :

    &#xA;&#xA;

    pi@raspberrypi:~ $ ffmpeg -thread_queue_size 512 -rtsp_transport udp -i "rtsp://10.x.x.x:554/user=user&amp;password=password&amp;channel=1&amp;stream=0.sdp?real_stream" -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/mykey&#xA;ffmpeg version git-2020-05-01-3c740f2 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 6.3.0 (Raspbian 6.3.0-18&#x2B;rpi1&#x2B;deb9u1) 20170516&#xA;  configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi&#xA;  libavutil      56. 43.100 / 56. 43.100&#xA;  libavcodec     58. 82.100 / 58. 82.100&#xA;  libavformat    58. 42.102 / 58. 42.102&#xA;  libavdevice    58.  9.103 / 58.  9.103&#xA;  libavfilter     7. 80.100 /  7. 80.100&#xA;  libswscale      5.  6.101 /  5.  6.101&#xA;  libswresample   3.  6.100 /  3.  6.100&#xA;  libpostproc    55.  6.100 / 55.  6.100&#xA;[udp @ 0x3a8c370] attempted to set receive buffer to size 393216 but it only ended up set as 327680&#xA;[udp @ 0x3a9ea80] attempted to set receive buffer to size 393216 but it only ended up set as 327680&#xA;[udp @ 0x3a8c3e0] attempted to set receive buffer to size 393216 but it only ended up set as 327680&#xA;[udp @ 0x3abf4d0] attempted to set receive buffer to size 393216 but it only ended up set as 327680&#xA;Guessed Channel Layout for Input Stream #0.1 : mono&#xA;Input #0, rtsp, from &#x27;rtsp://10.x.x.x:554/user=user&amp;password=password&amp;channel=1&amp;stream=0.sdp?real_stream&#x27;:&#xA;  Metadata:&#xA;    title           : RTSP Session&#xA;  Duration: N/A, start: 0.000000, bitrate: N/A&#xA;    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 20 fps, 20 tbr, 90k tbn, 180k tbc&#xA;    Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (copy)&#xA;  Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))&#xA;Press [q] to stop, [?] for help&#xA;[aac @ 0x3ae9f00] Too many bits 8832.000000 > 6144 per frame requested, clamping to max&#xA;Output #0, flv, to &#x27;rtmp://a.rtmp.youtube.com/live2/mykey&#x27;:&#xA;  Metadata:&#xA;    title           : RTSP Session&#xA;    encoder         : Lavf58.42.102&#xA;    Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 20 fps, 20 tbr, 1k tbn, 90k tbc&#xA;    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s&#xA;    Metadata:&#xA;      encoder         : Lavc58.82.100 aac&#xA;

    &#xA;&#xA;

    In youtube studio I see this :

    &#xA;&#xA;

    19:36 Good transmission. The quality is excellent.&#xA;19:39 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.&#xA;19:39 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.&#xA;19:39 Warning: The current bit rate (1974.24 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.&#xA;19:41 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.&#xA;19:41 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.&#xA;19:41 Warning: The current bit rate (2151.41 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.&#xA;19:43 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.&#xA;19:43 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.&#xA;19:45 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.&#xA;19:45 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.&#xA;19:45 Warning: The current bit rate (1737.61 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.&#xA;...&#xA;19:54: Error: YouTube does not receive enough video to maintain consistent streaming. Viewers will therefore experience buffering.&#xA;

    &#xA;&#xA;

    What is the problem ? How can I get rid of the buffering ?&#xA;I've also tried the below two commands, but found the output to be worse...

    &#xA;&#xA;

    ffmpeg -i rtsp://... -c:v libx264  -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output&#xA;ffmpeg -i rtsp://... -c:v h264_omx -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output&#xA;

    &#xA;