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  • Publier sur MédiaSpip

    13 juin 2013

    Puis-je poster des contenus à partir d’une tablette Ipad ?
    Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir

  • 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 (...)

  • Support audio et vidéo HTML5

    10 avril 2011

    MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
    Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
    Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
    Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)

Sur d’autres sites (8962)

  • How to modify pitch in audio of video file with singing voices

    9 janvier 2020, par Keith Bennett

    I have some karaoke .mp4 video files (legally obtained) for Thai songs, and want to convert the pitch downward to fit my singing range. I’ve gotten most of the way there thanks to https://superuser.com/questions/292833/how-to-change-audio-frequency/1076762#1076762
    using a command line like this :

    ffmpeg -i in.mp4 -af 'asetrate=35280.0,atempo=1.25' out.mp4

    ...but the human singing voices don’t sound natural at the modified pitch.

    Is there a better way to change the pitch ? I know some commercial products can do this.

    By the way, I wrote a Ruby script to simplify this ffmpeg call ; it’s at https://gist.github.com/keithrbennett/9ba7043792bfb2fcc92d615076a8413f. It enables you to specify a single factor, and modifies both pitch and tempo accordingly.

  • Is FFmpegAudioDecoder supposed to reinitialize upon append of new init segment

    14 novembre 2023, par martin

    I am attempting to switch audio tracks but when switching the FFmpegAudioDecoder never reinitializes like it does with video tracks of differing resolutions. I am not certain if this is the intended behavior of FFmpegAudioDecoder and would love to learn more about the expected behavior.

    


    When switching audio tracks I end up calling the following operations :

    


    if sourceBuffer.getIsUpdate() {sourceBuffer.abort()}
sourceBuffer.remove(0-videoDuration)
initSegmentDataStream = fetch init segment of new audio representation
sourceBuffer.appendBuffer(initSegmentDataStream)


    


    These are the Media tab messages from initial video load

    


    ChunkDemuxer
Selected FFmpegAudioDecoder for audio decoding, config: codec: aac, profile: unknown, bytes_per_channel: 2, channel_layout: STEREO, channels: 2, samples_per_second: 48000, sample_format: Signed 16-bit, bytes_per_frame: 4, seek_preroll: 0us, codec_delay: 0, has extra data: false, encryption scheme: Unencrypted, discard decoder delay: false, target_output_channel_layout: STEREO, target_output_sample_format: Unknown sample format, has aac extra data: true
Cannot select DecryptingVideoDecoder for video decoding
Cannot select VDAVideoDecoder for video decoding
Cannot select VpxVideoDecoder for video decoding
Selected Dav1dVideoDecoder for video decoding, config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1280,720], visible rect: [0,0,1280,720], natural size: [1280,720], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}
Dropping audio frame (DTS 0us PTS -105375us,-62709us) that is outside append window [0us,9223372036854775807us).
Dropping audio frame (DTS 42666us PTS -62708us,-20042us) that is outside append window [0us,9223372036854775807us).
Truncating audio buffer which overlaps append window start. PTS -20041us frame_end_timestamp 22625us append_window_start 0us
Effective playback rate changed from 0 to 1


    


    For comparison this is what I get when appending the init segment of a different video resolution / track

    


    video decoder config changed midstream, new config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1920,1080], visible rect: [0,0,1920,1080], natural size: [1920,1080], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}



    


    Chrome version : Version 119.0.6045.123 (Official Build)

    


    When appending the new init segment of an audio track I was expecting the FFmpegAudioDecoder to be reinitialized like the Dav1dVideoDecoder does for video tracks

    


  • FFmpeg : Encode x264 with AMD GPU on Windows ?

    20 septembre 2023, par ZeroTek

    I am currently trying to record a Video on my Lenovo Laptop with its Built-In Webcam using FFmpeg on Windows 10. One of my goals is to keep the CPU Usage as low as possible, that's why i want to push the h264 encoding to the GPU. 
Now it gets a bit tricky here with my Laptop. Because it uses two GPUs. The first GPU is a Intel HD 5500 Graphics Unit as Part of the CPU. This one is most likly used for non-demanding Applications like office etc. to save Energy. The other one is a AMD R5 M330 that will be used for graphic intense applications like gaming.

    



    Currently, i am using the following command to encode the Webcam Stream on the Intel HD GPU :

    



    ffmpeg -f dshow -vcodec mjpeg -video_size 1280x720 -framerate 30 video="Lenovo EasyCamera":audio="Mikrofon (Realtek High Definition Audio)" -c:v h264_qsv -g 60 -q 28 -look_ahead 0 -preset:v faster -c:a aac -q:a 0.6 -r 30 output.mp4


    



    This does work so far but it seems this GPU does not have enough Power to keep up with the framerate on higher bitrates or with a high amount of i-frames. The Video starts lacking and skipping frames. If i am using CPU encoding everything works smooth.

    



    Now that my Laptop got that second AMD GPU with a lot more Power it would be a nice Try to encode on that one, but i can't find any information about how to encode on AMD Hardware on Windows 10. So my question is : How does the ffmpeg command look like to use AMD Hardware for h264 encoding ?