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  • MediaSPIP 0.1 Beta version

    25 avril 2011, par

    MediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
    The zip file provided here only contains the sources of MediaSPIP in its standalone version.
    To get a working installation, you must manually install all-software dependencies on the server.
    If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...)

  • 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 (...)

  • ANNEXE : Les plugins utilisés spécifiquement pour la ferme

    5 mars 2010, par

    Le site central/maître de la ferme a besoin d’utiliser plusieurs plugins supplémentaires vis à vis des canaux pour son bon fonctionnement. le plugin Gestion de la mutualisation ; le plugin inscription3 pour gérer les inscriptions et les demandes de création d’instance de mutualisation dès l’inscription des utilisateurs ; le plugin verifier qui fournit une API de vérification des champs (utilisé par inscription3) ; le plugin champs extras v2 nécessité par inscription3 (...)

Sur d’autres sites (8878)

  • FFMPEG MP3 file size much larger than expected on Windows 10

    8 avril 2018, par The Gora

    I’ve been using FFMPEG on Windows to :

    1. Convert iTunes M4A files to MP3s (with a bit rate of 128k) ; and
    2. Create 30 sec sample MP3s of the above MP3s (same bit rate).

    When I run FFMEG on a Windows 7 64 bit machine, the size of the MP3s (both for 1. & 2.) is in line with the rough calculation of :

    (Audio length in seconds) X (Bit rate)

    For example, a 4 minute audio yields an approx. 3.7MB MP3 file ; a 30 second sample MP3 is approx. 470KB.

    However when I run the same FFMPEG binary (copied from the Windows 7 machine) on a Windows 10 64 bit machine, all of the MP3s (both for 1. and for 2.) are inflated by approx 5MB. I’m using the same batch file on both machines to execute FFMEG with the required parameters, so pretty confident the difference is not down to user error.

    My questions are :

    1. Why is there this apparent 5MB overhead on Windows 10 ? and more importantly ;
    2. What can I do to remove this ?

    The large file size is a problem as the sample MP3s are to be put on a website for people to listen to a snippet of the song, and the webpage with multiple tags takes a long time to load completely (several minutes).

    Here is the version and lib info :

    ffmpeg version 3.4.1 Copyright (c) 2000-2017 the FFmpeg developers
     built with gcc 7.2.0 (GCC)
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth --enable-libmfx
     libavutil      55. 78.100 / 55. 78.100
     libavcodec     57.107.100 / 57.107.100
     libavformat    57. 83.100 / 57. 83.100
     libavdevice    57. 10.100 / 57. 10.100
     libavfilter     6.107.100 /  6.107.100
     libswscale      4.  8.100 /  4.  8.100
     libswresample   2.  9.100 /  2.  9.100
     libpostproc    54.  7.100 / 54.  7.100

    And here are the command lines I’m using :

    1. ffmpeg -i input.m4a -id3v2_version 3 -b:a 128k -output.mp3
    2. ffmpeg -i input.m4a -ss 30 -t 30 -af "afade=in:st=30:d=5,afade=out:st=55:d=5" -id3v2_version 3 -b:a 128k -output.mp3
  • Delete file after discord.FFmpegPCMAudio is done playing

    12 novembre 2023, par Moni

    I want to delete a mp3 file in the same directory after discord.FFmpegPCMAudio is done playing, but i don't know how ! I've searched everywhere before posting here, but with no luck.

    


    @bot.command(name='play')
async def play(ctx, url):
 global current_audio_file
 voice_client = ctx.guild.voice_client

 if voice_client.is_connected():
    if ctx.author.voice and ctx.author.voice.channel == voice_client.channel:
        try:
            server = ctx.guild
            voice_channel = server.voice_client

            async with ctx.typing():
                if "youtube" or "youtu" in url:
                    result, title = await download_youtube_audio(url)
                    result_b = bytes(result, "utf-8")
                    audio_file_name = codecs.decode(result_b, 'charmap')

                    audio_file, title2 = find_file(audio_file_name, 2)
                elif "spotify" in url:
                    result = await download_spotify_audio(url)
                    result_b = bytes(result, "utf-8")
                    audio_file_name = codecs.decode(result_b, 'charmap')

                    audio_file, title = find_file(audio_file_name, 1)

                if result:
                    if "ERROR" not in result:
                        await ctx.send(f'**Now playing:** {title}')

                        if not voice_channel.is_playing():
                            voice_channel.on_audio_end = lambda e: asyncio.create_task(delete_file(audio_file))

                            voice_channel.play(discord.FFmpegPCMAudio(executable="ffmpeg.exe", source=audio_file))
                            current_audio_file = audio_file
                        else:
                            await ctx.send("Another song is playing right now!")
                    else:
                        print(f"An error occurred while trying to download the audio. Error: {result}")
                        send_error(ctx, Exception(result))
                else:
                    await ctx.send("An error occurred while trying to download the audio.")
        except Exception as e:
            print(f"An error occurred: {e}")
            send_error(ctx, e)
            await ctx.send("An error occurred while trying to play the audio.")
    else:
        await ctx.send("You are not in the same voice channel as the bot.")
else:
    await ctx.send("The bot is not connected to a voice channel.")

def delete_file(filename: str):
if os.path.exists(filename):
    os.remove(filename)
    print(f"Deleted {filename}")


    


    If anyone knows how to do it, please help me out.

    


  • avcodec/libx264 : bump minimum required version to 155

    9 avril 2024, par Niklas Haas
    avcodec/libx264 : bump minimum required version to 155
    

    This version is seven years old, and present in Debian oldoldstable,
    Ubuntu 20.04 and Leap 15.0.

    Allows cleaning up the file substantially. In particular, this is
    motivated by the desire to stop relying on init_static_data.

    • [DH] configure
    • [DH] libavcodec/libx264.c