Recherche avancée

Médias (91)

Autres articles (83)

  • Le profil des utilisateurs

    12 avril 2011, par

    Chaque utilisateur dispose d’une page de profil lui permettant de modifier ses informations personnelle. Dans le menu de haut de page par défaut, un élément de menu est automatiquement créé à l’initialisation de MediaSPIP, visible uniquement si le visiteur est identifié sur le site.
    L’utilisateur a accès à la modification de profil depuis sa page auteur, un lien dans la navigation "Modifier votre profil" est (...)

  • Configurer la prise en compte des langues

    15 novembre 2010, par

    Accéder à la configuration et ajouter des langues prises en compte
    Afin de configurer la prise en compte de nouvelles langues, il est nécessaire de se rendre dans la partie "Administrer" du site.
    De là, dans le menu de navigation, vous pouvez accéder à une partie "Gestion des langues" permettant d’activer la prise en compte de nouvelles langues.
    Chaque nouvelle langue ajoutée reste désactivable tant qu’aucun objet n’est créé dans cette langue. Dans ce cas, elle devient grisée dans la configuration et (...)

  • 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

Sur d’autres sites (14073)

  • ffmpeg unable to convert vhs-captured .ts files [closed]

    4 avril, par fredko

    I'm digitizing vhs tapes with a Hauppauge Colossus capture card and Mediaportal on windows 7. They're captured as .ts files and the .ts files play well, with no sign of corruption.

    


    But I'd like to convert them to .mkv (losslessly from the .ts), and ffmpeg (version 2023-08-28-git-b5273c619d-essentials_build-www.gyan.dev) fails at this. I use

    


    ffmpeg.exe -i test.ts -c copy test.mkv


    


    and get these errors repeated many times

    


    [mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 46105).
[mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 49107).
[h264 @ 00000000003fc580] non-existing PPS 0 referenced
    Last message repeated 1 times
[h264 @ 00000000003fc580] decode_slice_header error
[h264 @ 00000000003fc580] no frame!
...etc...
[in#0/mpegts @ 00000000003d6b80] corrupt input packet in stream 0
[mpegts @ 00000000003d6d40] Packet corrupt (stream = 0, dts = 247305).
...etc...


    


    The resulting mkv file is much smaller than the .ts, and displays solid black in mpc-hc. Ffprobe gives the same errors and ends with

    


    Input #0, mpegts, from 'test.ts':
  Duration: 00:24:26.80, start: 0.099956, bitrate: 4486 kb/s
  Program 137 
  Stream #0:0[0x30]: Video: h264 (Main) (HDMV / 0x564D4448), yuv420p(top first),
 720x480 [SAR 10:11 DAR 15:11], 29.97 fps, 29.97 tbr, 90k tbn
  Stream #0:1[0x40]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo,
 fltp, 188 kb/s


    


    Ffmpeg seems to be complaining about corruption, though again the .ts files play fine. Is there some way to use ffmpeg to convert these files to mkv ? Or is the problem with the capture setup ?

    


  • python subprocess ffmpeg return code = 69

    13 juin 2023, par Tim Chen

    I try to call ffmpeg through the subprocess.run(['ffmpeg', '-i', file_name, output_file_name], capture_output=True, text=True) command in python to convert the audio file incoming from the front end to wav format file. The backend code is as follows, using python+fastapi :

    


    @app.post("/api/upload/convert")
async def convert_upload_file(request: Request, file: UploadFile = File(...)):
    token = uuid.uuid4().hex
    tmpFileName = os.path.join(os.path.dirname(__file__), token)
    with open(tmpFileName, "wb") as buffer:
        buffer.write(await file.read())
    await file.seek(0)
    output_path = tmpFileName + '-output.wav'
    command = ['ffmpeg', '-i', tmpFileName, output_path]
    result = subprocess.run(command, capture_output=True, text=True)


    


    This code usually works, but there are some scenarios where it doesn't work. The audio file is recorded by js code (specifically navigator.mediaDevices.getUserMedia({audio: true})).
The code of the audio recorded in windows chrome can run normally and get the converted wav file, but the audio recorded from ios15 safari for more than 3 seconds cannot be converted, prompting returncode=69. The error message is as follows :

    


    CompletedProcess(args=['ffmpeg', '-i', '5cfb52c503a646bda0f422b517c8014a', '5cfb52c503a646bda0f422b517c8014a-output.wav'], returncode=69, stdout='', stderr="
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil      56. 70.100 / 56. 70.100
libavcodec     58.134.100 / 58.134.100
libavformat    58. 76.100 / 58. 76.100
libavdevice    58. 13.100 / 58. 13.100
libavfilter     7.110.100 /  7.110.100
libswscale      5.  9.100 /  5.  9.100
libswresample   3.  9.100 /  3.  9.100
libpostproc    55.  9.100 / 55.  9.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '5cfb52c503a646bda0f422b517c8014a':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2023-06-11T16:36:53.000000Z
  Duration: 00:00:07.06, start: 0.000000, bitrate: 187 kb/s
  Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 184 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '5cfb52c503a646bda0f422b517c8014a-output.wav':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    ISFT            : Lavf58.76.100
  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
      encoder         : Lavc58.134.100 pcm_s16le
size=       2kB time=00:00:00.00 bitrate=N/A speed=N/A    
[aac @ 0x55f1f8f19fc0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x55f1f8f19fc0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x55f1f8f19fc0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[aac @ 0x55f1f8f19fc0] Multiple frames in a packet.
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Number of bands (18) exceeds limit (13).
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.


    


    For the abnormal code, I tried to execute ffmpeg -i input output.wav after fastapi handle request on the command line and subprocess.run(['ffmpeg', '-i', file_name, output_path], capture_output =True, text=True), all succeeded, which means that the final file must be normal, otherwise the subsequent verification work will get the same error.

    


    This confuses me, is there some information I'm missing ?

    


  • File conversion to mp3 returning failure everytime using flutter package ffmpeg_kit_flutter

    6 novembre 2024, par Sanath balthar

    I am trying to convert a .wav audio file generated from a flutter's text to speech package - "flutter_tts" to mp3 file but it is failing everytime.
I have written the below code for file conversion. I have imported the package ffmpeg_kit_flutter. It doesnt even show why the conversion is failing.
I have looked up in stackoverflow and other sites but could not find any relevant solutions. I am using vscode as editor. I have attached flutter doctor output below as well. Could anyone please guide me ? Let me know if you need more information.

    


    List<string> command = [&#xA;              &#x27;-i&#x27;, &#x27;$filePath/998tts.wav&#x27;,&#xA;              &#x27;-c:a&#x27;, &#x27;mp3&#x27;,&#xA;              &#x27;$filePath/998.mp3&#x27;&#xA;            ];&#xA;&#xA; await FFmpegKitConfig.enableLogs();&#xA;            FFmpegKitConfig.enableLogCallback((log) =>print(&#x27;FFmpeg log: $log&#x27;));        &#xA;          FFmpegSession result = await FFmpegKit.executeWithArguments(command);&#xA;          dynamic resultcode = await result.getReturnCode();&#xA;          dynamic resultlogs = await result.getLogsAsString();&#xA;          // FFmpegKitConfig.setLogLevel(logLevel)&#xA;          if(ReturnCode.isSuccess(resultcode)){&#xA;          print("file saved after conversion at $filePath/998.mp3 and result : Success and logs : $resultlogs");&#xA;          }&#xA;          else{&#xA;            print("Result : failure and logs : $resultlogs");&#xA;          }&#xA;&#xA;Flutter doctor output:&#xA;[√] Flutter (Channel stable, 3.19.6, on Microsoft Windows [Version 10.0.22631.3296], locale en-IN)&#xA;[√] Windows Version (Installed version of Windows is version 10 or higher)&#xA;[√] Android toolchain - develop for Android devices (Android SDK version 34.0.0)&#xA;[√] Chrome - develop for the web&#xA;[!] Visual Studio - develop Windows apps (Visual Studio Community 2022 17.9.5)&#xA;X Visual Studio is missing necessary components. Please re-run the Visual Studio installer for the "Desktop development with C&#x2B;&#x2B;"&#xA;workload, and include these components:&#xA;MSVC v142 - VS 2019 C&#x2B;&#x2B; x64/x86 build tools&#xA;- If there are multiple build tool versions available, install the latest&#xA;C&#x2B;&#x2B; CMake tools for Windows&#xA;Windows 10 SDK&#xA;[√] Android Studio (version 2023.2)&#xA;[√] VS Code (version 1.89.0)&#xA;[√] Connected device (3 available)&#xA;[√] Network resources&#xA;&#xA;! Doctor found issues in 1 category.&#xA;&#xA;</string>

    &#xA;

    Edit : Attaching error logs :

    &#xA;

    I/flutter (25865): Loading ffmpeg-kit-flutter.&#xA;D/ffmpeg-kit-flutter(25865): FFmpegKitFlutterPlugin com.arthenica.ffmpegkit.flutter.FFmpegKitFlutterPlugin@a5d9788 started listening to events on io.flutter.plugin.common.EventChannel$IncomingStreamRequestHandler$EventSinkImplementation@4cfb5f2.&#xA;I/flutter (25865): Loaded ffmpeg-kit-flutter-android-audio-arm64-v8a-6.0.3.&#xA;I/flutter (25865): Result : failure and logs : ffmpeg version n6.0 Copyright (c) 2000-2023 the FFmpeg developers&#xA;I/flutter (25865):   built with Android (7155654, based on r399163b1) clang version 11.0.5 (https://android.googlesource.com/toolchain/llvm-project 87f1315dfbea7c137aa2e6d362dbb457e388158d)&#xA;&#xA;I/flutter (25865):   configuration: --cross-prefix=aarch64-linux-android- --sysroot=/Users/sue/Library/Android/sdk/ndk/22.1.7171670/toolchains/llvm/prebuilt/darwin-x86_64/sysroot --prefix=/Users/sue/Projects/arthenica/ffmpeg-kit/prebuilt/android-arm64/ffmpeg --pkg-config=/opt/homebrew/bin/pkg-config --enable-version3 --arch=aarch64 --cpu=armv8-a --target-os=android --enable-neon --enable-asm --enable-inline-asm --ar=aarch64-linux-android-ar --cc=aarch64-linux-android24-clang --cxx=aarch64-linux-android24-clang&#x2B;&#x2B; --ranlib=aarch64-linux-android-ranlib --strip=aarch64-linux-android-strip --nm=aarch64-linux-android-nm --extra-libs=&#x27;-L/Users/sue/Projects/arthenica/ffmpeg-kit/prebuilt/android-arm64/cpu-features/lib -lndk_compat&#x27; --disable-autodetect --enable-cross-compile &#xA;

    &#xA;