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Collections - Formulaire de création rapide
19 février 2013, par
Mis à jour : Février 2013
Langue : français
Type : Image
Autres articles (86)
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MediaSPIP version 0.1 Beta
16 avril 2011, parMediaSPIP 0.1 beta est la première version de MediaSPIP décrétée comme "utilisable".
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Pour avoir une installation fonctionnelle, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 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 (...) -
Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
Sur d’autres sites (12711)
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python subprocess ffmpeg return code = 69
13 juin 2023, par Tim ChenI 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, promptingreturncode=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 andsubprocess.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 ?


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File conversion to mp3 returning failure everytime using flutter package ffmpeg_kit_flutter
6 novembre 2024, par Sanath baltharI 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 = [
 '-i', '$filePath/998tts.wav',
 '-c:a', 'mp3',
 '$filePath/998.mp3'
 ];

 await FFmpegKitConfig.enableLogs();
 FFmpegKitConfig.enableLogCallback((log) =>print('FFmpeg log: $log')); 
 FFmpegSession result = await FFmpegKit.executeWithArguments(command);
 dynamic resultcode = await result.getReturnCode();
 dynamic resultlogs = await result.getLogsAsString();
 // FFmpegKitConfig.setLogLevel(logLevel)
 if(ReturnCode.isSuccess(resultcode)){
 print("file saved after conversion at $filePath/998.mp3 and result : Success and logs : $resultlogs");
 }
 else{
 print("Result : failure and logs : $resultlogs");
 }

Flutter doctor output:
[√] Flutter (Channel stable, 3.19.6, on Microsoft Windows [Version 10.0.22631.3296], locale en-IN)
[√] Windows Version (Installed version of Windows is version 10 or higher)
[√] Android toolchain - develop for Android devices (Android SDK version 34.0.0)
[√] Chrome - develop for the web
[!] Visual Studio - develop Windows apps (Visual Studio Community 2022 17.9.5)
X Visual Studio is missing necessary components. Please re-run the Visual Studio installer for the "Desktop development with C++"
workload, and include these components:
MSVC v142 - VS 2019 C++ x64/x86 build tools
- If there are multiple build tool versions available, install the latest
C++ CMake tools for Windows
Windows 10 SDK
[√] Android Studio (version 2023.2)
[√] VS Code (version 1.89.0)
[√] Connected device (3 available)
[√] Network resources

! Doctor found issues in 1 category.

</string>


Edit : Attaching error logs :


I/flutter (25865): Loading ffmpeg-kit-flutter.
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.
I/flutter (25865): Loaded ffmpeg-kit-flutter-android-audio-arm64-v8a-6.0.3.
I/flutter (25865): Result : failure and logs : ffmpeg version n6.0 Copyright (c) 2000-2023 the FFmpeg developers
I/flutter (25865): built with Android (7155654, based on r399163b1) clang version 11.0.5 (https://android.googlesource.com/toolchain/llvm-project 87f1315dfbea7c137aa2e6d362dbb457e388158d)

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++ --ranlib=aarch64-linux-android-ranlib --strip=aarch64-linux-android-strip --nm=aarch64-linux-android-nm --extra-libs='-L/Users/sue/Projects/arthenica/ffmpeg-kit/prebuilt/android-arm64/cpu-features/lib -lndk_compat' --disable-autodetect --enable-cross-compile 



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avformat/ffrtmpcrypt : Fix int-conversion warning
22 décembre 2023, par Frank Plowmanavformat/ffrtmpcrypt : Fix int-conversion warning
The gcrypt definition of `bn_new` used to use the return statement
on errors, with an AVERROR return value, regardless of the signature
of the function where the macro is used - it is called in
`dh_generate_key` and `ff_dh_init` which return pointers. As a result,
compiling with gcrypt and the ffrtmpcrypt protocol resulted in an
int-conversion warning. GCC 14 may upgrade these to errors [1].This patch fixes the problem by changing the macro to remove `AVERROR`
and instead set `bn` to null if the allocation fails. This is the
behaviour of all the other `bn_new` implementations and so the result is
already checked at all the callsites. AFAICT, this should be the only
change needed to get ffmpeg off Fedora's naughty list of projects with
warnings which may be upgraded to errors in GCC 14 [2].[1] : https://gcc.gnu.org/pipermail/gcc/2023-May/241264.html
[2] : https://www.mail-archive.com/devel@lists.fedoraproject.org/msg196024.htmlSigned-off-by : Frank Plowman <post@frankplowman.com>
Signed-off-by : Martin Storsjö <martin@martin.st>