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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 (11116)
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PHP-FFMPEG Waveform is a little noisy, how to fix that ?
18 avril 2023, par KranchiI made a waveform through sample code :


$waveform = $audio->waveform(2000, 500, array('#FFA500'));
$waveform->save('waveform.png');



But result has some noise in top & bottom of edges :
https://gcdnb.pbrd.co/images/80v6Up6etCrD.png?o=1


So I made a tiny change in line 140 of src\FFMpeg\Media\Waveform.php :


//'showwavespic=colors='.$this->compileColors().':s='.$this->width.'x'.$this->height,
'showwavespic=draw=full:colors='.$this->compileColors().':s='.$this->width.'x'.$this->height,



And problem fixed :
https://gcdnb.pbrd.co/images/KsCFsOujmJ9W.png?o=1


I'm not sure is this a bug or problem in FFMpeg dll file and my question :


is it possible to send the new command or override save method to avoid hacking the original file ? (the save method is using some protected methods)


https://github.com/PHP-FFMpeg/PHP-FFMpeg/blob/master/src/FFMpeg/Media/Waveform.php


Thanks.



PHP-FFMpeg : 23 | FFMpeg : 2023-02-27-git-891ed24f77 (Win X64) |
PHP : 8.2


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ffmpeg unable to convert vhs-captured .ts files [closed]
4 avril, par fredkoI'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 ?


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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 ?