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La conservation du net art au musée. Les stratégies à l’œuvre
26 mai 2011
Mis à jour : Juillet 2013
Langue : français
Type : Texte
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Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...) -
MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 is the first MediaSPIP stable release.
Its official release date is June 21, 2013 and is announced here.
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 (...) -
Qu’est ce qu’un masque de formulaire
13 juin 2013, parUn masque de formulaire consiste en la personnalisation du formulaire de mise en ligne des médias, rubriques, actualités, éditoriaux et liens vers des sites.
Chaque formulaire de publication d’objet peut donc être personnalisé.
Pour accéder à la personnalisation des champs de formulaires, il est nécessaire d’aller dans l’administration de votre MediaSPIP puis de sélectionner "Configuration des masques de formulaires".
Sélectionnez ensuite le formulaire à modifier en cliquant sur sont type d’objet. (...)
Sur d’autres sites (4484)
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Stream ffmpeg transcoding result to S3
7 juin 2019, par mabeadI want to transcode a large file using FFMPEG and store the result directly on AWS S3. This will be done inside of an AWS Lambda that has limited tmp space so I can’t store the transcoding result locally and then upload it to S3 in a second step. I won’t have enough tmp space. I therefore want to store the FFMPEG output directly on S3.
I therefore created a S3 pre-signed url that allows ’PUT’ :
var outputPath = s3Client.GetPreSignedURL(new Amazon.S3.Model.GetPreSignedUrlRequest
{
BucketName = "my-bucket",
Expires = DateTime.UtcNow.AddMinutes(5),
Key = "output.mp3",
Verb = HttpVerb.PUT,
});I then called ffmpeg with the resulting pre-signed url :
ffmpeg -i C:\input.wav -y -vn -ar 44100 -ac 2 -ab 192k -f mp3 https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D
FFMPEG returns an exit code of 1 with the following output :
ffmpeg version N-93120-ga84af760b8 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 8.2.1 (GCC) 20190212
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --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-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 47.100 / 58. 47.100
libavformat 58. 26.101 / 58. 26.101
libavdevice 58. 6.101 / 58. 6.101
libavfilter 7. 48.100 / 7. 48.100
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'C:\input.wav':
Duration: 00:04:16.72, bitrate: 3072 kb/s
Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D':
Metadata:
TSSE : Lavf58.26.101
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s32p, 192 kb/s
Metadata:
encoder : Lavc58.47.100 libmp3lame
size= 577kB time=00:00:24.58 bitrate= 192.2kbits/s speed=49.1x
size= 1109kB time=00:00:47.28 bitrate= 192.1kbits/s speed=47.2x
[tls @ 000001d73d786b00] Error in the push function.
av_interleaved_write_frame(): I/O error
Error writing trailer of https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D: I/O error
size= 1143kB time=00:00:48.77 bitrate= 192.0kbits/s speed= 47x
video:0kB audio:1144kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[tls @ 000001d73d786b00] The specified session has been invalidated for some reason.
[tls @ 000001d73d786b00] Error in the pull function.
[https @ 000001d73d784fc0] URL read error: -5
Conversion failed!As you can see, I have a
URL read error
. This is a little surprising to me since I want to output to this url and not read it.Anybody know how I can store directly my FFMPEG output directly to S3 without having to store it locally first ?
Edit 1
I then tried to use the-method PUT
parameter and use http instead of https to remove TLS from the equation. Here’s the output that I got when running ffmpeg with the-v trace
option.ffmpeg version N-93120-ga84af760b8 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 8.2.1 (GCC) 20190212
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --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-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 47.100 / 58. 47.100
libavformat 58. 26.101 / 58. 26.101
libavdevice 58. 6.101 / 58. 6.101
libavfilter 7. 48.100 / 7. 48.100
libswscale 5. 4.100 / 5. 4.100
libswresample 3. 4.100 / 3. 4.100
libpostproc 55. 4.100 / 55. 4.100
Splitting the commandline.
Reading option '-i' ... matched as input url with argument 'C:\input.wav'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
Reading option '-ab' ... matched as option 'ab' (audio bitrate (please use -b:a)) with argument '192k'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
Reading option '-method' ... matched as AVOption 'method' with argument 'PUT'.
Reading option '-v' ... matched as option 'v' (set logging level) with argument 'trace'.
Reading option 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option y (overwrite output files) with argument 1.
Applying option v (set logging level) with argument trace.
Successfully parsed a group of options.
Parsing a group of options: input url C:\input.wav.
Successfully parsed a group of options.
Opening an input file: C:\input.wav.
[NULL @ 000001fb37abb180] Opening 'C:\input.wav' for reading
[file @ 000001fb37abc180] Setting default whitelist 'file,crypto'
Probing wav score:99 size:2048
[wav @ 000001fb37abb180] Format wav probed with size=2048 and score=99
[wav @ 000001fb37abb180] Before avformat_find_stream_info() pos: 54 bytes read:65590 seeks:1 nb_streams:1
[wav @ 000001fb37abb180] parser not found for codec pcm_s32le, packets or times may be invalid.
Last message repeated 1 times
[wav @ 000001fb37abb180] All info found
[wav @ 000001fb37abb180] stream 0: start_time: -192153584101141.156 duration: 256.716
[wav @ 000001fb37abb180] format: start_time: -9223372036854.775 duration: 256.716 bitrate=3072 kb/s
[wav @ 000001fb37abb180] After avformat_find_stream_info() pos: 204854 bytes read:294966 seeks:1 frames:50
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'C:\input.wav':
Duration: 00:04:16.72, bitrate: 3072 kb/s
Stream #0:0, 50, 1/48000: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s
Successfully opened the file.
Parsing a group of options: output url https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D.
Applying option vn (disable video) with argument 1.
Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
Applying option ac (set number of audio channels) with argument 2.
Applying option ab (audio bitrate (please use -b:a)) with argument 192k.
Applying option f (force format) with argument mp3.
Successfully parsed a group of options.
Opening an output file: https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D.
[http @ 000001fb37b15140] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
[tcp @ 000001fb37b16c80] Original list of addresses:
[tcp @ 000001fb37b16c80] Address 52.216.8.203 port 80
[tcp @ 000001fb37b16c80] Interleaved list of addresses:
[tcp @ 000001fb37b16c80] Address 52.216.8.203 port 80
[tcp @ 000001fb37b16c80] Starting connection attempt to 52.216.8.203 port 80
[tcp @ 000001fb37b16c80] Successfully connected to 52.216.8.203 port 80
[http @ 000001fb37b15140] request: PUT /output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D HTTP/1.1
Transfer-Encoding: chunked
User-Agent: Lavf/58.26.101
Accept: */*
Connection: close
Host: landr-distribution-reportsdev-mb.s3.amazonaws.com
Icy-MetaData: 1
Successfully opened the file.
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
detected 8 logical cores
[graph_0_in_0_0 @ 000001fb37b21080] Setting 'time_base' to value '1/48000'
[graph_0_in_0_0 @ 000001fb37b21080] Setting 'sample_rate' to value '48000'
[graph_0_in_0_0 @ 000001fb37b21080] Setting 'sample_fmt' to value 's32'
[graph_0_in_0_0 @ 000001fb37b21080] Setting 'channel_layout' to value '0x3'
[graph_0_in_0_0 @ 000001fb37b21080] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x3
[format_out_0_0 @ 000001fb37b22cc0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
[format_out_0_0 @ 000001fb37b22cc0] Setting 'sample_rates' to value '44100'
[format_out_0_0 @ 000001fb37b22cc0] Setting 'channel_layouts' to value '0x3'
[format_out_0_0 @ 000001fb37b22cc0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 000001fb37b0d940] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 000001fb37b251c0] picking s32p out of 3 ref:s32
[auto_resampler_0 @ 000001fb37b251c0] [SWR @ 000001fb37b252c0] Using fltp internally between filters
[auto_resampler_0 @ 000001fb37b251c0] ch:2 chl:stereo fmt:s32 r:48000Hz -> ch:2 chl:stereo fmt:s32p r:44100Hz
Output #0, mp3, to 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D':
Metadata:
TSSE : Lavf58.26.101
Stream #0:0, 0, 1/44100: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s32p, delay 1105, 192 kb/s
Metadata:
encoder : Lavc58.47.100 libmp3lame
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
Last message repeated 6 times
size= 649kB time=00:00:27.66 bitrate= 192.2kbits/s speed=55.3x
size= 1207kB time=00:00:51.48 bitrate= 192.1kbits/s speed=51.5x
av_interleaved_write_frame(): Unknown error
No more output streams to write to, finishing.
[libmp3lame @ 000001fb37b147c0] Trying to remove 47 more samples than there are in the queue
Error writing trailer of https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D: Error number -10054 occurred
size= 1251kB time=00:00:53.39 bitrate= 192.0kbits/s speed=51.5x
video:0kB audio:1252kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (C:\input.wav):
Input stream #0:0 (audio): 5014 packets read (20537344 bytes); 5014 frames decoded (2567168 samples);
Total: 5014 packets (20537344 bytes) demuxed
Output file #0 (https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D):
Output stream #0:0 (audio): 2047 frames encoded (2358144 samples); 2045 packets muxed (1282089 bytes);
Total: 2045 packets (1282089 bytes) muxed
5014 frames successfully decoded, 0 decoding errors
[AVIOContext @ 000001fb37b1f440] Statistics: 0 seeks, 2046 writeouts
[http @ 000001fb37b15140] URL read error: -10054
[AVIOContext @ 000001fb37ac4400] Statistics: 20611126 bytes read, 1 seeks
Conversion failed!So it looks like it is able to connect to my S3 pre-signed url but I still have the
Error writing trailer
error coupled with aURL read error
. -
What is Behavioural Segmentation and Why is it Important ?
28 septembre 2023, par Erin — Analytics Tips -
Audio out of sync, direct capture device stream (Windows 10)
3 mai 2020, par user3459555Using ffplay, the video stays in sync using this command :



ffplay -f dshow -rtbufsize 702000k video="Cam Link"
ffplay version git-2020-05-01-39fb1e9 Copyright (c) 2003-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200328
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 43.100 / 56. 43.100
 libavcodec 58. 82.100 / 58. 82.100
 libavformat 58. 42.101 / 58. 42.101
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 80.100 / 7. 80.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, dshow, from 'video=Cam Link':vq= 0KB sq= 0B f=0/0
 Duration: N/A, start: 141954.961000, bitrate: N/A
 Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1920x1080, 59.94 fps, 59.94 tbr, 10000k tbn, 10000k tbc
142992.53 M-V: -0.001 fd= 3 aq= 0KB vq= 0KB sq= 0B f=0/0




Every controller button press stays in sync.



The audio however :



ffplay -f dshow audio="Digital Audio Interface (Cam Link)" -tune zerolatency
ffplay version git-2020-05-01-39fb1e9 Copyright (c) 2003-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200328
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --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-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 43.100 / 56. 43.100
 libavcodec 58. 82.100 / 58. 82.100
 libavformat 58. 42.101 / 58. 42.101
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 80.100 / 7. 80.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, dshow, from 'audio=Digital Audio Interface (Cam Link)':
 Duration: N/A, start: 143092.007000, bitrate: 1411 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
143103.21 M-A: 0.000 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0




Is always behined by about a full second.



I'm not trying to record this, just trying to directly play from the Elgato Cam Link 1:1 output to my computer screen. When this is played in the Elgato Game Capture software, the video and audio are 1:1, no issues. So I know it's not the console or the capture device.