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Head down (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Echoplex (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Discipline (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Letting you (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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1 000 000 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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999 999 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
Autres articles (92)
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Mise à jour de la version 0.1 vers 0.2
24 juin 2013, parExplications des différents changements notables lors du passage de la version 0.1 de MediaSPIP à la version 0.3. Quelles sont les nouveautés
Au niveau des dépendances logicielles Utilisation des dernières versions de FFMpeg (>= v1.2.1) ; Installation des dépendances pour Smush ; Installation de MediaInfo et FFprobe pour la récupération des métadonnées ; On n’utilise plus ffmpeg2theora ; On n’installe plus flvtool2 au profit de flvtool++ ; On n’installe plus ffmpeg-php qui n’est plus maintenu au (...) -
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 ;
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
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Sur d’autres sites (12475)
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Duration of source video and subtracted audio are different
22 avril 2018, par zpcThe duration of source video and subtracted wav audio is different , why ?
I’m recorgnizing subtitle from audio, and I need to add subtitle back to video. So I want the duration of audio and video the same.
ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y
My CLI :
[zhangpengcheng@mobiledev03v ifly]$ ffprobe http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 2>&1 | grep Duration
Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s
[zhangpengcheng@mobiledev03v ifly]$ ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y
ffmpeg version 3.1.3 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
configuration: --prefix=./build/ --enable-shared --enable-static --enable-libx264 --enable-avisynth --enable-libass --enable-libfdk-aac --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libopencv --enable-librtmp --enable-gpl --enable-nonfree
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 41.100 / 57. 41.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 47.100 / 6. 47.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Input #0, hls,applehttp, from 'http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8':
Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s
Program 0
Metadata:
variant_bitrate : 0
Stream #0:0: Video: h264 (Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m/bt709/bt709), 668x376, 15 tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, mono, fltp, 65 kb/s
[wav @ 0x10182e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'test.wav':
Metadata:
ISFT : Lavf57.41.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Metadata:
encoder : Lavc57.48.101 pcm_s16le
Stream mapping:
Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size= 31904kB time=00:06:51.84 bitrate= 634.6kbits/s speed= 146x
video:0kB audio:31904kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000239%
[zhangpengcheng@mobiledev03v ifly]$ ffprobe test.wav 2>&1 | grep Duration
Duration: 00:06:10.40, bitrate: 705 kb/s -
FFMPEG HLS stream for Android and IOS
30 juin, par PodaI'm trying to stream to mobile devices with ffmpeg and apache2.2 but I haven't been successful.



I used this command to create the segments and the playlist :



ffmpeg -i http://x.x.x.x:8080 -codec:v libx264 -r 25 -pix_fmt yuv420p -profile:v baseline -level 3 -b:v 500k -s 640x480 -codec:a aac -strict experimental -ac 2 -b:a 128k -movflags faststart -flags -global_header -map 0 -f hls -hls_time 10 -hls_list_size 5 -hls_allow_cache 0 -sc_threshold 0 -hls_flags delete_segments -hls_segment_filename out%05d.ts list.m3u8




The source is a http stream which is streamed by VLC media player.



Example content of the list.m3u8 file :



#EXTM3U
#EXT-X-VERSION:3
#EXT-X-ALLOW-CACHE:NO
#EXT-X-TARGETDURATION:10
#EXT-X-MEDIA-SEQUENCE:89
#EXTINF:10.000000,
out00089.ts
#EXTINF:10.000000,
out00090.ts
#EXTINF:10.000000,
out00091.ts
#EXTINF:10.000000,
out00092.ts
#EXTINF:9.000000,
out00093.ts
#EXT-X-ENDLIST




I created another playlist file - playlist.m3u8 :



#EXTM3U
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=512000
http://x.x.x.x/list.m3u8




If I open this (playlist.m3u8) file in VLC media player then it plays.
It also works in desktop chrome and desktop firefox browsers with Video-js plugin flash fallback.



I set the correct MIME types to the .ts and .m3u8 files in .htaccess file :



AddType application/x-mpegURL .m3u8
AddType video/MP2T .ts




FFprobe output for playlist.m3u8 :



Input #0, hls,applehttp, from 'playlist.m3u8':
 Duration: N/A, start: 1.400000, bitrate: N/A
 Program 0
 Metadata: variant_bitrate : 512000
Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 90k tbn, 50 tbc
Metadata: variant_bitrate : 512000
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 128 kb/s
Metadata: variant_bitrate : 512000




What should I do to make it work ?



UPDATE



It works if I provide a link to list.m3u8 file (created by ffmpeg).


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Use ffmpeg to stream live content to azure media services
19 mars 2018, par DadicoolI’ve been trying to stream content to azure media services using ffmpeg as it’s one of the options described here : http://azure.microsoft.com/blog/2014/09/18/azure-media-services-rtmp-support-and-live-encoders/
My command is :
ffmpeg -v verbose -i 300.mp4 -strict -2 -c:a aac -b:a 128k -ar 44100 -r 30 -g 60 -keyint_min 60 -b:v 400000 -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7
I have made sure that the streaming endpoint has one active streaming unit, I also made sure that the channel is actually Ready and I even get it to start streaming (which makes a PublishURL available).
When I execute the ffmpeg command to start streaming, I keep getting the following error :
ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers
built on Dec 30 2014 11:31:18 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 --arch=x86_64 --enable-runtime-cpudetect
libavutil 54. 15.100 / 54. 15.100
libavcodec 56. 13.100 / 56. 13.100
libavformat 56. 15.102 / 56. 15.102
libavdevice 56. 3.100 / 56. 3.100
libavfilter 5. 2.103 / 5. 2.103
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Routing option strict to both codec and muxer layer
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] overread end of atom 'colr' by 1 bytes
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] stream 0, timescale not set
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] max_analyze_duration 5000000 reached at 5003637 microseconds
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '300.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42isomavc1
creation_time : 2014-01-11 05:39:32
genre : Trailer
artist : Warner Bros.
title : 300: Rise of an Empire - Trailer 2
encoder : HandBrake 0.9.9 2013051800
date : 2014
Duration: 00:02:33.24, start: 0.000000, bitrate: 7377 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 (1920x1088), 7219 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc (default)
Metadata:
creation_time : 2014-01-11 05:39:32
encoder : JVT/AVC Coding
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 157 kb/s (default)
Metadata:
creation_time : 2014-01-11 05:39:32
Stream #0:2: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 101x150 [SAR 72:72 DAR 101:150], 90k tbr, 90k tbn, 90k tbc
rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7: Input/output errorThe Azure blog post clearly states that this should be possible but I can’t find a working example anywhere.
Environment :
- MacOS Maverick
- FFMPEG installed from official build
- 300.mp4 : 1080p trailer of the latest 300 movie