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Autres articles (111)
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HTML5 audio and video support
13 avril 2011, parMediaSPIP uses HTML5 video and audio tags to play multimedia files, taking advantage of the latest W3C innovations supported by modern browsers.
The MediaSPIP player used has been created specifically for MediaSPIP and can be easily adapted to fit in with a specific theme.
For older browsers the Flowplayer flash fallback is used.
MediaSPIP allows for media playback on major mobile platforms with the above (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)
Sur d’autres sites (11199)
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Stream ffmpeg output to flash player in node.js
10 décembre 2013, par daveI'm using ffmpeg to take a video source and live stream it to the browser in flash.
Here's the code (using expressjs) :
app.get('/stream', function(req, res) {
var command = spawn('ffmpeg', ['-re','-i','video.avi','-c:v','libx264','-c:a','libfaac','-ar',44100,'-f','flv','-']);
res.setHeader("Accept-Ranges", "bytes");
res.setHeader("Content-Type", "video/x-flv");
command.stdout.pipe(res);
});EDIT : The command this uses is :
ffmpeg -re -i video.avi -c:v libx264 -c:a libfaac -ar 44100 -f flv -
If I load this URL in the browser directly, it downloads the flv file and I can play it back in VLC player. However, when I use this in a player like JWPlayer or even try to simply embed, it doesn't show or anything.
Any idea what I could be missing here ?
Thank you !
EDIT : Here's the output from ffmpeg, after this it's just the bytes that continue on as it encodes.
ffmpeg version N-52420-gfc69033 Copyright (c) 2000-2013 the FFmpeg developers
built on Apr 25 2013 17:10:30 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libaacplus --enable-libass --enable-libcelt --enable-libfaac --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-openssl --enable-libopus --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --prefix=/usr/local
libavutil 52. 27.100 / 52. 27.100
libavcodec 55. 6.100 / 55. 6.100
libavformat 55. 3.100 / 55. 3.100
libavdevice 55. 0.100 / 55. 0.100
libavfilter 3. 60.101 / 3. 60.101
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
[avi @ 0x7ffb12004800] non-interleaved AVI
Input #0, avi, from 'video.avi':
Metadata:
encoder : AVI-Mux GUI 1.17.8.3, Feb 16 201019:42:50
JUNK :
Duration: 01:58:39.79, start: 0.000000, bitrate: 1710 kb/s
Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 720x304 [SAR 1:1 DAR 45:19], 29.97 tbr, 29.97 tbn, 29.98 tbc
Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 192 kb/s
Metadata:
title : videotest
[libx264 @ 0x7ffb12022000] using SAR=1/1
[libx264 @ 0x7ffb12022000] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
[libx264 @ 0x7ffb12022000] profile High, level 3.0
[libx264 @ 0x7ffb12022000] 264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, flv, to 'pipe:':
Metadata:
JUNK :
encoder : Lavf55.3.100
Stream #0:0: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 720x304 [SAR 1:1 DAR 45:19], q=-1--1, 1k tbn, 29.97 tbc
Stream #0:1: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, 5.1, s16, 128 kb/s
Metadata:
title : videotest
Stream mapping:
Stream #0:0 -> #0:0 (mpeg4 -> libx264)
Stream #0:1 -> #0:1 (ac3 -> libfaac)
Press [q] to stop, [?] for help
FLVme= -43 fps= 28 q=0.0 size= 0kB time=00:00:01.48 bitrate= 2.2kbits/s
videodatarateionframerate@=?Q??s
audiodatarate@_@audiosamplerate@刀audiosamplesize@0stereo
audiocodecid@$JUNKencoder
Lavf55.3.10filesize 8 -d??gd??@?'?c1--?h???"??0 ?B?????E???H??,? ?#??x264 - core 125 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00?Re??g?$?<< -
FFmpeg remux rtp to mpegts [on hold]
16 décembre 2013, par ArdoramorI am trying to remux rtp stream into mptegts format. I have an SDP file with the following contents :
v=0
o=- 0 0 IN IP4 127.0.0.1
s=Unnamed
i=N/A
c=IN IP4 192.168.17.44
t=0 0
a=recvonly
a=orient:portrait
m=video 8202 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=428028;sprop-parameter-sets=Z0KAKJWgKA9E,aM48gA==;
a=control:trackID=1I execute the following ffmpeg command :
ffmpeg -i "test.sdp" -f mpegts -vcodec copy "/tmp/test.ts"
And I get the following information :
Input #0, sdp, from 'test.sdp':
Metadata:
title : Unnamed
comment : N/A
Duration: N/A, start: 0.066622, bitrate: N/A
Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
[mpegts @ 0x1101d4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, mpegts, to '/tmp/test.ts':
Metadata:
title : Unnamed
comment : N/A
encoder : Lavf53.4.0
Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
Stream mapping:
Stream #0.0 -> #0.0I receive the following error :
[mpegts @ 0x1c85f960] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
av_interleaved_write_frame(): Operation not permittedSo I add the suggested bitstream filter :
ffmpeg -i "test.sdp" -f mpegts -vbsf h264_mp4toannexb "/tmp/test.ts"
But the h264 encoding now becomes h262 (mpeg2video) :
~$ffprobe /tmp/test.ts
Input #0, mpegts, from '/tmp/test.ts':
Duration: 00:00:04.13, start: 1.400000, bitrate: 640 kb/s
Program 1
Metadata:
service_name : Unnamed
service_provider: FFmpeg
Stream #0.0[0x100]: Video: mpeg2video (Main), yuv420p, 640x480 [PAR 1:1 DAR 4:3], 104857 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbcIs there any way to keep the h264 codec without re-encoding it ? Doing so becomes very CPU intensive.
Update
Hopefully this will clear up the issue and remove the off-topic stamp.
I'm writing an Android app that is based off of SpyDroids streaming architecture. The app communicates with the server, providing it the SDP. The server spawns an ffmpeg process to remux the incoming video stream into mpegts and broadcasts it on multicast (right now just file).
SpyDroid performs streaming by sending recorded mp4 file through localsocket, received h264 packets, supposedly (according to code removed mp4 h264 prefix [annexb]), wraps it with rtp headrs and sends it on its way. Thus, the RPT stream I get is clearly not originally generated as such.
As @Wagner Patriota has mentioned, I should add '-vcodec copy'. I had run the remuxing with it before as well but the error is still present (full output) :
~$ffmpeg -i "test.sdp" -f mpegts -vcodec copy -vbsf h264_mp4toannexb "/tmp/test.ts"
ffmpeg version 0.8.6, Copyright (c) 2000-2011 the FFmpeg developers
built on Jan 30 2012 17:17:54 with gcc 4.5.2
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --disable-avisynth --enable-libdc1394 --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libx264 --enable-libxvid --extra-cflags='-O2 -g -m64 -mtune=generic -fPIC' --disable-stripping --disable-demuxer=v4l --disable-demuxer=v4l2 --disable-indev=v4l --disable-indev=v4l2
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 7. 0 / 53. 7. 0
libavformat 53. 4. 0 / 53. 4. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 23. 0 / 2. 23. 0
libswscale 2. 0. 0 / 2. 0. 0
libpostproc 51. 2. 0 / 51. 2. 0
[h264 @ 0x16b4b1c0] concealing 232 DC, 232 AC, 232 MV errors
[h264 @ 0x16b4b1c0] concealing 63 DC, 63 AC, 63 MV errors
[h264 @ 0x16b4b1c0] concealing 25 DC, 25 AC, 25 MV errors
[h264 @ 0x16b4b1c0] concealing 138 DC, 138 AC, 138 MV errors
[h264 @ 0x16b4b1c0] concealing 69 DC, 69 AC, 69 MV errors
[sdp @ 0x16b43400] Estimating duration from bitrate, this may be inaccurate
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
Input #0, sdp, from 'test.sdp':
Metadata:
title : Unnamed
comment : N/A
Duration: N/A, start: 0.033256, bitrate: N/A
Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
[mpegts @ 0x16b4a4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, mpegts, to '/tmp/test.ts':
Metadata:
title : Unnamed
comment : N/A
encoder : Lavf53.4.0
Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop, [?] for help
h264_mp4toannexb failed for stream 0, codec copy: Invalid argument
[mpegts @ 0x16b4a4c0] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
av_interleaved_write_frame(): Operation not permittedThe error reports that the invalid argument has been supplied. Increased loglevel does not give any more information. I know that ffmpeg is sometimes finicky with argument order. However, they seen to be in order of documentation as well as suggested order by @Wagner Patriota.
-
FFmpeg remux rtp to mpegts [closed]
16 décembre 2013, par ArdoramorI am trying to remux rtp stream into mptegts format. I have an SDP file with the following contents :
v=0
o=- 0 0 IN IP4 127.0.0.1
s=Unnamed
i=N/A
c=IN IP4 192.168.17.44
t=0 0
a=recvonly
a=orient:portrait
m=video 8202 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=428028;sprop-parameter-sets=Z0KAKJWgKA9E,aM48gA==;
a=control:trackID=1I execute the following ffmpeg command :
ffmpeg -i "test.sdp" -f mpegts -vcodec copy "/tmp/test.ts"
And I get the following information :
Input #0, sdp, from 'test.sdp':
Metadata:
title : Unnamed
comment : N/A
Duration: N/A, start: 0.066622, bitrate: N/A
Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
[mpegts @ 0x1101d4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, mpegts, to '/tmp/test.ts':
Metadata:
title : Unnamed
comment : N/A
encoder : Lavf53.4.0
Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
Stream mapping:
Stream #0.0 -> #0.0I receive the following error :
[mpegts @ 0x1c85f960] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
av_interleaved_write_frame(): Operation not permittedSo I add the suggested bitstream filter :
ffmpeg -i "test.sdp" -f mpegts -vbsf h264_mp4toannexb "/tmp/test.ts"
But the h264 encoding now becomes h262 (mpeg2video) :
~$ffprobe /tmp/test.ts
Input #0, mpegts, from '/tmp/test.ts':
Duration: 00:00:04.13, start: 1.400000, bitrate: 640 kb/s
Program 1
Metadata:
service_name : Unnamed
service_provider: FFmpeg
Stream #0.0[0x100]: Video: mpeg2video (Main), yuv420p, 640x480 [PAR 1:1 DAR 4:3], 104857 kb/s, 60 fps, 60 tbr, 90k tbn, 120 tbcIs there any way to keep the h264 codec without re-encoding it ? Doing so becomes very CPU intensive.
Update
Hopefully this will clear up the issue and remove the off-topic stamp.
I'm writing an Android app that is based off of SpyDroids streaming architecture. The app communicates with the server, providing it the SDP. The server spawns an ffmpeg process to remux the incoming video stream into mpegts and broadcasts it on multicast (right now just file).
SpyDroid performs streaming by sending recorded mp4 file through localsocket, received h264 packets, supposedly (according to code removed mp4 h264 prefix [annexb]), wraps it with rtp headrs and sends it on its way. Thus, the RPT stream I get is clearly not originally generated as such.
As @Wagner Patriota has mentioned, I should add '-vcodec copy'. I had run the remuxing with it before as well but the error is still present (full output) :
~$ffmpeg -i "test.sdp" -f mpegts -vcodec copy -vbsf h264_mp4toannexb "/tmp/test.ts"
ffmpeg version 0.8.6, Copyright (c) 2000-2011 the FFmpeg developers
built on Jan 30 2012 17:17:54 with gcc 4.5.2
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --disable-avisynth --enable-libdc1394 --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libx264 --enable-libxvid --extra-cflags='-O2 -g -m64 -mtune=generic -fPIC' --disable-stripping --disable-demuxer=v4l --disable-demuxer=v4l2 --disable-indev=v4l --disable-indev=v4l2
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 7. 0 / 53. 7. 0
libavformat 53. 4. 0 / 53. 4. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 23. 0 / 2. 23. 0
libswscale 2. 0. 0 / 2. 0. 0
libpostproc 51. 2. 0 / 51. 2. 0
[h264 @ 0x16b4b1c0] concealing 232 DC, 232 AC, 232 MV errors
[h264 @ 0x16b4b1c0] concealing 63 DC, 63 AC, 63 MV errors
[h264 @ 0x16b4b1c0] concealing 25 DC, 25 AC, 25 MV errors
[h264 @ 0x16b4b1c0] concealing 138 DC, 138 AC, 138 MV errors
[h264 @ 0x16b4b1c0] concealing 69 DC, 69 AC, 69 MV errors
[sdp @ 0x16b43400] Estimating duration from bitrate, this may be inaccurate
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
Input #0, sdp, from 'test.sdp':
Metadata:
title : Unnamed
comment : N/A
Duration: N/A, start: 0.033256, bitrate: N/A
Stream #0.0: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
[mpegts @ 0x16b4a4c0] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, mpegts, to '/tmp/test.ts':
Metadata:
title : Unnamed
comment : N/A
encoder : Lavf53.4.0
Stream #0.0: Video: libx264, yuv420p, 640x480, q=2-31, 90k tbn, 90k tbc
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop, [?] for help
h264_mp4toannexb failed for stream 0, codec copy: Invalid argument
[mpegts @ 0x16b4a4c0] h264 bitstream malformated, no startcode found, use -vbsf h264_mp4toannexb
av_interleaved_write_frame(): Operation not permittedThe error reports that the invalid argument has been supplied. Increased loglevel does not give any more information. I know that ffmpeg is sometimes finicky with argument order. However, they seen to be in order of documentation as well as suggested order by @Wagner Patriota.