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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.
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Support de tous types de médias
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Sur d’autres sites (7490)
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FFMPEG Incredibly Slow On Windows Server 2016
1er août 2019, par Ben GardnerI have a t2.small (2GB RAM, 1 vCPU) Amazon EC2 instance running a process using FFMPEG. It runs just fine ( 30 fps) on my computer (i7, 12GB RAM), but at around 2 fps on the server. Here’s the command :
rescale_command = f'ffmpeg -i {srcVideo} -filter_complex \"scale={owidth}:{oheight}, setsar=1:1, pad={dim[0]}:{dim[1]}:{oofx}:{oofy}\" {destVideo}'
I’ve tried uninstalling and reinstalling ffmpeg/ffprobe
Edited per llogan’s request, I’ve also discovered that it’s relatively speedy towards the beginning (I’m rescaling 100 times over the course of the program) and slows down towards the end.
It utilizes 100% of the CPU even from the beginning, though.
Example command/output :
ffmpeg -i media/8-1-2019/hi/dl-107.mp4 -filter_complex "scale=607:1080, setsar=1:1, pad=1920:1080:656:0" media/8-1-2019/hi/sl-107.mp4
ffmpeg version 3.2 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 34.100 / 55. 34.100
libavcodec 57. 64.100 / 57. 64.100
libavformat 57. 56.100 / 57. 56.100
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'media/8-1-2019/hi/dl-107.mp4':
Metadata:
minor_version : 512
major_brand : isom
compatible_brands: isomiso2avc1mp41
comment : vid:v09044ce0000bks3802jqrog167l3rf0
encoder : Lavf58.20.100
Duration: 00:00:15.12, start: 0.000000, bitrate: 1046 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 540x960 [SAR 1:1 DAR 9:16], 972 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 64 kb/s (default)
Metadata:
handler_name : SoundHandler
[Parsed_setsar_1 @ 00000000023ac840] num:den syntax is deprecated, please use num/den or named options instead
[Parsed_setsar_1 @ 00000000023ad2c0] num:den syntax is deprecated, please use num/den or named options instead
[libx264 @ 00000000023ad800] using SAR=1/1
[libx264 @ 00000000023ad800] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 00000000023ad800] profile High, level 4.0
[libx264 @ 00000000023ad800] 264 - core 148 r2721 72d53ab - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - 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=1 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, mp4, to 'media/8-1-2019/hi/sl-107.mp4':
Metadata:
minor_version : 512
major_brand : isom
compatible_brands: isomiso2avc1mp41
comment : vid:v09044ce0000bks3802jqrog167l3rf0
encoder : Lavf57.56.100
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc (default)
Metadata:
encoder : Lavc57.64.100 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
handler_name : SoundHandler
encoder : Lavc57.64.100 aac
Stream mapping:
Stream #0:0 (h264) -> scale (graph 0)
pad (graph 0) -> Stream #0:0 (libx264)
Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
frame= 453 fps=3.6 q=29.0 Lsize= 3016kB time=00:00:15.01 bitrate=1645.7kbits/s speed=0.12x
video:2763kB audio:237kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.576820%
[libx264 @ 00000000023ad800] frame I:6 Avg QP:20.93 size: 38467
[libx264 @ 00000000023ad800] frame P:118 Avg QP:23.12 size: 11624
[libx264 @ 00000000023ad800] frame B:329 Avg QP:25.19 size: 3725
[libx264 @ 00000000023ad800] consecutive B-frames: 1.8% 2.6% 4.6% 90.9%
[libx264 @ 00000000023ad800] mb I I16..4: 16.4% 76.7% 6.9%
[libx264 @ 00000000023ad800] mb P I16..4: 2.3% 5.8% 0.9% P16..4: 9.8% 3.8% 1.6% 0.0% 0.0% skip:75.7%
[libx264 @ 00000000023ad800] mb B I16..4: 0.4% 0.7% 0.1% B16..8: 9.5% 1.6% 0.3% direct: 0.6% skip:86.8% L0:43.0% L1:50.8% BI: 6.2%
[libx264 @ 00000000023ad800] 8x8 transform intra:67.4% inter:78.3%
[libx264 @ 00000000023ad800] coded y,uvDC,uvAC intra: 32.5% 40.1% 9.4% inter: 2.7% 3.9% 0.1%
[libx264 @ 00000000023ad800] i16 v,h,dc,p: 50% 25% 6% 19%
[libx264 @ 00000000023ad800] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 38% 11% 27% 3% 3% 5% 3% 6% 3%
[libx264 @ 00000000023ad800] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 43% 11% 12% 5% 7% 9% 4% 7% 3%
[libx264 @ 00000000023ad800] i8c dc,h,v,p: 62% 11% 22% 5%
[libx264 @ 00000000023ad800] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 00000000023ad800] ref P L0: 64.4% 14.3% 16.1% 5.2%
[libx264 @ 00000000023ad800] ref B L0: 90.6% 7.5% 2.0%
[libx264 @ 00000000023ad800] ref B L1: 97.7% 2.3%
[libx264 @ 00000000023ad800] kb/s:1496.84
[aac @ 0000000000628120] Qavg: 754.761 -
ffmpeg stream chrome kiosk mode ubuntu 16.04 server
15 février 2021, par RaulI have a weird out-of-sync issue while using ffmpeg to stream to youtube live a chrome browser from an ub untu 16.04 server.



Issue : output video streamed to youtube has audio/video out of sync, sometimes with as much as 3s



Current flow :



1) start pulseaudio - we using something like this to start it :



pulseaudio --start -vvv --disallow-exit --log-target=syslog --high-priority --exit-idle-time=-1 --daemonize




2) start Xvfb



Xvfb :0 -ac -screen 0 1920x1080x24




3) start chrome linux in kiosk mode



google-chrome --kiosk --disable-gpu --incognito --no-first-run --disable-java --disable-plugins --disable-translate --disk-cache-size=$((1024 * 1024)) --disk-cache-dir=/tmp/chrome/ --user-data-dir=/tmp/chrome/ --force-device-scale-factor=1 --window-size=1920,1080 --window-position=0,0 LOCATION_URL




4) start ffmpeg



ffmpeg -y \
 -thread_queue_size 8192 -rtbufsize 250M -f x11grab -video_size 1920x1080 -framerate 24 -i :0 \
 -thread_queue_size 8192 -channel_layout stereo -f alsa -i pulse \
 -c:v libx264 -pix_fmt yuv420p -c:v libx264 -g 48 -crf 24 -filter:v fps=24 -preset ultrafast -tune zerolatency \
 -c:a aac -strict -2 -channel_layout stereo -ab 96k -ac 2 -flags +global_header \
 -f flv YOUTUBE_LIVE_STREAMING_RTMP




Note : this is running on an amazon ec2 instance, meaning there is no soundcard, so alsa and pulseaudio are creating a dummy audio card. However, the latency does not come from there. Logs :



Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Final latency 69.66 ms = 23.22 ms + 2*11.61 ms + 23.22 ms




At this point, here's what we observed :



- 

-
if we start ffmpeg exactly after issuing the command to start chrome, we see the DTS errors from ffmpeg. Audio is out of sync with the video and has delay of 3-5seconds AHEAD. We also noticed the out of sync remains the same for the full duration of the stream
-
if we start ffmpeg after around 10seconds, audio and video are almost in sync. We then manually added a -itsoffset -0.125 to the ffmpeg command and everything is perfect.







Questions :



- 

- Why would ffmpeg have so much lag if it's started right after chrome ?
- Is starting the ffmpeg after 10s or X seconds the expected behavior ? That is, is this because the system needs to wait for audio/video signals to be "ready" or something ?
- Is there a way to 100% calculate or know when Chrome is fully ready and start ffmpeg ? We found sometimes it takes 5s, sometimes 10. Depends on the URL we load.
- Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything. And a restart is required to "re-balance" the audio/video inputs and get them back in sync.
- Can pulseaudio be the problem in this scenario ?













Thank you



UPDATE Dec 20



We were able to do some tricks to force chrome to start the audio on page load, and that will force connect to pulseaudio. Doing that, plus adding a 3s delay for ffmpeg to start, there is no more delay when ffmpeg starts.
However, our app is a webRTC app, and we have a STRANGER thing happening : if we start the page with no webcam/audio, once the webcam/audio is enabled, ffmpeg (while showing no errors) has a delay of 2s or so. While keep talking, in about max 30s, that delay is GONE.



So the new questions are :



- 

- Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything.
- What could cause the initial audio/video out of sync issue and then catching up ?






-
-
FFMPEG "Segmentation fault" with stream source and watermark
4 mars 2020, par MagikeyI use release : 4.2.2 (static) from "https://johnvansickle.com/ffmpeg/"
Final code will be on "Amazon AWS lambda"
Goal : use a url stream and add watermak
Link to video : https://feoval.fr/519.mp4
Link to Watermak : https://feoval.fr/watermark.png
./ffmpeg -i "https://feoval.fr/519.mp4" -i "./watermark.png" -filter_complex "overlay=W-w-10:H-h-10:format=rgb" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4
return "Segmentation fault"
I have the same error on my computer and on AWS Lambda server
./ffmpeg -i "https://feoval.fr/519.mp4" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4
work (but not watermak)
./ffmpeg -i "./519.mp4" -i "./watermark.png" -filter_complex "overlay=W-w-10:H-h-10:format=rgb" -f "mp4" -movflags "frag_keyframe+empty_moov" -pix_fmt "yuv420p" test.mp4
work (but not with stream)
Thanks you very much !
Logs for the first case who return "Segmentation fault" :
...
Stream mapping:
Stream #0:0 (h264) -> overlay:main (graph 0)
Stream #1:0 (png) -> overlay:overlay (graph 0)
overlay (graph 0) -> Stream #0:0 (libx264)
Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, ? for help
[libx264 @ 0x742e480] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x742e480] profile High, level 3.1, 4:2:0, 8-bit
[libx264 @ 0x742e480] 264 - core 159 r2991 1771b55 - H.264/MPEG-4 AVC codec - Copyleft 2003-2019 - 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=12 lookahead_threads=2 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, mp4, to 'test.mp4':
Metadata:
major_brand : mp42
minor_version : 1
compatible_brands: isommp41mp42
encoder : Lavf58.29.100
Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 480x848, q=-1--1, 30 fps, 15360 tbn, 30 tbc (default)
Metadata:
encoder : Lavc58.54.100 libx264
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata:
creation_time : 2020-01-13T08:54:26.000000Z
handler_name : Core Media Audio
encoder : Lavc58.54.100 aac
Segmentation fault (core dumped)