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Spoon - Revenge !
15 septembre 2011, par
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Sur d’autres sites (13375)
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Facing delay in mic audio while using ffmpeg with h264_nvenc encoder [closed]
30 avril 2023, par Aniket BoseI am trying to capture my windows10 desktop with desktop duplication api and on-gpu D3D11 textures, in the form of ffmpeg D3D11VA frames @60 fps. Required command is given here. On top of that I am trying to also intercept my mic audio. For that I am using this command,


ffmpeg -init_hw_device d3d11va -filter_complex ddagrab=framerate=60 -f dshow -i audio="Microphone (High Definition Audio Device)" -c:v h264_nvenc -rc vbr -cq 24 -qmin 24 -qmax 24 -profile:v main -b:v 0K output.mp4



After the desktop capture process when I am watching output.mp4 I am facing an audio delay. More Precisely my mic audio is coming after 28/29 video frames. ie. (28/60)*1000 = 466 ms delay in audio.


I tried to capture @30fps but no improvement. Now I am getting 14 frames delay ie. (14/30)*1000 = 466 ms delay in audio.


After some research I came to know about keyframe intervals. at default h264_nvenc uses 200 sec keyframe interval. So i tried to lower that with the -g option of h264_nvenc encoder. No improvement here too.


One possible solution could be to delay my video using -itsoffset option. These stack overflow and superuser solutions only work with pre-recorded videos,




In ffmpeg, how to delay only the audio of a .mp4 video without converting the audio ?


But I am capturing and encoding on the go (ie. at same time).


I am a novice at video editing and ffmpeg. I am using latest release build from gyan.dev.


ffmpeg version 6.0-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
 configuration: 
--enable-gpl --enable-version3 --enable-static --disable-w32threads 
--disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls 
--enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma 
--enable-libsnappy --enable-zlib --enable-librist --enable-libsrt 
--enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray 
--enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libdav1d 
--enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e 
--enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl 
--enable-libopenjpeg --enable-libvpx --enable-mediafoundation 
--enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi 
--enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg 
--enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec 
--enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 
--enable-libvpl --enable-libshaderc --enable-vulkan --enable-libplacebo 
--enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug 
--enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame 
--enable-libshine --enable-libtheora --enable-libtwolame 
--enable-libvo-amrwbenc --enable-libilbc --enable-libgsm 
--enable-libopencore-amrnb --enable-libopus --enable-libspeex 
--enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite 
--enable-libmysofa --enable-librubberband --enable-libsoxr 
--enable-chromaprint
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...



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A Comprehensive Guide to Robust Digital Marketing Analytics
30 octobre 2023, par Erin -
FFMPEG update from 5.0 to 6.0 out_0_0 buffer queued [closed]
16 mai 2023, par KevittoI've been using ffmpeg 5.0 for some time, encoding an audio stream to an rtp server, but since I updated to ffmpeg 6.0 I get this :


[out_0_0 @ 0x55ac187b60] 100 buffers queued in out_0_0, something may be wrong.



Below is the ffmpeg call :


ffmpeg -re -f alsa -i default:CARD:card1 -ac 2 -af aresample=async=1 -acodec libopus -b:a 48000 -f rtp "rtp://127.0.0.1:5002"



And here is the full startup log :


ffmpeg version 549430e Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 10 (Debian 10.2.1-6)
 configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --extra-libs='-lpthread -lm -latomic' --arch=arm64 --enable-gmp --enable-gpl --enable-libopus --enable-nonfree --enable-version3 --target-os=linux --enable-pthreads --enable-openssl --enable-hardcoded-tables
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'default:CARD=pisound':
 Duration: N/A, start: 1684250059.973334, bitrate: 1536 kb/s
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://127.0.0.1:5002':
 Metadata:
 encoder : Lavf60.3.100
 Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, 48 kb/s
 Metadata:
 encoder : Lavc60.3.100 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 60.3.100
m=audio 5002 RTP/AVP 97
b=AS:48
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1

size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
[out_0_0 @ 0x559f530c60] 100 buffers queued in out_0_0, something may be wrong.
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.93x 
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.797x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.703x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.624x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.562x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.511x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.939x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.866x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.805x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.751x 
size= 10kB time=00:00:05.69 bitrate= 14.8kbits/s speed=0.707x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 13kB time=00:00:05.95 bitrate= 17.8kbits/s speed=0.696x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.939x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.89x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.847x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.806x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.77x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 21kB time=00:00:11.37 bitrate= 14.8kbits/s speed=0.981x 



I tried changing the output to
-f null /dev/null
to see if the rtp was the issue, but I get the same thing. I made sure the user running it was a member to the "audio" group andarecord -l
andaplay -l
both show the card with the right name and information. I even tried to use its hw code instead of the default name, and same issue.