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Valkaama DVD Label
4 octobre 2011, par
Mis à jour : Février 2013
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On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
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List of compatible distributions
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Sur d’autres sites (8942)
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How can i merge 2 videos with audio using ffmpeg ?
4 juin 2020, par Easy Tricks For AndroidI tried this command

ffmpeg -f concat -i mylist.txt -c copy output.mp4

but output.mp4 has no audio. please help me.


the mylist.txt contains these words : file '1.mp4'
file '2.mp4'

this is the log





C :\Users\Admin\OneDrive\Desktop\New folder>ffmpeg -n -i 1.mp4 -i 2.mp4
 ffmpeg version git-2020-05-23-26b4509 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200523
 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. 48.100 / 56. 48.100
 libavcodec 58. 87.101 / 58. 87.101
 libavformat 58. 43.100 / 58. 43.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 83.100 / 7. 83.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1.mp4' :
 Metadata :
 major_brand : mp42
 minor_version : 0
 compatible_brands : mp41isom
 creation_time : 2020-05-31T08:12:56.000000Z
 Duration : 00:00:05.93, start : 0.000000, bitrate : 186 kb/s
 Stream #0:0(und) : Video : h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 184 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata :
 creation_time : 2020-05-31T08:21:56.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '2.mp4' :
 Metadata :
 major_brand : mp42
 minor_version : 0
 compatible_brands : mp41isom
 creation_time : 2020-06-01T03:32:58.000000Z
 Duration : 00:00:12.23, start : 0.000000, bitrate : 1909 kb/s
 Stream #1:0(und) : Video : h264 (Main) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 1708 kb/s, 30 fps, 30 tbr, 30k tbn, 60 tbc (default)
 Metadata :
 creation_time : 2020-06-01T04:17:17.000000Z
 handler_name : VideoHandler
 encoder : AVC Coding
 Stream #1:1(und) : Audio : aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 196 kb/s (default)
 Metadata :
 creation_time : 2020-06-01T04:17:17.000000Z
 handler_name : SoundHandler
 At least one output file must be specified




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I am not able to create an RTMP stream using ffmpeg
11 juin 2020, par Navneet RajuI have set up an rtmp server using Nginx and that seems to be working fine. But when I try to create a RTMP stream using FFmpeg, it gets stuck at a frame and streaming doesn't work(just keeps buffering).



The FFmpeg command run :



ffmpeg -re -i CD_Final_compress.mp4 -vcodec copy -loop -1 -c:a aac -b:a 160k -ar 44100 -strict -2 -f flv rtmp://127.0.0.1/live/bbb




Output :



ffmpeg version N-98112-g7aa7d68 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
 configuration: --prefix=/home/navneet/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/navneet/ffmpeg_build/include --extra-ldflags=-L/home/navneet/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/navneet/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
 libavutil 56. 53.100 / 56. 53.100
 libavcodec 58. 92.100 / 58. 92.100
 libavformat 58. 46.101 / 58. 46.101
 libavdevice 58. 11.100 / 58. 11.100
 libavfilter 7. 86.100 / 7. 86.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'CD_Final_compress.mp4':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.33.100
 Duration: 00:17:33.85, start: 0.000000, bitrate: 521 kb/s
 Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 386 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 126 kb/s (default)
 Metadata:
 handler_name : SoundHandler
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
 Stream #0:1 -> #0:1 (aac (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, flv, to 'rtmp://127.0.0.1/live/bbb':
 Metadata:
 major_brand : isom
 minor_version : 512
 compatible_brands: isomiso2avc1mp41
 encoder : Lavf58.46.101
 Stream #0:0(und): Video: h264 (High) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 386 kb/s, 30 fps, 30 tbr, 1k tbn, 15360 tbc (default)
 Metadata:
 handler_name : VideoHandler
 Stream #0:1(und): Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 160 kb/s (default)
 Metadata:
 handler_name : SoundHandler
 encoder : Lavc58.92.100 aac
frame= 1652 fps= 30 q=-1.0 size= 4895kB time=00:00:54.96 bitrate= 729.5kbits/s speed=0.999x




Note : I am running this on WSL 1 Ubuntu on Windows


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youtube stream from ffmpeg is buffering
1er juin 2020, par BartonsenI'm using ffmpeg running on a Raspberry Pi 3B with 1GB RAM to stream live video on youtube.
In the beginning the audio+video stream is excellent, but after some minutes I see error messages in YT studio, and video starts buffering.
After some more time (could be 30 mins or 1 hr) the youtube stream is gone, although ffmpeg is still running.



ffmpeg is configured like this :



./configure --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi




Running ffmpeg :



pi@raspberrypi:~ $ ffmpeg -thread_queue_size 512 -rtsp_transport udp -i "rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream" -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/mykey
ffmpeg version git-2020-05-01-3c740f2 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
 configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi
 libavutil 56. 43.100 / 56. 43.100
 libavcodec 58. 82.100 / 58. 82.100
 libavformat 58. 42.102 / 58. 42.102
 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
[udp @ 0x3a8c370] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a9ea80] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a8c3e0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3abf4d0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream':
 Metadata:
 title : RTSP Session
 Duration: N/A, start: 0.000000, bitrate: N/A
 Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 20 fps, 20 tbr, 90k tbn, 180k tbc
 Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
 Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))
Press [q] to stop, [?] for help
[aac @ 0x3ae9f00] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/mykey':
 Metadata:
 title : RTSP Session
 encoder : Lavf58.42.102
 Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 20 fps, 20 tbr, 1k tbn, 90k tbc
 Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s
 Metadata:
 encoder : Lavc58.82.100 aac




In youtube studio I see this :



19:36 Good transmission. The quality is excellent.
19:39 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:39 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:39 Warning: The current bit rate (1974.24 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:41 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:41 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:41 Warning: The current bit rate (2151.41 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:43 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:43 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:45 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Warning: The current bit rate (1737.61 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
...
19:54: Error: YouTube does not receive enough video to maintain consistent streaming. Viewers will therefore experience buffering.




What is the problem ? How can I get rid of the buffering ?
I've also tried the below two commands, but found the output to be worse...



ffmpeg -i rtsp://... -c:v libx264 -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output
ffmpeg -i rtsp://... -c:v h264_omx -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output