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  • Ffmpeg change aspect ratio do not succeeded

    23 mars 2020, par Ngoral

    I’m trying to change the aspect ratio of video, cause it’s being showed in a wrong way (it sould be 16:9 but shows 3:4).
    I’ve tried a lot of things, and none worked.
    E.g. I’ve tried to set SAR, but it changes DAR, so the aspect ratio stays the same. Here’s an example :

    ffmpeg -y -i rtmp://localhost/in/air-hdmi -vf "setsar=sar=16/9" -f flv rtmp://localhost/in/ngoraltestffmpeg


       ffmpeg version N-80388-gfd1d84b Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
     configuration: --prefix=/home/anastasia/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/anastasia/ffmpeg_build/include --extra-ldflags=-L/home/anastasia/ffmpeg_build/lib --bindir=/home/anastasia/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
     libavutil      55. 24.100 / 55. 24.100
     libavcodec     57. 46.100 / 57. 46.100
     libavformat    57. 38.101 / 57. 38.101
     libavdevice    57.  0.101 / 57.  0.101
     libavfilter     6. 46.101 /  6. 46.101
     libswscale      4.  1.100 /  4.  1.100
     libswresample   2.  1.100 /  2.  1.100
     libpostproc    54.  0.100 / 54.  0.100

    [flv @ 0x38143c0] audio stream discovered after head already parsed
    [aac @ 0x3818f20] element type mismatch 1 != 0
    [flv @ 0x38143c0] video stream discovered after head already parsed
    Input #0, flv, from 'rtmp://localhost/in/air-hdmi':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
     Duration: 00:00:00.00, start: 181748.084000, bitrate: N/A
       Stream #0:0: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp
       Stream #0:1: Video: h264 (High), yuv420p, 720x576, 25 fps, 25 tbr, 1k tbn, 50 tbc
    [flv @ 0x39bf5a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
       Last message repeated 1 times
    Output #0, flv, to 'rtmp://localhost/in/ngoraltest':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
       encoder         : Lavf57.38.101
       Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 16:9 DAR 20:9], q=2-31, 200 kb/s, 25 fps, 1k tbn, 25 tbc
       Metadata:
         encoder         : Lavc57.46.100 flv
       Side data:
         cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
       Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp
       Metadata:
         encoder         : Lavc57.46.100 libmp3lame
    Stream mapping:
     Stream #0:1 -> #0:0 (h264 (native) -> flv1 (flv))
     Stream #0:0 -> #0:1 (aac (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    [aac @ 0x3a37000] element type mismatch 1 != 0
       Last message repeated 7 times
    [flv @ 0x39bf5a0] Failed to update header with correct duration.ate= 942.7kbits/s speed=2.37x    
    [flv @ 0x39bf5a0] Failed to update header with correct filesize.
    frame=  112 fps= 48 q=31.0 Lsize=     633kB time=00:00:05.18 bitrate= 999.9kbits/s speed=2.23x    
    video:546kB audio:82kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.846813%
    Exiting normally, received signal 2.

    Using setdar=16/9 do not succeeds : it changes PAR, so the result is the same :

    Input #0, flv, from 'rtmp://localhost/in/air-hdmi':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
     Duration: 00:00:00.00, start: 287464.746000, bitrate: N/A
       Stream #0:0: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp
       Stream #0:1: Video: h264 (High), yuv420p, 720x576, 25 fps, 25 tbr, 1k tbn, 50 tbc
    [flv @ 0x3a5ea20] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
       Last message repeated 1 times
    Output #0, flv, to 'rtmp://localhost/in/ngoraltest':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
       encoder         : Lavf57.38.101
       Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 1k tbn, 25 tbc
       Metadata:
         encoder         : Lavc57.46.100 flv
       Side data:
         cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
       Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp
       Metadata:
         encoder         : Lavc57.46.100 libmp3lame
    Stream mapping:
     Stream #0:1 -> #0:0 (h264 (native) -> flv1 (flv))
     Stream #0:0 -> #0:1 (aac (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help

    (When use setsar, the params are 720x576 [SAR 16:9 DAR 20:9], when setdar720x576 [SAR 64:45 DAR 16:9])

    I’ve also tried to apply scale=720:-1 and -aspect 16:9 and all left the same.

    BUT ! Wnen I write ffplay -vf setsar=16/9 rtmp://localhost/in/ngoraltest it shows perfectly what I need.
    What could be thae problem and hoe to solve it ?

    P.S. I’m little bit confused that there’s no onformation about SAR and DAR of input signal, but I can do totally nothing with it.

  • Ffmpeg change aspect ratio do not succeeded

    22 novembre 2016, par Ngoral

    I’m trying to change the aspect ratio of video, cause it’s being showed in a wrong way (it sould be 16:9 but shows 3:4).
    I’ve tried a lot of things, and none worked.
    E.g. I’ve tried to set SAR, but it changes DAR, so the aspect ratio stays the same. Here’s an example :

    ffmpeg -y -i rtmp://localhost/in/air-hdmi -vf "setsar=sar=16/9" -f flv rtmp://localhost/in/ngoraltestffmpeg


       ffmpeg version N-80388-gfd1d84b Copyright (c) 2000-2016 the FFmpeg developers
     built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
     configuration: --prefix=/home/anastasia/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/anastasia/ffmpeg_build/include --extra-ldflags=-L/home/anastasia/ffmpeg_build/lib --bindir=/home/anastasia/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
     libavutil      55. 24.100 / 55. 24.100
     libavcodec     57. 46.100 / 57. 46.100
     libavformat    57. 38.101 / 57. 38.101
     libavdevice    57.  0.101 / 57.  0.101
     libavfilter     6. 46.101 /  6. 46.101
     libswscale      4.  1.100 /  4.  1.100
     libswresample   2.  1.100 /  2.  1.100
     libpostproc    54.  0.100 / 54.  0.100

    [flv @ 0x38143c0] audio stream discovered after head already parsed
    [aac @ 0x3818f20] element type mismatch 1 != 0
    [flv @ 0x38143c0] video stream discovered after head already parsed
    Input #0, flv, from 'rtmp://localhost/in/air-hdmi':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
     Duration: 00:00:00.00, start: 181748.084000, bitrate: N/A
       Stream #0:0: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp
       Stream #0:1: Video: h264 (High), yuv420p, 720x576, 25 fps, 25 tbr, 1k tbn, 50 tbc
    [flv @ 0x39bf5a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
       Last message repeated 1 times
    Output #0, flv, to 'rtmp://localhost/in/ngoraltest':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
       encoder         : Lavf57.38.101
       Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 16:9 DAR 20:9], q=2-31, 200 kb/s, 25 fps, 1k tbn, 25 tbc
       Metadata:
         encoder         : Lavc57.46.100 flv
       Side data:
         cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
       Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp
       Metadata:
         encoder         : Lavc57.46.100 libmp3lame
    Stream mapping:
     Stream #0:1 -> #0:0 (h264 (native) -> flv1 (flv))
     Stream #0:0 -> #0:1 (aac (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    [aac @ 0x3a37000] element type mismatch 1 != 0
       Last message repeated 7 times
    [flv @ 0x39bf5a0] Failed to update header with correct duration.ate= 942.7kbits/s speed=2.37x    
    [flv @ 0x39bf5a0] Failed to update header with correct filesize.
    frame=  112 fps= 48 q=31.0 Lsize=     633kB time=00:00:05.18 bitrate= 999.9kbits/s speed=2.23x    
    video:546kB audio:82kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.846813%
    Exiting normally, received signal 2.

    Using setdar=16/9 do not succeeds : it changes PAR, so the result is the same :

    Input #0, flv, from 'rtmp://localhost/in/air-hdmi':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
     Duration: 00:00:00.00, start: 287464.746000, bitrate: N/A
       Stream #0:0: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp
       Stream #0:1: Video: h264 (High), yuv420p, 720x576, 25 fps, 25 tbr, 1k tbn, 50 tbc
    [flv @ 0x3a5ea20] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
       Last message repeated 1 times
    Output #0, flv, to 'rtmp://localhost/in/ngoraltest':
     Metadata:
       Server          : NGINX RTMP (github.com/arut/nginx-rtmp-module)
       displayWidth    : 720
       displayHeight   : 576
       fps             : 0
       profile         :
       level           :
       encoder         : Lavf57.38.101
       Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 1k tbn, 25 tbc
       Metadata:
         encoder         : Lavc57.46.100 flv
       Side data:
         cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
       Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp
       Metadata:
         encoder         : Lavc57.46.100 libmp3lame
    Stream mapping:
     Stream #0:1 -> #0:0 (h264 (native) -> flv1 (flv))
     Stream #0:0 -> #0:1 (aac (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help

    (When use setsar, the params are 720x576 [SAR 16:9 DAR 20:9], when setdar720x576 [SAR 64:45 DAR 16:9])

    I’ve also tried to apply scale=720:-1 and -aspect 16:9 and all left the same.

    BUT ! Wnen I write ffplay -vf setsar=16/9 rtmp://localhost/in/ngoraltest it shows perfectly what I need.
    What could be thae problem and hoe to solve it ?

    P.S. I’m little bit confused that there’s no onformation about SAR and DAR of input signal, but I can do totally nothing with it.

  • Audio Slowly Desynchronizing When Segmenting

    14 avril 2018, par Nimble

    I use ffmpeg’s ability to segment video while I record so I can record constantly without my hard drive filling up.

    It works really well, expect the audio desynchronizes from the video when the file segments. The video seems to be uninterrupted but I can actually hear a tiny jump in the audio when I join segments later on. One would think that ffmpeg would store packets in a queue during segmentation so nothing is lost but that doesn’t seem to be the case... Any way I could force it to do something like that ?

    Here is my current block :

    ffmpeg -y -thread_queue_size 5096 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M -framerate 100 -pixel_format nv12 ^
    -itsoffset 00:00:00.012 -i video="Video (00 Pro Capture HDMI 4K+)" -thread_queue_size 5096 -guess_layout_max 0 -f dshow ^
    -rtbufsize 2147.48M -i audio="SPDIF/ADAT (1+2) (RME Fireface UC)" -map 0:0,1:0 -map 1:0 -c:v h264_nvenc -preset: llhp ^
    -pix_fmt nv12 -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M -b:a 384k -ac 2 -r 100 -vsync 1 ^
    -max_muxing_queue_size 5096 -segment_time 600 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\PC\PC%02d.mp4

    I am delaying the video stream because right out the gate it’s a little bit ahead of the audio.

    PS : aresample or async seem to have no effect or at least not a desirable one.