August 2019 - ffmpeg-user - lists.ffmpeg.org

Re: [FFmpeg-user] JPGs->video1 ; MOV -> Mp4 video2 ; Concatenate - video2 audio OK but video is slow . .
by Philip Rhoades 18 Mar '23

18 Mar '23

Carl, There has been some improvement - I will top-post for clarity but the slow motion video has disappeared and the only remaining problem is that the audio and video are out of sync: I stuck with concat because it alllows me to specify image files more conveniently in the concat_demuxer?.txt file: ffmpeg -f concat -i concat_demuxer1.txt -vf scale=1024:576 -r 30000/1001 output1.mp4 ffmpeg -f concat -i concat_demuxer2.txt -vf scale=1024:576 -r 30000/1001 output2.mp4 ffmpeg -f concat -i concat_demuxer3.txt -vf scale=1024:576 -r 30000/1001 output3.mp4 - see attached output. for i in 1 2 3 do ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -i /dev/zero \ -i output${i}.mp4 \ -vcodec copy \ -acodec aac \ -shortest output${i}_silent.mp4 done - see attached output. #!/bin/sh ffmpeg -f concat -i inputs.txt -vcodec copy -acodec copy 2019-03-22_#1_XR_ClimateEmergencyDeclarationDayNSW_LoRes.mp4 - see attached output. On 2019-03-27 10:48, Carl Eugen Hoyos wrote: > 2019-03-27 0:23 GMT+01:00, Philip Rhoades <phil(a)pricom.com.au>: > >> On 2019-03-27 07:22, Carl Eugen Hoyos wrote: >>> 2019-03-26 21:15 GMT+01:00, Philip Rhoades <phil(a)pricom.com.au>: >>> >>>> I converted 21 JPGs to a video with: >>> >>> Usually, converting jpgs with the image2 input works better than >>> concat. >> >> Right - I will try that. > > This is probably not your issue but the concat demuxer > has the disadvantage that the console output is not > very helpful. > >>>> ffmpeg -f concat -i concat_demuxer1.txt -vf scale=1024:576 -r >>>> 29.97 >>> >>> This framerate is never correct, a possible frame rate is 30000/1001 >> >> Hmm . . I was just using what was reported for the frame for the MOVs >> . >> . so using: >> >> "-r 30000/1001" >> >> is OK? > > It is better if that is the framerate of the mov (as I assume). > >>> You can add the audio already in this step. >> >> I did have a few attempts at that but couldn't get it working so I >> resorted to a two-step process - I will have another look. >> >> >>>> output1.mp4 >>>> >>>> I then added a silent audio track with: >>>> >>>> ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -channel_layout >>>> 2.1 >>> >>> The channel number and layout do not match. >> >> Should it be just "stereo"? > > Or drop channel_layout, for -ac 2 it should not be needed. > >>>> -i /dev/zero \ >>>> -i output${i}.mp4 \ >>>> -vcodec copy \ >>>> -acodec aac \ >>>> -shortest output${i}_silent.mp4 >>> >>>> I had another MOV file (fr=29.97) that I converted with: >>>> >>>> newfn=`basename $fn .MOV` >>>> ffmpeg -i $fn -vf scale=1024:576 ./${newfn}.mp4 >>>> >>>> When I concatenate them with (and other converted JPGs and MOVS on >>>> the >>>> end) using: >>>> >>>> ffmpeg -f concat -i inputs.txt -vcodec copy -acodec copy \ >>>> 2019-03-22_#1_XR_ClimateEmergencyDeclarationDayNSW_LoRes.mp4 >>>> >>>> the concat succeeds OK but when I play the video, the audio is OK >>>> but >>>> then when it gets to the converted MOV section, people are moving in >>>> slow motion . . >>>> >>>> Another odd thing, the first 21 converted JPGs at 4 sec each >>>> correctly >>>> shows 84 sec when viewed with mpv but when the converted MOV starts >>>> it >>>> jumps to ~3m 50s . . >>>> >>>> So there is at least one problem but maybe two? Can anyone tell me >>>> what >>>> I am missing? >>> >>> You may have to re-encode, feel free to provide the complete, uncut >>> console output. > > ..feel free to provide the complete, uncut console output for the > command line you already tried and also ffmpeg -i output for > the input files (as concat doesn't show it). > > Carl Eugen -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: phil(a)pricom.com.au

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Recording audio from mic and speakers from commandline
by Mohan G 28 Sep '21

28 Sep '21

Hi, I am using ffmpeg version 2.4.7 on Fedora-21 workstation. I want to record audio, simultaneously from mic and speakers as separate streams. I am using the following command for it. ffmpeg -f alsa -i pulse -f alsa -i default -acodec libmp3lame -map 0:0 -map 1:0 outfile.mkv But after executing this command in terminal, I have to goto "Volume Control" settings in GUI, then goto "Recording" tab, change the "capture from" of one channel to "Built-in Audio Analog Stereo" (i.e., Mic) and the "capture from" of another channel to "Monitor of Built-in Audio Analog Stereo" (i.e., Speakers). But how can this be done completely from command-line, for automating it to use in scripts. - Mohan

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How to have a secure UDP BroadCast
by hassan shatnawi 19 Aug '20

19 Aug '20

Dears, How to have a secure connection with username and password while broadcasting "UPD" I've use this command ffmpeg -f dshow -video_size 1280x720 -rtbufsize 702000K -framerate 30 -i video="Video (00-0 Pro Capture Quad HDMI)" -r 30 -threads 4 -vcodec libx264 -crf 0 -preset ultrafast -f mpegts "udp://username:password@192.168.1.12:6666" but it didn't work can you advice me plz. Thanks in advance. *Hassan Shatnawi* Software Developer *Software Development Department* *Haupshy Establishment for Electronics* Al Madina Al Monawara St. bld 31 | P.O.Box 6875 Amman 11118, Jordan. *T: *+962 6 5820 927* | F: *+962 6 5820 926* | M: *+962 78 5792 096 *E*: hassan(a)haupshy.com *|* www.haupshy.com

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delay time in live ultrasound converter
by Michael Koch 18 Apr '20

18 Apr '20

Hello, I'm using the following batch file (Windows 7) for a live ultrasound conversion. set "SR=44100" :: Sample rate set "F=14000" :: Subtracted frequency set "VOL=10" :: Volume factor set /a "N=4096*%F%/%SR%" :: N = 4096 * F / SR c:\ffmpeg\ffmpeg -f dshow -sample_rate %SR% -sample_size 16 -channels 2 -i audio="Microphone (SoundMAX Integrated" -af volume=%VOL%,afftfilt='real=if(lt(b+%N%,nb),real(b+%N%,ch),0)':'imag=if(lt(b+%N%,nb),imag(b+%N%,ch),0)' -f nut - | c:\ffmpeg\ffplay - FFmpeg gets the audio input from the computer's microphone input and passes the converted output to FFplay. This works fine for hearing bats, but it has about 1-2 seconds delay between input and output. Are there any options for minimizing the delay? The console output is copied below. Thanks, Michael C:\Users\mkoch\Desktop\Ultrasound Live>c:\ffmpeg\ffmpeg -f dshow -sample_rate 44 100 -sample_size 16 -channels 2 -i audio="Microphone (SoundMAX Integrated" -af v olume=10,afftfilt='real=if(lt(b+1300,nb),real(b+1300,ch),0)':'imag=if(lt(b+1300, nb),imag(b+1300,ch),0)' -f nut - | c:\ffmpeg\ffplay - ffmpeg version N-92857-g3e461f9d4c Copyright (c) 2000-2018 the FFmpeg developers built with gcc 8.2.1 (GCC) 20181201 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfi g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib freetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy -- enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-l ibwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 -- enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --en able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --en able-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enab le-avisynth --enable-libopenmpt libavutil 56. 25.100 / 56. 25.100 libavcodec 58. 42.104 / 58. 42.104 libavformat 58. 25.100 / 58. 25.100 ffplay version N-92857-g3e461f9d4c libavdevice 58. 6.101 / 58. 6.101 Copyr ight (c) 2003-2018 the FFmpeg developers libavfilter 7. 46.101 / 7. 46.101 built with gcc 8.2.1 (GCC) 20181201 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfi g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-lib freetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amr wb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy -- enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-l ibwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 -- enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --en able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --en able-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --e nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enab le-avisynth --enable-libopenmpt libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 libavutil 56. 25.100 / 56. 25.100 libavcodec 58. 42.104 / 58. 42.104 libavformat 58. 25.100 / 58. 25.100 libavdevice 58. 6.101 / 58. 6.101 libavfilter 7. 46.101 / 7. 46.101 libswscale 5. 4.100 / 5. 4.100 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 Guessed Channel Layout for Input Stream #0.0 : stereo 0B f=0/0 Input #0, dshow, from 'audio=Microphone (SoundMAX Integrated': Duration: N/A, start: 1701.243000, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> vorbis (libvorbis)) Press [q] to stop, [?] for help Output #0, nut, to 'pipe:': aq= 0KB vq= 0KB sq= 0B f=0/0 Metadata: encoder : Lavf58.25.100 Stream #0:0: Audio: vorbis (libvorbis) (oV[0][0] / 0x566F), 44100 Hz, stereo , fltp Metadata: encoder : Lavc58.42.104 libvorbis Input #0, nut, from 'pipe:': Metadata: encoder : Lavf58.25.100 Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Audio: vorbis (oV[0][0] / 0x566F), 44100 Hz, stereo, fltp Metadata: encoder : Lavc58.42.104 libvorbis 16.27 M-A: 0.000 fd= 0 aq= 3KB vq= 0KB sq= 0B f=0/0

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Can FFmpeg convert from ArriRaw MXF sources?
by Harvey Pikelberger 19 Nov '19

19 Nov '19

Is FFmpeg capable of transcoding from ArriaRaw to h.264?

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denoise after scaling or before?
by Jon bae 27 Oct '19

27 Oct '19

Hello, I just have an general question: As I know denoiser are specialized to specific noise pattern. Makes it a different when I denoise after scaling a video, or is the quality better when I denoise before scaling. In terms of speed it would be better for me, denoising after scaling, but when the quality is better, then I would go that way. Have a nice day! Jonathan

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[avi @ 0x9873ba0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 [avi @ 0x987d2a0] Application provided invalid, non monotonically increasing dts to muxer only when using tee muxer
by Michael Kohne 11 Sep '19

11 Sep '19

I'm using ffmpeg 4.1 to grab an rtsp stream (includes both H.264 video and G.711 audio) from a Sony EP580 IP camera. If I transcode to mpeg4v2 and aac to a single avi file, things seem to work fine. This command line will go for as long as I like (I regularly do 15 minutes at a go). ffmpeg -min_port 62000 -max_port 62004 -i rtsp://192.168.0.113/media/video1 -codec:v msmpeg4v2 -codec:a ac3 -ar 44100 -map 0 -f avi /data/PatientData/vidtmp/test.avi If I instead use the tee muxer to make two avi files: ffmpeg -min_port 62000 -max_port 62004 -i rtsp://192.168.0.113/media/video1 -codec:v msmpeg4v2 -codec:a ac3 -ar 44100 -map 0 -f tee "[f=avi]/data/PatientData/vidtmp/one.avi|[f=avi]/data/PatientData/vidtmp/two.avi" Then within 2 minutes of starting the stream, it fails with the errors: [avi @ 0x9873ba0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 [avi @ 0x987d2a0] Application provided invalid, non monotonically increasing dts to muxer in stream 1: 1104 >= 1104 I've repeated these tests over and over for a week or so now, so I am fairly confident that it's not simply that the camera is janky some of the time. I've also tested using the rtsp stream 'rtsp:// 184.72.239.149/vod/mp4:BigBuckBunny_175k.mov'. That stream does not produce any errors with the tee muxer. Note that those errors both appear to come from the avi muxers, not the tee muxer, which is why I find it surprising: if the camera's output causes this, then why doesn't it happen with the single avi muxer? Test note: the tee invocation is the simplest command line that I could come up with that produces the problem. In the real world one avi muxer goes to a pipe for a live output display, and the other goes to the segment muxer to produce output files in manageable chunks. I did also try this with the latest ffmpeg snapshot (as of yesterday morning), and there was no difference in behavior. Any clues as to how to debug this would be greatly appreciated. ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-11) configuration: --prefix=/data/SHR4634/ffmpeg/build --extra-cflags=-I/data/SHR4634/ffmpeg/build/include --extra-ldflags=-L/data/SHR4634/ffmpeg/build/lib --extra-libs='-lm -ldl -lpthread -lrt' --enable-gpl --enable-nonfree --disable-libfdk_aac --enable-libmp3lame --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libfreetype --enable-libspeex --enable-libtheora --cpu=i686 --enable-runtime-cpudetect libavutil 56. 22.100 / 56. 22.100 libavcodec 58. 35.100 / 58. 35.100 libavformat 58. 20.100 / 58. 20.100 libavdevice 58. 5.100 / 58. 5.100 libavfilter 7. 40.101 / 7. 40.101 libswscale 5. 3.100 / 5. 3.100 libswresample 3. 3.100 / 3. 3.100 libpostproc 55. 3.100 / 55. 3.100 OS is CentOS 6, 32 bit. Michael Kohne Senior Software Engineer Office: 215.283.0860 x208 mhkohne(a)moberg.com -- Celebrating 20 Years Transforming Neurocritical Care Moberg Research, Inc. 224 S Maple Street, Ambler, PA 19002 24/7 Customer Support: 888.662.7246 www.moberg.com <https://www.moberg.com/>

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coding video for some old Sony
by Erik Dobák 06 Sep '19

06 Sep '19

Hi, i have an old Sony Bravia which accepts only MPEG4 as video and MP3 as audio. If i want to watch videos i do it with this command: ffmpeg -i the_kingdom_of_dreams_and_madness.mp4 -b:v 1M -vcodec mpeg4 -vtag xvid -acodec libmp3lame -q:a 2 the_kingdom_of_dreams_and_madness.avi but for some reason some video do not play at all. some do. is there a way that the input video could influence the output? I thought that when i resample with ffmpeg to these basic codecs all will work... Any hints? E

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Unable to record VP8+Opus from RTP+SDP
by Juan Navarro 03 Sep '19

03 Sep '19

Hi, I'd like to record an RTP stream with one VP8 video and one OPUS audio; these are originated from Chrome and then distributed as plain RTP streams. I've written a basic SDP file to use as input for ffmpeg, but my results have been mixed: 1) Recording only the video works fine with a WEBM container (using "-vcodec copy -an -f webm"). ffmpeg is able to automatically find out properties of the video, such as the resolution, and it stores it fine. Other players (VLC tested) are able to open the file. 2) Recording only the audio is possible, again if using WEBM (with "-acodec copy -vn -f webm"), but seems that no Opus headers/metadata are stored. While ffplay is able to play the recorded file, other players complain about it. E.g. VLC says "opus error: cannot read Opus header", "opus error: initial Opus header is corrupted". There is, however, no error or warning message printed out from the debug FFmpeg output. Interestingly enough, using other container, e.g. OGG (with "-f opus" and a .ogg output file), then the recording won't start at all due to missing extradata: [opus @ 0x5e5d380] No extradata present Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input 3) Recording of both audio and video in WEBM or Matroska doesn't work, ffmpeg complains about not being able to get the video resolution (but it was able to do so when recording only video): [sdp @ 0x6094a00] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, yuv420p): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [webm @ 0x6098640] dimensions not set Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument Note that I've already tried setting analyzeduration and probesize to 10M, this takes a long time to start the ffmpeg command but in the end doesn't really change the outcome. This is the input SDP: v=0 o=- 0 0 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 5006 RTP/AVP 109 a=rtpmap:109 opus/48000/2 m=video 5004 RTP/AVP 120 a=rtpmap:120 VP8/90000 And these are the commands: # Audio recording ffmpeg -protocol_whitelist file,rtp,udp \ -fflags +genpts -i input.sdp \ -map 0:a:0 -acodec copy -vn \ -f webm -flags +global_header -y recording.webm # Video recording ffmpeg -protocol_whitelist file,rtp,udp \ -fflags +genpts -i input.sdp \ -map 0:v:0 -vcodec copy -an \ -f webm -flags +global_header -y recording.webm # Audio+Video recording, this is the final use case I'd like to end up solving, although being able to run the previous two commands will help me learn and understand more about how FFmpeg works: ffmpeg -protocol_whitelist file,rtp,udp -nostdin -loglevel debug \ -fflags +genpts -i input.sdp \ -map 0:a:0 -map 0:v:0 -acodec copy -vcodec copy \ -f webm -flags +global_header -y recording.webm Given how easy it is to break the recording of the audio (eg. when trying OGG format, instead of WEBM), I'm inclined to think that these problems are matter of finding the right options for some of the ffmpeg commands, and make the streams play well with the container. I want to avoid transcoding, so all codecs should be "copy". After all, the target here is to just store incoming media from Chrome, not to reencode it... Could anyone give me a hand here? PS. If anyone has a solution and wants the karma, I posted this question in SuperUser: https://superuser.com/questions/1468786/rtp-ffmpeg-dimensions-not-set-only-… ---- Full ffmpeg debug output: ffmpeg version 4.2-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2019 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 / 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 / 55. 5.100 Splitting the commandline. Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,rtp,udp'. Reading option '-nostdin' ... matched as option 'stdin' (enable or disable interaction on standard input) with argument 0. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-fflags' ... matched as AVOption 'fflags' with argument '+genpts'. Reading option '-i' ... matched as input url with argument 'input.sdp'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:a:0'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:v:0'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'copy'. Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'. Reading option '-f' ... matched as option 'f' (force format) with argument 'webm'. Reading option '-flags' ... matched as AVOption 'flags' with argument '+global_header'. Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'. Reading option 'recording.webm' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option nostdin (enable or disable interaction on standard input) with argument 0. Applying option loglevel (set logging level) with argument debug. Applying option y (overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input url input.sdp. Successfully parsed a group of options. Opening an input file: input.sdp. [NULL @ 0x6094a00] Opening 'input.sdp' for reading [sdp @ 0x6094a00] Format sdp probed with size=2048 and score=50 [sdp @ 0x6094a00] audio codec set to: opus [sdp @ 0x6094a00] audio samplerate set to: 48000 [sdp @ 0x6094a00] audio channels set to: 2 [sdp @ 0x6094a00] video codec set to: vp8 [udp @ 0x609d480] end receive buffer size reported is 131072 [udp @ 0x6096f80] end receive buffer size reported is 131072 [sdp @ 0x6094a00] setting jitter buffer size to 500 [udp @ 0x6097e80] end receive buffer size reported is 131072 [udp @ 0x6097600] end receive buffer size reported is 131072 [sdp @ 0x6094a00] setting jitter buffer size to 500 [sdp @ 0x6094a00] Before avformat_find_stream_info() pos: 157 bytes read:157 seeks:0 nb_streams:2 [sdp @ 0x6094a00] max_analyze_duration 5000000 reached at 5000000 microseconds st:0 [sdp @ 0x6094a00] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, yuv420p): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options [sdp @ 0x6094a00] After avformat_find_stream_info() pos: 157 bytes read:157 seeks:0 frames:252 Input #0, sdp, from 'input.sdp': Metadata: title : - Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0, 252, 1/48000: Audio: opus, 48000 Hz, stereo, fltp Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, yuv420p, 90k tbr, 90k tbn, 90k tbc Successfully opened the file. Parsing a group of options: output url recording.webm. Applying option map (set input stream mapping) with argument 0:a:0. Applying option map (set input stream mapping) with argument 0:v:0. Applying option acodec (force audio codec ('copy' to copy stream)) with argument copy. Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy. Applying option f (force format) with argument webm. Successfully parsed a group of options. Opening an output file: recording.webm. [file @ 0x60e7000] Setting default whitelist 'file,crypto' Successfully opened the file. [webm @ 0x6098640] dimensions not set Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Last message repeated 1 times [AVIOContext @ 0x6123800] Statistics: 0 seeks, 0 writeouts [AVIOContext @ 0x6095480] Statistics: 157 bytes read, 0 seeks

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ffprobe aspect ration
by Jonathan Baecker 01 Sep '19

01 Sep '19

Hello, I figure out that ffprobe not always show the "display_aspect_ratio", but mediainfo does. Is there any technical reason for that? Or is there another way to get this info, instead of using "-show_streams". I could calculate the aspect from resolution but how could I be sure that the pixels are square? Here are a uncut output: ffprobe -i D:\char.mp4 -select_streams v:0 -show_streams ffprobe version N-94265-gecdb6fbe18 Copyright (c) 2007-2019 the FFmpeg developers built with gcc 9.1.0 (Rev3, Built by MSYS2 project) configuration: --enable-sdl2 --enable-gmp --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-libx265 --enable-fontconfig --enable-libfreetype --enable-libmysofa --enable-libopenjpeg --enable-libsoxr --enable-libtwolame --enable-libwavpack --enable-libwebp --enable-libxml2 --enable-libzimg --enable-gpl --enable-avisynth --enable-chromaprint --enable-libfdk-aac --enable-libfribidi --enable-librubberband --enable-libzmq --enable-opengl --enable-libsrt --enable-libaom --enable-schannel --extra-cflags=-DLIBTWOLAME_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-cflags=-DLIBXML_STATIC --enable-version3 --enable-nonfree --disable-stripping libavutil 56. 30.100 / 56. 30.100 libavcodec 58. 53.101 / 58. 53.101 libavformat 58. 28.101 / 58. 28.101 libavdevice 58. 7.100 / 58. 7.100 libavfilter 7. 56.101 / 7. 56.101 libswscale 5. 4.101 / 5. 4.101 libswresample 3. 4.100 / 3. 4.100 libpostproc 55. 4.100 / 55. 4.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\char.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf58.28.101 Duration: 00:04:07.53, start: 0.000000, bitrate: 9085 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 8953 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 127 kb/s (default) Metadata: handler_name : SoundHandler [STREAM] index=0 codec_name=h264 codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 profile=Constrained Baseline codec_type=video codec_time_base=1/50 codec_tag_string=avc1 codec_tag=0x31637661 width=1920 height=1080 coded_width=1920 coded_height=1088 has_b_frames=0 sample_aspect_ratio=N/A display_aspect_ratio=N/A pix_fmt=yuv420p level=40 color_range=unknown color_space=unknown color_transfer=unknown color_primaries=unknown chroma_location=left field_order=unknown timecode=N/A refs=1 is_avc=true nal_length_size=4 id=N/A r_frame_rate=25/1 avg_frame_rate=25/1 time_base=1/12800 start_pts=0 start_time=0.000000 duration_ts=3168256 duration=247.520000 bit_rate=8953725 max_bit_rate=N/A bits_per_raw_sample=8 nb_frames=6188 nb_read_frames=N/A nb_read_packets=N/A DISPOSITION:default=1 DISPOSITION:dub=0 DISPOSITION:original=0 DISPOSITION:comment=0 DISPOSITION:lyrics=0 DISPOSITION:karaoke=0 DISPOSITION:forced=0 DISPOSITION:hearing_impaired=0 DISPOSITION:visual_impaired=0 DISPOSITION:clean_effects=0 DISPOSITION:attached_pic=0 DISPOSITION:timed_thumbnails=0 TAG:language=und TAG:handler_name=VideoHandler [/STREAM]

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