February 2021 - ffmpeg-user - lists.ffmpeg.org

Issue when adding a title to an audio stream within a mp4 file
by PPRJ01 02 Mar '21

02 Mar '21

Hello All, I'm not a developper. Therefore I'm not able to analyze the ffmpeg/ffprobe source code. I observed that the command below seem to work properly when adding a title to an audio stream within an MP4 file : ffmpeg -i INPUT.mp4 -c copy -metadata:s:a:0 "title=mono" OUTPUT.mp4 Both VLC and Exiftool do display correctly the audio stream title of the OUTPUT.mp4 file when needed. The point is that neither ffmpeg nor ffprobe can do it. For example, the command : ffmpeg -i OUTPUT.mp4 (with no output file) do not show the audio title. I don't know where ffmpeg stores the title metadata but VLC and Exiftool are able to retrieve it. Maybe this is a designed behavior of ffmpeg/ffprobe. Or a bug ? Please let me know.

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SSIM filter showing (small) differences in identical(?) streams
by Ian Pilcher 02 Mar '21

02 Mar '21

I am trying to understand how the SSIM and VMAF filters work, with an eye to finding the "best" compression settings for a video which will be composed from a series of TIFF images. Unfortunately, I'm stuck at the beginning, as I can't get the SSIM filter to behave as expected. ./source contains the original sequence of images. $ tiffinfo source/000000.tif TIFF Directory at offset 0x473108 (4665608) Image Width: 1440 Image Length: 1080 Bits/Sample: 8 Sample Format: unsigned integer Compression Scheme: None Photometric Interpretation: RGB color Samples/Pixel: 3 Rows/Strip: 1 Planar Configuration: single image plane I attempt to create a lossless video of the first minute. $ ffmpeg -start_number 0 -framerate 60000/1001 -i source/%06d.tif -t 00:01:00 -c:v huffyuv lossless.mkv The result appears reasonable. $ mediainfo lossless.mkv General Unique ID : 235140899628261703308032414639716345340 (0xB0E67D6EF6B78362D9BCF9EA3080A5FC) Complete name : lossless.mkv Format : Matroska Format version : Version 4 File size : 8.54 GiB Duration : 59 s 994 ms Overall bit rate : 1 223 Mb/s Writing application : Lavf58.45.100 Writing library : Lavf58.45.100 ErrorDetectionType : Per level 1 Video ID : 1 Format : HuffYUV Format version : Version 2 Codec ID : V_MS/VFW/FOURCC / HFYU Duration : 59 s 994 ms Bit rate : 1 199 Mb/s Width : 1 440 pixels Height : 1 080 pixels Display aspect ratio : 4:3 Frame rate mode : Constant Frame rate : 59.940 FPS Color space : RGB Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 12.860 Stream size : 8.37 GiB (98%) Writing library : Lavc58.91.100 huffyuv Default : Yes Forced : No Now let's see what the SSIM filter says. $ ffmpeg -i lossless.mkv -start_number 0 -framerate 60000/1001 -i source/%06d.tif -t 00:01:00 -filter_complex ssim -f null - ... Input #0, matroska,webm, from 'lossless.mkv': Metadata: ENCODER : Lavf58.45.100 Duration: 00:00:59.99, start: 0.000000, bitrate: 1223104 kb/s Stream #0:0: Video: huffyuv (HFYU / 0x55594648), bgr0, 1440x1080, 59.94 fps, 59.94 tbr, 1k tbn, 1k tbc (default) Metadata: ENCODER : Lavc58.91.100 huffyuv DURATION : 00:00:59.994000000 Input #1, image2, from 'source/%06d.tif': Duration: 01:47:16.00, start: 0.000000, bitrate: N/A Stream #1:0: Video: tiff, rgb24, 1440x1080, 59.94 tbr, 59.94 tbn, 59.94 tbc ... [Parsed_ssim_0 @ 0x55fb09c738c0] not matching timebases found between first input: 1/1000 and second input 1001/60000, results may be incorrect! ... [Parsed_ssim_0 @ 0x55fb09c738c0] SSIM R:0.833774 (7.793009) G:0.835401 (7.835723) B:0.831058 (7.722615) All:0.833411 (7.783532) That's not what I expected. My understanding is that the R, G, B, and All values should all be "1.000000 (inf)". The "not matching timebases" warning is the obvious thing to look at. After much searching, I came upon the -video_track_timescale option, but it seems to only take an integer, and 60 is not the same as 59.94, so it seems that I simply can't directly compare a video stream with a non- integer framerate to an image sequence. As a workaround, I tried extracting the lossless video frames as a separate image sequence. $ ffmpeg -i lossless.mkv -start_number 0 lossless/%06d.png This created the expected sequence of image files (000000.png - 003595.png). Since I have both the "source" and the "lossless" streams as images sequences, I can use ImageMagick to compare them. $ for I in `seq -w 0 3595` ; do compare -metric AE source/00${I}.tif lossless/00${I}.png /tmp/diff.png 2>/dev/null || echo $I ; done This produces no output, indicating that ImageMagick thinks that the TIFF files in ./source and the PNG files in ./lossless contain completely identical image data. What does the SSIM filter say? $ ffmpeg -framerate 60000/1001 -start_number 0 -i lossless/%06d.png -framerate 60000/1001 -start_number 0 -i source/%06d.tif -t 00:01:00 -filter_complex ssim -f null - ... Input #0, image2, from 'lossless/%06d.png': Duration: 00:00:59.99, start: 0.000000, bitrate: N/A Stream #0:0: Video: png, rgb24(pc), 1440x1080, 59.94 fps, 59.94 tbr, 59.94 tbn, 59.94 tbc Input #1, image2, from 'source/%06d.tif': Duration: 01:47:16.00, start: 0.000000, bitrate: N/A Stream #1:0: Video: tiff, rgb24, 1440x1080, 59.94 tbr, 59.94 tbn, 59.94 tbc ... [Parsed_ssim_0 @ 0x556c7e948980] SSIM R:0.999775 (36.484195) G:0.999777 (36.508857) B:0.999774 (36.451744) All:0.999775 (36.481536) Close, but not the "1.000000 (inf)" that I would expect for identical streams. I suppose that it may be related to the "rgb24(pc)" vs. rgb24 pixel formats; it's hard to be sure, as Google isn't turning up anything that describes what "rgb24(pc)" actually means. Does anyone know what the heck is going on here? Am I doing something wrong? -- ======================================================================== In Soviet Russia, Google searches you! ========================================================================

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Re: [FFmpeg-user] BER TOO HIGH
by KRISHNAKUMAR N K 01 Mar '21

01 Mar '21

On Wed, 24 Feb 2021 at 17:40, Christian David <christian(a)cdavid.eti.br> wrote: > Generally this error means something on reception, in parallel at ffmpeg > running you have a vlc or something like that playing the stream? The > stream plays well ? Otherwise you can try Make a Linux service with > autorestart and add the argument "-abort_on empty_output" at ffmpegEnviado > do meu Galaxy > Yes, the stream auto recover's in VLC but not in ffmpeg. As suggested, i will try to use " -abort_on empty_output" Thanks Krishnakumar > -------- Mensagem original --------De : KRISHNAKUMAR N K < > nk.krishnakumar(a)gmail.com> Data: 24/02/2021 08:57 (GMT-03:00) Para: > FFmpeg user questions <ffmpeg-user(a)ffmpeg.org> Assunto: [FFmpeg-user] BER > TOO HIGH Hi Guys,I am trying to convert a UDP input to a RTMP live stream, > whenever thereare any alarm's like (*BER Too High on DVB-S/S2 In Port 3*) > on the IRD end,ffmpeg hangs and stop's feed push, it stops printing the log > lines as well.Not sure how to handle this issue. At present, I am stopping > and startingthe conversion for recovery. If someone can help me the > context of theissue, that would be great*Complete Logs :* > https://pastebin.com/qUagbwK4*Sample Logs: *[mpeg2video @ 0x4efdd40] > MT_DMV in progressive_sequence[mpeg2video @ 0x4efdd40] 00 motion_type at 0 > 274[mpeg2video @ 0x4efdd40] 00 motion_type at 0 403[mpeg2video @ 0x4efdd40] > 00 motion_type at 3 287[mpeg2video @ 0x4efdd40] 00 motion_type at 0 > 276[mpeg2video @ 0x4efdd40] 00 motion_type at 0 277[mpeg2video @ 0x4efdd40] > 00 motion_type at 0 278[mpeg2video @ 0x4efdd40] 00 motion_type at 0 > 279[mpeg2video @ 0x4efdd40] 00 motion_type at 0 408[mpeg2video @ 0x4efdd40] > mb incr damaged[mpeg2video @ 0x4efdd40] 00 motion_type at 0 289[mpeg2video > @ 0x4efdd40] MT_DMV in progressive_sequence[mpeg2video @ 0x4efdd40] 00 > motion_type at 3 281[mpeg2video @ 0x4efdd40] 00 motion_type at 0 > 282[mpeg2video @ 0x4efdd40] MT_DMV in progressive_sequence[mpeg2video @ > 0x4efdd40] Warning MVs not available*ffmpeg Command :*ffmpeg -y -v error > -progress tcp://127.0.0.1:39953 -i udp:// > 230.1.1.4:10000?fifo_size=10000000&overrun_nonfatal=1 > -filter_complex[i:0x1171]yadif[vout];[vout]split=1[out0];[out0]setdar=1280/720[v0];[i:0x1239]aresample=async=1:min_hard_comp=0.100000:first_pts=0[aout];[aout]asplit=1[a0]-f > flv -vcodec h264_nvenc -s 1280x720 -r 25/1 -b:v 1500000 -acodeclibfdk_aac > -ar 48000 -b:a 128000 -gpu 1 -sc_threshold 0 -pix_fmt yuv420p-flags > +global_header+cgop -profile:v high -level 3.2 -preset fast -rc:vcbr > -maxrate 1500000 -minrate 1500000 -bufsize 3000000 -g 50 -map [v0] -map[a0] > rtmp://ipaddress/feed_pathThanks*KrishnaKumar **N > K*_______________________________________________ffmpeg-user mailing > listffmpeg-user@ffmpeg.orghttps:// > ffmpeg.org/mailman/listinfo/ffmpeg-userTo unsubscribe, visit link above, > or emailffmpeg-user-request(a)ffmpeg.org with subject "unsubscribe". > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user(a)ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-request(a)ffmpeg.org with subject "unsubscribe".

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Unable to sync audio and video
by Livio Tenze 28 Feb '21

28 Feb '21

Dear group, I am trying to write a code to encode (in realtime) a stream with audio and video in Linux environment. -The video source is read with av_read_frame and the /dev/videoX is opened. The pts values with respect to the first packed seem to be reasonable. -The audio source comes from alsa: here I see really strange behaviour. The pts value of the second packet with respect to the first pts packet seems to be "delayed" from 0.4s to 1.8s. I don't understand this behaviour: I checked the time from start to the second packet and the elapsed time cannot be justified (ms and not s). -Finally, when I write audio and video in the output MP4 stream (I use libx264 and libfdk_aac for video and audio respectively) the audio stream is delayed. I don't know what to check. Please give me suggestions to solve this issue. What can I check? Thanks in advance and best regards. Livius

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Ffmpeg electron build
by Martin Barker 28 Feb '21

28 Feb '21

Hello, I am trying to make an electron app that uses ffmpeg, and I am trying to release this electron program on the mac apple store, which means the app needs to be sandboxed. I've been able to create a static ffmpeg build, but my ffmpeg build stops working once I build it with my electron mac-apple-store (mas) build, will throw this error: "illegal instruction 4" I know this is a pretty niche project, but I know people have successfully packaged ffmpeg into an electron app on the mac-apple-store before, like lossless-cut (https://github.com/mifi/lossless-cut) I documented all my steps in much greater detail here: https://stackoverflow.com/questions/66286440/electron-ffmpeg-mas-build-ille… Any help would be greatly appreciated, thank you. Martin

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Cutting - Video Missing
by Denis Połeć 28 Feb '21

28 Feb '21

Hello, I have question about simple cutting/slice a video. I am working with a bash script, which should simply randomly slice one video in a loop function: ################################################################################################### function portion(){ durationInput=$(ffprobe -v error -hide_banner -i $1 -show_format | grep duration | cut -d= -f2);\ r=$RANDOM;\ ran=$(printf '%.2f' $(echo "scale=2; $r/32768" | bc ));\ start=$(printf '%.2f' $(echo "scale=2; $durationInput*($ran*0.9)" | bc ));\ duration=$(printf '%.2f' $(echo "scale=2; ($ran*4)+2" | bc ));\ filename=${1%%.*}_portion_$c.mp4 ffmpeg -i "$1" -ss $start -t $duration -c:v copy -c:a copy -y $filename } ################################################################################################### in=GoogleMaps_.mp4 for ((c=1; c<=10; c++)); do portion $in done When I am just copying the video codec sometimes the video is missing and I just get a file with sound. If I decode like with -c:v libx264 it works probably what could be the reason? Thanks. :) denis

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How can I force a 360kHz time base?
by Mark Filipak (ffmpeg) 27 Feb '21

27 Feb '21

How can I force ffmpeg to use a 360kHz time base? (I spent 1/2 day searching.) Thanks! Mark. With a 1kHz time base: 65535 = largest 16-bit integer. 0:01:05.535 = longest running time for 16-bit integer with 1kHz time base.. Conclusion: FFmpeg cannot be using 16-bit integers. 16777215 = largest 24-bit integer. 4:39:37.215 = longest running time for 24-bit integer with 1kHz time base. Conclusion: FFmpeg could use 24-bit integers if time base is capped at 1kHz. With a 90kHz time base: 16777215 = largest 24-bit integer. 00:03:06.4135 = longest running time for 24-bit integer with 90kHz time base. Conclusion: FFmpeg cannot be using 24-bit integers. 4294967295 = largest 32-bit integer. 13:15:21.8588[3..] = longest running time for 32-bit integer with 90kHz time base. 24/1.001fps ==> deltaPTS = 3753.75. 24fps ==> deltaPTS = 3750. 25fps ==> deltaPTS = 3600. 30/1.001fps ==> deltaPTS = 3003. 30fps ==> deltaPTS = 3000. 50fps ==> deltaPTS = 1875. 60/1.001fps ==> deltaPTS = 1501.5. 60fps ==> deltaPTS = 1500. Conclusion: Since 24/1.001fps requires rounding, a 90kHz time base is not optimal. With a 360kHz time base: 4294967295 = largest 32-bit integer. 3:18:50.464708[3..] = longest running time for 32-bit integer with 360kHz time base. 24/1.001fps ==> deltaPTS = 15015. 24fps ==> deltaPTS = 15000. 25fps ==> deltaPTS = 14400. 30/1.001fps ==> deltaPTS = 12012. 30fps ==> deltaPTS = 12000. 50fps ==> deltaPTS = 7200. 60/1.001fps ==> deltaPTS = 6006. 60fps ==> deltaPTS = 6000. 100fps ==> deltaPTS = 3600. 120/1.001fps ==> deltaPTS = 3003. 120fps ==> deltaPTS = 3000. Conclusion: A 360kHz time base is optimal, even for 100fps & 120fps.

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aax to m4b with chapters
by Max Blindenhoefer 26 Feb '21

26 Feb '21

Hi, is it possible to copy chapters with ffmpeg, when you encode an aax file to m4b or convert to mp3 format? I tried "-map_chapters" but couldn’t find out which parameters work with that command. Greetings Max Gesendet von Mail<https://go.microsoft.com/fwlink/?LinkId=550986> für Windows 10

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Trouble capturing speaker audio with video on Ubuntu 20.04
by Bo Berglund 25 Feb '21

25 Feb '21

I am transfering a streaming video recorder program from Windows to Ubuntu Linux and I am having trouble capturing video and audio together. When I run ffmpeg commands using the following arguments I get only the audio coming in to the microphone, which includes the ambient sound plus the output from the laptop speaker as listened to by the laptop microphone. Really bad quality... -f pulse -i default -f alsa -i hw:0 When I use this I get no audio at all: -f pulse -i alsa_output.pci.....analog-stereo-monitor the i parameter is really long and is retrieved using the command: pacmd list-sources On Windows I use this full capture command and it works fine: ffmpeg -hide_banner -f gdigrab -framerate 25 -offset_x 1042 -offset_y 228 -video_size 854x480 -i desktop -f dshow -i audio="virtual-audio-capturer" -vcodec libx264 -pix_fmt yuv420p -t 3600 test.mp4 But this does not work on Ubuntu of course... I can supply more details if you tell me what is needed. I am writing this on Windows and the Linux computer is a bit difficult to reach for copying command results etc.... But it can be done if I am given instructions on what is needed to find out the proper command. -- Bo Berglund Developer in Sweden

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HTML source
by adam smith 25 Feb '21

25 Feb '21

hi gang hope you are all good today. I would like to record a single html page for a fixed duration as if it was a video source. I know I could use OBS to show the web page as an input on a scene and then record it, but I wondered if I could do it directly from FFmpeg? Does anybody know if this is possible? Thanks for your advice as always. Adam

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