September 2019 - ffmpeg-user - lists.ffmpeg.org

[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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Question regarding ffmpeg source code
by Cihan Topal 10 Sep '19

10 Sep '19

Hi Everybody, I am working on some project involving video streaming using ffmpeg library source codes. Although I searched the web, I could not find a proper answer for lossless encoding with H264. I have two questions: 1- Constant Rate Factor (crf), preset and map are some of the parameters available in the ffmpeg process. However, they are not to be found in the source code available? 2- How can we set these parameters in the source code? Or do you have other suggestion in how to use source code for lossless streaming of multiple videos using MPEG-TS? I would appreciate if you can kindly help me on these. Best Regards, Cihan Topal

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STOP RTSP STREAM IF CONNECTION FAILS
by Alejandro Escudero 10 Sep '19

10 Sep '19

Hi! I am trying to Stop an RTSP STREAMING if connection to IP Camera fails (example if camera goes offline) I tried these command with -timeout 3 (to exit after 3 seconds if connection fails, but it never exists), how this should be done correctly? ffmpeg -rtsp_transport tcp -i "rtsp://admin:pass@myipcameras:554/cam/realmonitor?channel=1&subtype=0"-timeout 3 -c copy TEST.mp4 thanks! Alejandro [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Libre de virus. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

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Video output distortion on ffpmeg live stream
by Ben Houghton 10 Sep '19

10 Sep '19

I am helping build a node server that runs ffmpeg to stream a live WebRTC input to HLS and MPEGDASH. Often when the encoding starts it produces a corrupted video output that looks something like this (viewed using VLC): HLS output distorted image <https://i.stack.imgur.com/Dxx03.png> MPEGDASH output distorted image <https://i.stack.imgur.com/9YQri.png> It isn't doing this every time, but roughly 50%. The ffmpeg commands being run are: ffmpeg -re -y -loglevel error -hide_banner -threads 1 -use_wallclock_as_timestamps 1 -f rawvideo -pix_fmt yuv420p -s 640x480 -framerate 30.000030517578125 -thread_queue_size 512 -i tcp://127.0.0.1:43713 -thread_queue_size 512 -i tcp://127.0.0.1:41871 -vsync 1 -r 30.000030517578125 -c:a aac -c:v h264 -profile:v high -b_strategy 0 -bf 1 -f dash -use_template 1 -use_timeline 1 -window_size 5 -live 1 -min_seg_duration 2000000 -preset veryfast -crf 22 -g 48 -keyint_min 48 -sc_threshold 0 -filter_complex [0]format=pix_fmts=yuv420p[temp0];[temp0]scale=-2:240[A0] -map [A0]:v -b:v:0 400k -maxrate 428k -bufsize 600k -map 1:a -b:a:0 64k -filter_complex [0]format=pix_fmts=yuv420p[temp1];[temp1]scale=-2:360[A1] -map [A1]:v -b:v:1 700k -maxrate 749k -bufsize 1050k -map 1:a -b:a:1 96k /usr/src/app/cdn-path/webrtc_test.mpd ffmpeg -re -y -loglevel error -hide_banner -threads 1 -use_wallclock_as_timestamps 1 -f rawvideo -pix_fmt yuv420p -s 640x480 -framerate 30.000030517578125 -thread_queue_size 512 -i tcp://127.0.0.1:43713 -thread_queue_size 512 -i tcp://127.0.0.1:41871 -vsync 1 -r 30.000030517578125 -c:a copy -c:v h264 -profile:v high -b_strategy 0 -bf 1 -f mpegts /usr/src/app/cdn-stream-save/webrtc_test_1567437442625.ts -c:a aac -c:v h264 -profile:v high -b_strategy 0 -bf 1 -f hls -hls_time 2 -hls_list_size 5 -hls_flags delete_segments -preset veryfast -crf 22 -g 48 -keyint_min 48 -sc_threshold 0 -filter_complex [0]format=pix_fmts=yuv420p[temp0];[temp0]scale=-2:240[A0] -map [A0]:v -b:v:0 400k -maxrate 428k -bufsize 600k -map 1:a -b:a 64k /usr/src/app/cdn-path/webrtc_test_1.m3u8 -hls_time 2 -hls_list_size 5 -hls_flags delete_segments -preset veryfast -crf 22 -g 48 -keyint_min 48 -sc_threshold 0 -filter_complex [0]format=pix_fmts=yuv420p[temp1];[temp1]scale=-2:360[A1] -map [A1]:v -b:v:1 700k -maxrate 749k -bufsize 1050k -map 1:a -b:a 96k /usr/src/app/cdn-path/webrtc_test_2.m3u8 ffmpeg details: ffmpeg version 3.4.6 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.11) 20160609 configuration: --disable-debug --disable-doc --disable-ffplay --enable-shared --enable-avresample --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-gpl --enable-libass --enable-libfreetype --enable-libvidstab --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxvid --enable-libx264 --enable-nonfree --enable-openssl --enable-libfdk_aac --enable-libkvazaar --enable-postproc --enable-small --enable-version3 --extra-cflags=-I/opt/ffmpeg/include --extra-ldflags=-L/opt/ffmpeg/lib --extra-libs=-ldl --prefix=/opt/ffmpeg libavutil 55. 78.100 / 55. 78.100 libavcodec 57.107.100 / 57.107.100 libavformat 57. 83.100 / 57. 83.100 libavdevice 57. 10.100 / 57. 10.100 libavfilter 6.107.100 / 6.107.100 libavresample 3. 7. 0 / 3. 7. 0 libswscale 4. 8.100 / 4. 8.100 libswresample 2. 9.100 / 2. 9.100 libpostproc 54. 7.100 / 54. 7.100 Does anyone have any ideas on what might cause this type of distortion on the output stream or how I can test further to track it down? Thank you

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yuv420p vs yuv420p(tv)
by George R. Welch 09 Sep '19

09 Sep '19

Hi folks, I have two versions of the same video and I would like to only keep one. If one is higher quality, I'd obviously want to keep that one. I can't tell tell any difference through casual watching with VLC, so perhaps it does not matter. But anyway, when I run ffmpeg -i on the two, the difference is: < Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080, 4011 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc (default) --- > Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv), 1920x1080 [SAR 1:1 DAR 16:9], 4173 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc (default) I think I understand the difference between yuv420p(pc) and yuv420p(tv), but here I have yuv420p vs yuv420p(tv). Can someone offer suggestions as to which of these two videos might be the better encoding? Thanks, --George

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Color shift when converting static BMP/JPG image (but not PNG) to video
by Benjamin Peng 09 Sep '19

09 Sep '19

I've posted this to trac but didn't get much traction (yet), and I found this problem to be very simple but weird to happen at the same time, so I'd like to see if anyone here have some insight. Please read the full description here: https://trac.ffmpeg.org/ticket/8056 Simply put, if you convert a BMP image to video (tried x264 and HEVC), the color will shifted to be slightly green-yellow-ish. Pure white (255,255,255) will become something like 252,255,252. (In the ticket, I added some extra parameters to make sure the video correctly use BT.709 conversion since it's HD, but it shouldn't matter for white color) The most weird part of this bug is, it doesn't happen if the image is in PNG format, which should be identical to the BMP counterpart considering it's a lossless format. The same bug DOES happen if you use JPG images, though.

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FFV1 and CELLAR
by Noeck 08 Sep '19

08 Sep '19

Hi, I have a question about the current FFV1 codec (encoder): Is this implementation compliant to the current state of CELLAR¹? Is the CELLAR spec just documenting what ffmpeg does already? Or does the spec evolve and the implementation will be adapted later? In other words, if I encode a video with current ffmpeg, will the result be compliant to the IETF spec? If not, when will that happen? Best, Daniel ¹: https://tools.ietf.org/wg/cellar/

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FFMPEG producing multiple outputs (one streaming and one snapshot)
by Thomas Chen 08 Sep '19

08 Sep '19

i am trying to capture video from a camera and produce to two outputs 1) streaming (h264) 2) taking periodic snapshot of the image (jpg) is this possible ? i read https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs that ffmpeg can have multiple outputs but i am not sure how to handle when one output is just snapshot of image (and triggered by an external signal or a periodic timer ?) btw, the camera module i am using can support MPEG (/dev/video0) and H264 (/dev/video1), if i am streaming with H264 (single output), then the MPEG (video0) is not operable. if it is not possible, please advise what other software that can do this ? gstreamer ? thanks

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Streaming phone call live via wav file as hls stream
by Samuel Widmer 07 Sep '19

07 Sep '19

Dear >From a live Phone Call we do append audio data to a Windows wave file. The goal: We would like to stream that wave file as hls stream. Our system is able to keep the wave file open while the whole phone call. But also opening for append and closing the file would be possible. We know how to create a hls stream with m3u9 playlist from a wave file. For this we use for ex. The following cmd line: ffmpeg -i Music.wav -c:a aac -hls_time 2 -hls_list_size 0 -hls_segment_filename file%03d.aac out.m3u8 But now how about when the wave file is still appended ? We would like to build live each aac ts and ongoing updating the m3u8 playlist. This as long as the wave file is opened and appended by our system. Is there any command FFmpeg command existing for this usage ? Is there any command FFmpeg command existing for this usage when using pipe instead of file ? Thanks for any help! Best Reagard Samuel Widmer Blue Call AG <http://www.bluecall.ch/> www.bluecall.ch, <mailto:Samuel.Widmer@bluecall.ch> Samuel.Widmer(a)bluecall.ch

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Visualization filter colors and scale.
by Zsolt 07 Sep '19

07 Sep '19

Hello! Does anyone know how do I set the color matrix and range when using filter_complex? I use this filter: -filter_complex "[0:a]showfreqs=s=1920x1080:cmode=combined:colors=#990000|#770000,format=yuv444p[vid]" picture: [image: 1.jpg] The colors look rather orange, even though I only set the red channel. Normally I use "-vf scale=out_color_matrix=bt709:out_range=pc" when dealing with full range to get proper video, but of course I can't use vf with filter_complex. How do I do it in this case? *A different question:* I also tried the showspectrum filter. When I use fscale=log the result is quite horrible. as if it's the same linear scale only the lower frequencies obscenely stretched, can this be helped? snapshot: [image: 2.jpg]

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