April 2018 - ffmpeg-user - lists.ffmpeg.org

Re: [FFmpeg-user] Compiling ffmpeg with an output device (Blackmagick Decklink)
by Gonzalo Garramuño 18 Apr '18

18 Apr '18

El 18/04/2018 a las 14:40, Carl Eugen Hoyos escribió: >> *ffbuild/config.mak:6: *** missing separator. Stop.* >> >> I attempted tracking down the issue and found this ticket on the ffmpeg >> tracker: https://trac.ffmpeg.org/ticket/1209. I'm not using the git pull, >> instead I'm just unpacking the source tar. > > I can only guess that "upacking" involves selecting line breaks. > > [...] > As Carl guesses, you probably have an issue with line breaks ( unix vs. windows ) and that's why the config.mak file fails. -- Gonzalo Garramuño

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Need Help!! WEBVTT & DVBSUB encoding options/metadata
by Eleftherios Antoniades 18 Apr '18

18 Apr '18

Carl hi, Sorry for the delayed answer, but i was out of office lately.. I have my comments under yours! pls check, cause i did not have any better outcome..Any idea would be great..! >* Thank you for your reply! Actually i made this question several times last *>* month (subject: NEED HELP PLEASE!! Soft subtitles DVB subtitles stream, *>* 1080p size problem on direct convert a MPEG-TS video), before someone *>* answered me, and i really appreciate this!! * Shouting is usually a good idea if you want to make sure that nobody answers. (Ha, it was my 4th attempt..) >* I have a MPEG-TS video file with one video and one audio stream and i am *>* trying to create a direct video convert with soft subs, by adding a *>* DVBSUB stream inside *>* it. My original subs filewas a stl file with greek subs. *>>* After many workarounds i finally managed to add the DVBSUB stream, after *>* converting the stl file into a sub file with idx. I used the idx, so i get *>* the palette information correct. * Just curious: How do create the idx files? (I used 'Subtitle Edit v.3.5.6' & 'AVIAddXSubs" to convert the stl file to sup. That created an .idx also, that i used for the color palette of the dvbsub encoding) >* So the command i used is like this (see bellow for complete console output): *>>* ffmpeg -i testvideo.ts -i subtitles.idx -map 0:v -map 0:a -map 1 -c:v copy *>* -c:a copy -c:s:0 dvbsub -metadata:s:s:0 language=ell -streamid 2:151 *>* output.ts *>>* The video and subtitles size is 1920x1080, but the output.ts file comes out *>* with wrong subtitles size. * Yes, this is ticket #7117. (So, in that ticket the input file is an mp4 video. Not a TS video..I just try to add the DVB stream into the TS with a specific streamid.) >* It looks like the dvbsub codec converts the *>* dvbsubs stream in 720x576 size by default, rather 1920x1080?? * No, the subtitles use the input resolution of hd1080 but this size is not stored in the subtitle stream and the default size is pal (that gets stretched to match the video size). (So the default size of the dvb stream is 720x576? Can this be forced changed somehow?) >* Is there a way to force the subs into the correct size? * I suspect it should work if you produce smaller subtitles (pal). (When i reduce the size of the subtitles the size is correct but the resolution is wrong. They are not HD subs, they are SD abd you can see that the subs outline is not 'fine'...) Carl is there a way to add subtitles as teletext on a TS input video, to create a TS output video with Teletext and page number? Thank you in advance!! Lefteris

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Compiling ffmpeg with an output device (Blackmagick Decklink)
by David Di Biase 18 Apr '18

18 Apr '18

I've gone through the ffmpeg compile tutorial and have managed to get it working as expected. The build looks stable and usable. However now I need to include the Blackmagick Decklink SDK in the build using the --enable-decklink flag. *cd ~/ffmpeg_sources && \* *wget -O ffmpeg-snapshot.tar.bz2 https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2 <https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2> && \* *tar xjvf ffmpeg-snapshot.tar.bz2 && \* *cd ffmpeg && \* *PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \* * --prefix="$HOME/ffmpeg_build" \* * --pkg-config-flags="--static" \* * --extra-cflags="-I$HOME/ffmpeg_build/include" \* * --extra-ldflags="-L$HOME/ffmpeg_build/lib" \* * --extra-libs="-lpthread -lm" \* * --bindir="$HOME/bin" \* * --enable-gpl \* * --enable-libass \* * --enable-libfdk-aac \* * --enable-libfreetype \* * --enable-libmp3lame \* * --enable-libopus \* * --enable-libtheora \* * --enable-libvorbis \* * --enable-libvpx \* * --enable-libx264 \* * --enable-libx265 \* * --enable-decklink \* * --enable-nonfree && \* *PATH="$HOME/bin:$PATH" make && \* *make install && \* *hash -r* However after a few seconds it fails and reports that it cannot find the decklink libs: *ERROR: DeckLinkAPI.h not found* If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "ffbuild/config.log" produced by configure as this will help solve the problem. I've placed the libraries in the same folder specified by extra-cflags and in the ldflags. So basically the Decklink SDK is available here and here: *~/ffmpeg_build/include/decklink* *~/ffmpeg_build/lib/decklink* That's not apparently how you do it though (new to ffmpeg). I'm wondering where I should be putting the SDK source to ensure the compiler can read it. Thanks. -- This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

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Stereo AAC
by antony baxter 18 Apr '18

18 Apr '18

Hi all, I have some video files with a DTS audio stream. I'm trying to add a couple of additional audio streams into the file, transcoded from the DTS stream. My command line is: ffmpeg -i "${input}" -map 0:0 -map 0:1 -map 0:1 -map 0:1 -c:v copy -movflags +faststart -c:a:0 aac -b:a:0 192k -ac 2 -metadata:s:a:0 title="AAC Stereo Audio" -c:a:1 ac3 -ac 6 -metadata:s:a:1 title="AC3 5.1 Audio" -c:a:2 copy -disposition:a -default -disposition:a:0 default -f mp4 "${output}.mp4" However, this isn't doing exactly what I expected. The output is: Output #0, mp4, to 'output.mp4': Metadata: Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 1920x1036 [SAR 1:1 DAR 480:259], q=2-31, 23.98 fps, 23.98 tbr, 16k tbn, 1k tbc (default) Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 192 kb/s (default) Metadata: title : AAC Stereo Audio Stream #0:2(eng): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1, fltp, 448 kb/s Metadata: title : AC3 5.1 Audio Stream #0:3(eng): Audio: dts (DTS) (mp4a / 0x6134706D), 48000 Hz, 5.1(side), fltp, 1536 kb/s Note that Stream #0:1 is being output as AAC 5.1 instead of AAC stereo. How come the -ac 2 flag is being ignored - do I need to specify which stream this applies to, and if so how? Thank you all Ant.

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[ffmpeg-user] Auto generate Webvtt
by Tri Quang Nguyen 18 Apr '18

18 Apr '18

Dear Friends, Currently, I am looking for command to create Webvtt for my video by using FFFMPEG. The requirement is creating a webvtt file with preview-thumbnail for using as THEOPlayer (https://support.theoplayer.com/hc/en-us/articles/207460505-Preview-Thumbnai…<https://support.theoplayer.com/hc/en-us/articles/207460505-Preview-Thumbnai…>) Example: WEBVTT 00:00.000 --> 00:10.000 /thumbnails/thumbnail1.jpg 00:10.001 --> 00:20.000 /thumbnails/thumbnail2.jpg 00:20.001 --> 00:30.000 /thumbnails/thumbnail3.jpg What I did so far: 1. I get duration for video (Cause: requirement required 100 sections for each video) #Get duration of mp4 file need to generate thumbnail duration="$(ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 $video)" 1. Then, I creating thumbnail with duration above: #Create folder to store thumbnail if not exist. mkdir -p $BASEDIR/$filename #Create thumbnail with size 200:100 ffmpeg -i $video -filter:v scale=200:100:force_original_aspect_ratio=increase,crop=200:100,fps=100/$duration $BASEDIR/$filename/thumbnail%04d.png –benchmark Now, I was successful to create thumbnail as I wish, but for creating Webvtt, I still looking the document and googling with no sign. Please someone help to point me the correct way or suggestion. Thanks and Best Regards, Tri Nguyen This e-mail is intended solely for the person or entity to which it is addressed and may contain confidential and/or privileged information. Any review, dissemination, copying, printing or other use of this e-mail by persons or entities other than the addressee is prohibited. If you have received this e-mail in error, please contact the sender immediately and delete the material from any computer. (HCVN0116)

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Problem with a onvif camera
by Denis 17 Apr '18

17 Apr '18

Hi. With this command line I can capture the video stream from my personal onvif camera using my linux machine and the official already built ffmpeg executable from the ffmpeg website. *./ffmpeg -i rtsp://192.168.1.162:554/onvif1 a.avi* But I can't capture the video stream from another onvif camera, the command line is *./ffmpeg -i rtsp://admin:admin@172.30.153.33:554/1 a.avi* The ffmpeg output is denis@tomcat:~/Scaricati/ffmpeg-3.4.2-64bit-static$ ./ffmpeg -i rtsp:// admin:admin@172.30.153.33:554/1 a.avi ffmpeg version 3.4.2-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2018 the FFmpeg developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516 [rtsp @ 0x43a4bc0] UDP timeout, retrying with TCP [rtsp @ 0x43a4bc0] method PAUSE failed: 455 Method Not Valid In This State [rtsp @ 0x43a4bc0] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size *Consider increasing the value for the 'analyzeduration' and 'probesize' options* Guessed Channel Layout for Input Stream #0.1 : mono Input #0, rtsp, from 'rtsp://admin:admin@172.30.153.33:554/1': Metadata: title : h264.mp4 Duration: N/A, bitrate: 64 kb/s Stream #0:0: Video: h264, *none*, 90k tbr, 90k tbn, 180k tbc Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0:1 -> #0:0 (pcm_mulaw (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help [rtsp @ 0x43a4bc0] UDP timeout, retrying with TCP [rtsp @ 0x43a4bc0] method PAUSE failed: 455 Method Not Valid In This State rtsp://admin:admin@172.30.153.33:554/1: Operation not permitted Finishing stream 0:0 without any data written to it. Output #0, avi, to 'a.avi': Metadata: INAM : h264.mp4 ISFT : Lavf57.83.100 Stream #0:0: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 8000 Hz, mono, s16p Metadata: encoder : Lavc57.107.100 libmp3lame size= 6kB time=00:00:00.00 bitrate=N/A speed= 0x video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) denis@tomcat:~/Scaricati/ffmpeg-3.4.2-64bit-static$ I have already tried to increase the value of analyzeduration and probesize I can watch the camera using *vlc* with the same command line. Can someone suggest me how to resolve? Is ffmpeg not fully compatible with all video cameras? Regards. +39.347.4070897 www.labcsp.com[1] www.denisgottardello.it[2] GMT+1 Skype: mrdebug -------- [1] http://www.labcsp.com [2] http://www.denisgottardello.it

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Ffmpeg Nvidia GPU mkv encoding output file size
by oktay eşgül 17 Apr '18

17 Apr '18

I have been working with below environment. ==>Ubuntu 16.04.3 ==>FFmpeg :3.4.2 -- configuration: --prefix=/usr/local/ffmpeg_new/ --enable-cuda --enable-cuvid --enable-nvenc --enable-nonfree --enable-libnpp --extra-cflags=-I/usr/local/cuda/include --extra-ldflags=-L/usr/local/cuda/lib64 --enable-libvpx --enable-libvorbis --enable-libfreetype ==>Cuda 9.1.85 ==> GPU: GeForce GTX 1080ti ==> I have two media files : caller_video.webm (80MB) and callee_video.webm (120MB) ==> Then I am demuxing audios from these videos via below command: time ffmpeg -y -i caller.webm -vn -ab 256 caller.wav Caller.wav = 260 MB ==>Then, muxing caller.wav file with callee_video.webm time ffmpeg -y -i callee_video.webm -i caller.wav -filter_complex '[0:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=0.5[a1]; [1:a]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,volume=0.5[a2]; [a1][a2]amerge,pan=stereo|c0 File Size of tempfile : 87MB ==>Then, I am encoding the temp file via below command: time ffmpeg -y -i temp_file.mkv -pixel_format yuv444p -preset default -vf drawtext=fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans-Bold.ttf:x=8:y=8:fontsize=16:fontcolor=yellow@1:expansion=strftime:basetime=1518172680000000:text='%Y-%m-%d %H-%M-%S' -acodec libvorbis -c:v h264_nvenc final_video.mkv Final_video Size: 270 MB ==>The thing I am trying to understand is that when I encode the orig webm file which is 80MB,the encoded output mkv file size ~20MB . ==>Yet, when I performed the same operation on muxed temp media file(callee_video.webm + caller.wav) that the size is 87MB before encoding, becomes 270MB . Would you please share your comments what I am missing or doing wrong. Thank you

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H265 multi
by José María Infanzón 17 Apr '18

17 Apr '18

Hi all, I'm trying to get multiple outputs (different resuloutions) out of the same input. I was able to achieve this with h264 based on the guide: https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs but on the case of HEVC, I'm being unable I've tried with: /root/ffmpeg_sources/ffmpeg/ffmpeg -loglevel error -y -i udp://@ 224.2.2.3:1012 -s 640X360 -vcodec libx265 -pix_fmt yuv420p -tune zerolatency -x265-params "bitrate=700:vbv-bufsize=1400" -acodec libfdk_aac -profile:a aac_he_v2 -b:a 32K -f mpegts - -s 768X432 -vcodec libx265 -pix_fmt yuv420p -tune zerolatency -x265-params "bitrate=1100:vbv-bufsize=2200" -acodec libfdk_aac -profile:a aac_he_v2 -b:a 32K -f mpegts - Where I have 2 resolutions (640x360 & 768x432) with the same audio encoding. But it doesn't work. -- José María Infanzón

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Audio Desynchronization When Utilizing Segmentation
by Gabriel Balaich 16 Apr '18

16 Apr '18

Hey all, I was hoping I could a get a bit of help on a problem I'm having. Here is my code: ffmpeg -y -thread_queue_size 5096 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M -framerate 100 -pixel_format nv12 ^ -itsoffset 00:00:00.015 -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 -async 1 -vsync 1 ^ -max_muxing_queue_size 5096 -segment_time 300 -segment_wrap 9 -f segment C:\Users\djcim\Videos\PC\PC\PC%02d.mp4 My goal is to continuously record without filling up my hard drive keeping the last hour and 30 minutes of footage at all times. I'd like to have the ability to edit a group or even just a single segment without any dependencies. As in be able to drag one or more clips into a video editing program like adobe premiere regardless of how long I have been recording and have everything line up perfectly, including desirable audio to video synchronization.Segmentation seemed perfect for this... but when I drag the segments into a video editing program like Adobe premiere the audio slowly drifts from the video after each break. If I remove segmentation and record one continuous clip this does not seem to be a problem, video and audio stay synchronized. Is there anyway to force the beginning of each segment to pick up exactly where the last one left off? Apologies if any part of my code is redundant, and thank you for any help.

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Delay Only Video Stream Of Audio Linked dshow Input
by Gabriel Balaich 16 Apr '18

16 Apr '18

Hello everyone, I'm have a slight issue trying to sync my audio and video up with an acceptable margin of error. Here is my command: ffmpeg -y -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M ^ -framerate 100 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ^ -map 0:0,0:1 -map 0:1 -flags +cgop -force_key_frames expr:gte(t,n_forced*2) -c:v h264_nvenc -preset: llhp -pix_fmt nv12 ^ -b:v 250M -minrate 250M -maxrate 250M -bufsize 250M -c:a aac -ar 44100 -b:a 384k -ac 2 -r 100 -af "aresample=async=250" ^ -vsync 1 -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ^ C:\Users\djcim\Videos\PC\PC\PC%02d.ts My problem is the video comes in slightly ahead of the audio, I can use -itsoffset but then I have to call the video and audio as separate inputs as -itsoffset offsets both audio and video. While this may seem the obvious solution it causes inconsistent audio synchronization if the audio isn't called with the video. Basically if both audio and video aren't called at the same time the video can now be ahead or behind with a 2-3 frame margin. When I call them at the same time the video consistently comes in 2 frames ahead of the audio, every time. I just need a way to delay only the video stream without delaying the audio while keeping both audio and video linked from the beginning. I've tried this with no luck: ffmpeg -y -thread_queue_size 9999 -indexmem 9999 -guess_layout_max 0 -f dshow -video_size 3440x1440 -rtbufsize 2147.48M ^ -framerate 200 -pixel_format nv12 -i video="Video (00 Pro Capture HDMI 4K+)":audio="SPDIF/ADAT (1+2) (RME Fireface UC)" ^ -flags +cgop -force_key_frames expr:gte(t,n_forced*2) -c:v h264_nvenc -preset: llhp -pix_fmt nv12 -b:v 250M ^ -minrate 250M -maxrate 250M -bufsize 250M -c:a aac -ar 44100 -b:a 384k -ac 2 -r 100 ^ -filter_complex "[0:v] setpts=PTS-STARTPTS+.032/TB [v]; [0:a] asetpts=PTS-STARTPTS, aresample=async=250 [a]" -map [v] ^ -map [a] -vsync 1 -max_muxing_queue_size 9999 -f segment -segment_time 600 -segment_wrap 9 -reset_timestamps 1 ^ C:\Users\djcim\Videos\PC\PC\PC%02d.ts Just like -itsoffset both video and audio are being delayed. You can delay solely audio with adelay, but there doesn't seem to be a video delaying equivalent. Any help or advice would be greatly appreciated.

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