May 2017 - ffmpeg-user - lists.ffmpeg.org

ffmpeg version
by negin tebyani 07 May '17

07 May '17

I have two systems that I am running ffmpeg on them. when I use the ffmpeg -version command, the versions are so different. this is the first system's ffmpeg version: ffmpeg version N-85203-g400378b and this is for the second system: ffmpeg version 2.8.11 I tried to install 2.8.11 version on the first one. but still no result. how can I fix this? i need to use x264opts. it works on second system. but the first system does not include libx264 on the encoders list. so it can not find the opts options.

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ffmpeg 2017 synology?
by Mikkel Thomsen 07 May '17

07 May '17

Hello is there a 2017 release for my synology ds1817 +? ffmpeg

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Hello Fffmpeg Problem hELP mE
by Ümit Tftk 07 May '17

07 May '17

ERROR: libgsm 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 "config.log" produced by configure as this will help solve the problem. If ffmpeg compiled and installed with no major errors you may now close this window and continue

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Differing behaviours by fade and afade filters
by Gyan 07 May '17

07 May '17

The video fade filter has a start_frame option which starts the fade effect when the ordinal count of the processed frame reaches the start_frame value e.g. ffmpeg -f lavfi -i testsrc2=d=5 -vf trim=2,fade=s=25:d=1 vfade-works-by-frame-count.mp4 In the command above, the video output fades in starting at pts_time 3.00 whereas the command, ffmpeg -f lavfi -i testsrc2=d=5 -vf trim=2,fade=st=1:d=1 vfade-fails-by-timestamp.mp4 fails because 'st' references timestamps and trim does not reset them. With the afade filter, the command ffmpeg -f lavfi -i sine=d=5 -af atrim=2,afade=ss=44100:d=1 afade-fails-by-sample-count.mp4 fails. Audio remains at native volume throughout. But the command ffmpeg -f lavfi -i sine=d=5 -af atrim=2,afade=ss=88200:d=1 afade-works-if-sample-count-is-pts.mp4 works. Audio fades in from pts_time 2.00 to 3.00. Which indicates that 'ss' refers to the original sample PTS rather than the ordinal count of processed audio samples. Is this intended? If yes, can one emulate the behaviour of the video fade filter. If no, should I file a bug report? All commands executed with N-85917 git-cc25a887c5 Thanks, Gyan

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Re: [FFmpeg-user] creating a slideshow
by gheine＠mathnmaps.com 06 May '17

06 May '17

Some more information about this warning: It only occurs when one image in the input list differs in size from the previous image. If the dimensions are significantly different, visible distortion will occur. Also, in contrast to my previous post, it appears that PNG images are bypassed with a warning. JPG images do work; I don't know what other formats might be possible. _______________________________________________ 2017-05-01 09:25:52 UTC <ceffmpeg(a)gmail.com> [swscaler @ 0x9d13040] deprecated pixel format used,make sure you did set range correctly You cannot fix this warning and it should not trouble you. Carl Eugen _______________________________________________ ffmpeg -f concat -i input.txt -vsync vfr -pix_fmt yuv420p test.mp4 This did work. "test.mp4" is a video with all the images, each displaying for about 5 seconds. But ffmpeg gave the following complaint for each image: [swscaler @ 0x9d13040] deprecated pixel format used, make sure you did set range correctly

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Wanted to batch convert mp3 folder for youtube.
by Mike King 05 May '17

05 May '17

I wanted to simply convert a folder of mp3's into a video with an image in the folder as the video (simple youtube stuff) and found this line that claims to do the trick for %i in (*.mp3) do ffmpeg -f image2 -loop 1 -i folder.jpg -framerate 60 ^ -i "%i" -c:v libx264 -tune stillimage -c:a copy "%~ni.mp4" but i get a red line of text in my CMD window saying Option Framerate not found. Googling this problem has produced nothing i can perceive as the answer. nothing suggesting i spelled anything wrong (especially since i copy pasted it) and i am TERRIBLY fustrated. I always hated command line programs but this seems to be the only thing that lets me do what i am aiming to do. Everything else i saw in google results has a conversion option but nothing that lets me attach an image and do it to the entire mp3 contents of a folder. seems so simple but i have spent 3 hours on this. any friendly advice on the probably obvious thing I'm doing wrong would really help me out. thank you I can provide screenshots if necessary but i didnt want to risk worrying anyone with attachments from a stranger and breaching some protocol.

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Long delay on subtitle overlay from a subtitles embedded video
by Dennis Huang 05 May '17

05 May '17

Hi all, First of all, thanks for the great ffmpeg software. It is useful and neat. Good job. My question is regarding the long delay before ffmpeg transcodes a file, with following command, ffmpeg -i embed_sub.mkv -vf subtitles=embed_sub.mkv -sn result.mkv With -loglevel trace, it shows the command takes over 30 seconds going through ALL subtitles in mkv before it starts the transcoding. Yet this causes problem for a real-time transcoding server like Universal Media server, which needs the output stream within certain time, usually a few seconds. The long wait before the transcoding causes the server to timeout. When the delay is taking place, it stops with following messages, [Parsed_subtitles_0 @ 0x1c8f6e0] Shaper: FriBidi 0.19.6 (SIMPLE) HarfBuzz-ng 1.4.5 (COMPLEX) [Parsed_subtitles_0 @ 0x1c8f6e0] Using font provider fontconfig I have 2 questions, 1. Why ffmpeg needs to go through all the subtitles before the transcoding? And2. Is there any way ffmpeg could start the transcoding without the long delay in this case? The file, embed_sub.mkv includes following tracksTrack1 libx264 videoTrack2 dts audioTrack3 ASS subtitles Thank you for the time and information in advance. -Dennis

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wmv3+wma2.mov: Failed to open codec / Unable to parse option value "(null)" as sample format
by ffmpeg＠zoot.org 04 May '17

04 May '17

Hi, macosx-10.11.6, ffmpeg-3.2.4 (via macports) I have a strange old video that i'd like to convert. It's strange because it's a MOV file containing WMA2 and WMV3. I can only play it using quicktime on one old mac. It doesn't play on a recent mac or on windows10. The output of ffmpeg -codecs includes: DEA.L. aac AAC (Advanced Audio Coding) (decoders: aac aac_fixed aac_at libfdk_aac ) (encoders: aac aac_at libfdk_aac ) DEV.LS h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (decoders: h264 h264_vda ) (encoders: libx264 libx264rgb h264_videotoolbox ) DEA.L. wmav2 Windows Media Audio 2 D.V.L. wmv3 Windows Media Video 9 The output of ffmeg -formats includes: E mov QuickTime / MOV D mov,mp4,m4a,3gp,3g2,mj2 QuickTime / MOV E mp4 MP4 (MPEG-4 Part 14) If I try to convert it to mp4/h264/aac with: ffmpeg -i in.mov -vcodec h264 -acodec aac out.mp4 I get: ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable-libass --enable-libbluray --enable-lzma --enable-gnutls --enable-fontconfig --enable-libfreetype --enable-libfribidi --disable-libxcb --disable-libxcb-shm --disable-libxcb-xfixes --disable-indev=jack --disable-outdev=xv --enable-audiotoolbox --enable-sdl2 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-vda --enable-videotoolbox --arch=x86_64 --enable-yasm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-nonfree --enable-libfdk-aac libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libavresample 3. 1. 0 / 3. 1. 0 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 / 54. 1.100 [wmav2 @ 0x7fdbeb01cc00] block_align is not set [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fdbeb001200] Failed to open codec in av_find_stream_info [wmav2 @ 0x7fdbeb01cc00] block_align is not set Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'bj1.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2013-11-23T00:52:27.000000Z Duration: 00:01:00.09, start: 0.000000, bitrate: 6315 kb/s Stream #0:0(eng): Audio: wmav2 (WMA2 / 0x32414D57), 44100 Hz, stereo, 141 kb/s (default) Metadata: creation_time : 2013-11-23T00:52:27.000000Z handler_name : Apple Alias Data Handler Stream #0:1(eng): Video: vc1 (Advanced) (WVC1 / 0x31435657), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 6204 kb/s, 29.97 fps, 29.97 tbr, 2997 tbn, 59.94 tbc (default) Metadata: creation_time : 2013-11-23T00:52:27.000000Z handler_name : Apple Alias Data Handler [abuffer @ 0x7fdbea5006c0] Unable to parse option value "(null)" as sample format Last message repeated 1 times [abuffer @ 0x7fdbea5006c0] Error setting option sample_fmt to value (null). [graph 1 input from stream 0:0 @ 0x7fdbea5007e0] Error applying options to the filter. Error opening filters! Any idea what this is trying to tell me? Is there something wrong with the file? Or with the ffmpeg installation? Or the command? Or something else? Thanks.

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Extracting mp4 from a disk image
by John P 04 May '17

04 May '17

I am attempting to extract all available mp4 files from an image of a drive which was created by a Hikvision DVR. If I use ffplay then it will start to play back recorded video, but can't do much with it (I expect about 60 hours of video). There's lots of errors on the console (illegal reordering etc) but it plays. I tried using ffmpeg -err_detect ignore_err -i image.img -c copy output.mp4 but this just generated a large file with many hours of blank h264 1920x1080 video. Is there some way to use ffmpeg to just extract all the video into a new file, or files ? thanks in advance John

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Getting silence level to be used with silencedetect automatically
by Alexander Winnig 04 May '17

04 May '17

My goal is to find the average or maximum silent level in dB to use it with|silencedetect|. I found|volumedetect|and I thought to use the|histogram_|results to find the lowest dB numbers(low as in -40dB, -50dB etc.) with a high number of occurrences. What is a better idea? Can it be combined with the silencedetect command, so I don't need to enter the dB-Value at all? Best

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