January 2013 - ffmpeg-user - lists.ffmpeg.org

Brighten dark video before reducing size?
by Gilles 06 Oct '14

06 Oct '14

Hello Using a basic Sony camcorder, I recorded a two-hour conference whose picture is a bit dark. I tried applying a filter in AviDemux and saving the file, but for some reason, there's no sound when played in VLC :-/ Before trying other NLE applications (VirtualDub, Sony Vegas, etc.), I was wondering if ffmpeg could apply some filters before further reducing the filesize? FWIW... 1. here's what the original file looks like: Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s 2. Using the following command, ffmpeg reduces the original 9GB to 340MB: ffmpeg -i input.mpg -c:v libx264 -vf scale=iw/2:-1 -preset faster -c:a ac3 -ab 192k output.mp4 Thank you.

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Re: [FFmpeg-user] Problem trying to convert PAL file to NTSC
by vsethi＠iglou.com 30 Jun '14

30 Jun '14

belcampo wrote: > You need to specify -sn on your command line, to disable the > subtitle-part in the source. > What you seem to have AFAIK is a MPEG2-TS container, broadcast, with > 1 video > 1 subtitle > 4 audio-streams. > ffmpeg AFAIK doesn't know how to handle the subtitle stream. You'll have > to explicitedly disable it with the -sn parameter. > If you would like to have another audiostream then the first one, you'll > have to use the map functions. > For video + 2nd audio stream you would need: > ffmpeg -i source.mpg copy or encode parameters -map 0:0 -map 0:3 > destination.mpg Thanks for your help. I've continued to play with this, without success. I've specified not only -sn, as you suggest, but also -an as a test to keep the audio streams from interfering. ffmpeg accepts this command line but, unfortunately, it's crashing with an illegal instruction error and generates only an empty output file. I am wondering if any of the following needs to be addressed: 1. max_analyze_duration reached 2. stream 0 codec frame rate differs from container frame rate 3. bitrate: -2147483 kb/s This is what it is doing now: ========================================================================== ffmpeg -f mpeg -i image.mpg -sn -an -target ntsc-dvd image.out FFmpeg version git-31ff9bd, Copyright (c) 2000-2011 the FFmpeg developers built on Mar 12 2011 11:21:15 with gcc 4.1.2 configuration: libavutil 50. 39. 0 / 50. 39. 0 libavcodec 52.113. 2 / 52.113. 2 libavformat 52.102. 0 / 52.102. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1. 76. 0 / 1. 76. 0 libswscale 0. 12. 0 / 0. 12. 0 [mpeg @ 0x8b8f600] max_analyze_duration reached [mpeg @ 0x8b8f600] Further flags set but no bytes left Last message repeated 1 times Last message repeated 2 times Last message repeated 3 times Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 5000.00 (5000/1) Last message repeated 3 times Input #0, mpeg, from 'image.mpg': Duration: 00:00:02.73, start: 0.066600, bitrate: -2147483 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 9282 kb/s, 25 fps, 5k tbr, 90k tbn, 50 tbc Stream #0.1[0x20]: Subtitle: dvdsub Stream #0.2[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 384 kb/s Stream #0.3[0x1ce]: Audio: mp2, 0 channels Stream #0.4[0x1d1]: Audio: mp2, 0 channels Stream #0.5[0x1d7]: Audio: mp2, 0 channels [buffer @ 0x8ba2bd0] w:720 h:576 pixfmt:yuv420p [scale @ 0x8ba2f90] w:720 h:576 fmt:yuv420p -> w:720 h:480 fmt:yuv420p flags:0xa0000004 Output #0, dvd, to 'image.out': Metadata: encoder : Lavf52.102.0 Stream #0.0: Video: mpeg2video, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=2-31, 6000 kb/s, 90k tbn, 29.97 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 0 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbits/s Illegal instruction

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axis mpeg4 stream w/ aac audio
by danielp＠choma.hu 30 May '14

30 May '14

Hello, i've an Axis 247s video server which provides mpeg4 stream w/ aac audio, and i would like to get an flv stream from this. i've compiled ffmpeg/ffserver with --enable-libmp3lame --enable- shared --enable-libamr-nb --enable-nonfree --enable-gpl --enable- libfaad --enable-libfaac i can get the stream work with ffserver without audio, but not with audio. my ffserver.conf: Port 8090 BindAddress 0.0.0.0 MaxClients 1000 MaxBandwidth 1000 CustomLog - NoDaemon <Feed feed1.ffm> File /tmp/feed1.ffm FileMaxSize 1G ACL allow 127.0.0.1 </Feed> <Stream test.flv> # the source feed Feed feed1.ffm # the output stream format - FLV = FLash Video Format flv VideoCodec flv # this must match the ffmpeg -r argument VideoFrameRate 25 # generally leave this is a large number VideoBufferSize 80000 # another quality tweak VideoBitRate 500 # quality ranges - 1-31 (1 = best, 31 = worst) VideoQMin 1 VideoQMax 5 VideoSize 640x480 # this sets how many seconds in past to start PreRoll 0 AudioBitRate 64 AudioChannels 1 AudioSampleRate 44100 AudioCodec mp3 </Stream> i've tried several things like: openRTSP -b 80000 -y -c -4 -f 25 -w 640 -h 480 -t rtsp://xxx.hu:554/mpeg4/1/media.amp | \ ffmpeg -v 9 -vcodec mpeg4 -acodec mpeg4aac -ab 32k -i - -acodec libmp3lame -ar 44100 -ab 64k -ac 1 -vcodec mpeg4 http://localhost:8090/ or: mencoder \ -rtsp-stream-over-tcp rtsp://xxx.hu:554/mpeg4/1/media.amp \ -ovc lavc -lavcopts vcodec=mpeg4:vbitrate=500:mbd=2:mv0:trell:v4mv:cbp:last_pred=3 \ -oac mp3lame -lameopts abr:br=64:q=1:mode=3 \ -o - | \ ffmpeg \ -ab 32 -acodec mp3 -ar 44100 -ac 1 -f mpeg -r 25 -i - \ -r 25 -ab 32 -acodec mp3 -ar 44100 -ac 1 http://localhost:8090/ and tons of switches, but nothing worked. can you please help me somehow? thx, -- dp

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converting to non HD avi with ffmpeg
by Kevin Wilson 12 Apr '14

12 Apr '14

Hi, I have downloaded from the web a video file with webm extension: ubunutuArm.webm (some lecture in video) Now, I converted it to avi thus: ffmpeg -i ubunutuArm.webm ubunutuArm.avi and put the ubunutuArm.avi on disk On Key. When trying to view it on TV with a DVD I see this messsage: "HD is not supported". Indeed, HD is not supported in my DVD and TV. Is there a way I can convert this video file with ffmpeg to an AVI so the result will be non HD ? (Second question: given some avi file, is there a way I can know whether it is in HD or not ?) Regards, Kevin

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Combining multiple MP3's?
by Gilles 03 Mar '14

03 Mar '14

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AC3 encoding problem - wrong audio tag?
by Stefan Gofferje 14 Jan '14

14 Jan '14

Hi, I have recently changed my home video editing to Linux (kdenlive) to get finally completely rid of Windows. Now I'm trying to encode some home-video with ac3 sound. I do the testing/playing with a Pinnacle Showcenter network media player which is connected via fiber-optic cable to a Haman Kardon amplifier. When I play my the encoded file, the amp show "Dolby 2/0" and is silent. The Showcenter shows as audio type "Ogg". File shows RIFF (little-endian) data, AVI, 624 x 352, 23.98 fps, video: XviD, audio: (6 channels, 48000 Hz) If I check older files that work on the Showcenter, the output of file is: RIFF (little-endian) data, AVI, 624 x 352, 23.98 fps, video: XviD, audio: Dolby AC3 (stereo, 48000 Hz) So in my encoding, the "Dolby AC3" tag seems to be missing. I use ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers built on Jun 15 2012 15:33:11 with gcc 4.5.1 20101208 [gcc-4_5-branch revision 167585] configuration: --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib64 --shlibdir=/usr/lib64 --disable-stripping --enable-pthreads --enable-shared --enable-avfilter --disable-static --enable-libfaac --enable-libmp3lame --enable-libxvid --enable-libx264 --enable-nonfree --enable-postproc --enable-shared --enable-libdc1394 --enable-libtheora --enable-libschroedinger --enable-libvpx --enable-gpl --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-x11grab libavutil 51. 54.100 / 51. 54.100 libavcodec 54. 23.100 / 54. 23.100 libavformat 54. 6.100 / 54. 6.100 libavdevice 54. 0.100 / 54. 0.100 libavfilter 2. 77.100 / 2. 77.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 and my command-line is like ffmpeg -i video.mp4 -i left.wav -i right.wav -i surr_left.wav -i surr_right.wav -i center-wav -i lfe-wav -vcodec libxvid -s 624x352 -b:v 1100k -acodec ac3 -ar 48000 -ab 448k video.avi Any hints? -Stefan -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -------------------------------------------------------------------------------- This mail was scanned for malware by ClamAV (http://www.clamav.net/) --------------------------------------------------------------------------------

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meaning of macroblock colors when using vis_mb_type option
by Wesley S. 29 Dec '13

29 Dec '13

Hi, I was wondering what all the different macroblock colors meant when using the vis_mb_type option? Kind regards, Wesley

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x11grab video stalls every 30 seconds in screencapture
by Troy Gardner 11 Dec '13

11 Dec '13

We are using latest ffmpeg, for screencapture in the cloud. We are seeing odd dropped frame stalls in video capture every 30 seconds for about 1-2 seconds in duration. Audio is captured without stalls, and Audio Video sync is perfectly fine despite stalls. Hoping someone had ideas, this is driving us nuts and there's nothing obvious to explain it. This is the capture setting we are using. ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -acodec libvorbis -ar 44100 -vcodec libx264 -preset ultrafast -gp 0 -threads 0 ffg4.mkv This is the a drop frame test case capture with lossless x264 that shows dropped frames around 25-26 and significant stall errors at 30 and 60 seconds http://66.220.0.170/XFG4.mkv We get this trying different options, like keyframes. This is the output with flashsrv, which shows similar pause around 23 and 53 seconds, and indicates it's not a libx264 specific issue. flashsrv http://66.220.0.170/out.mov This is running on an instance with 16 vcpu + 16gb ram, on the Amazon AWS there is no noticeable background process that we can see at least that would explaing and happens regardless if it's debian, or ubuntu. This is the version of ffmpeg. ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers built on Jun 26 2012 12:42:06 with gcc 4.7.1 configuration: --arch=x86_64 --enable-avfilter --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static --enable-libx264 --enable-libvpx --enable-libmp3lame --enable-libfaac --enable-nonfree --enable-libopencore-amrwb --enable-libopencore-amrnb --enable-version3

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feature request: able to link against libjpeg
by Roger Pack 14 Nov '13

14 Nov '13

Hello. I suppose that ffmpeg has its own "internal" jpeg library for decoding/encoding jpeg's. My assumption is also that it is "unoptimized" (at least compared to similar libraries, like libjpeg-turbo). My wish/goal would be to have the speed of libjpeg-turbo. Maybe a configure option to use libjpeg instead of the internal stuff? Thoughts? Thanks. -r

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Audio sync issues when transcoding Hauppauge HDPVR content
by Scott Naef 27 Oct '13

27 Oct '13

I am attempting to transcode recordings made with an HDPVR. I am using Mythtv to do the capture. I originally had trouble with playback in Myth but was able to use the environment variable FORCE_DTS_TIMESTAMPS=1 to get proper playback in Myth. I am an absolute newbie when it comes to the finer details of the workings of these various media packages. I have read several places that the HDPVR produces broken PTS timestamps. However, if there is a workable solution to playback content in Myth, I am hoping there is a workable solution to do transcoding with FFMPEG. I have tried several options in FFMPEG such as -async 1 but nothing I try seems to work. I always end up with about 1.5 second delay in video from audio. From what I understand DTS timestamps are used to decode content and offer a deterministic way to predict the number of frames to decode and PTS timestamps are used to show the correct frames in the correct order. If this is true (long shot!) then I guess I need a way to mux the audio and video streams using DTS not PTS. Is this way the Myth player does it? Version: ffmpeg version git-2013-01-05-7e5d4fa Copyright (c) 2000-2013 the FFmpeg developers built on Jan 5 2013 17:29:28 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 13.100 / 52. 13.100 libavcodec 54. 85.101 / 54. 85.101 libavformat 54. 59.104 / 54. 59.104 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 32.100 / 3. 32.100 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Any help would be greatly appreciated. If needed I can grab a few seconds of a news clip (talking heads help to see this) and offer it up somewhere. Thank you! -Scott Naef

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