December 2012 - ffmpeg-user - lists.ffmpeg.org

Compiling FFMPEG with ffplay support
by Stas Oskin 13 Jun '17

13 Jun '17

Hi. Did the compilation parameters changed in recent FFMPEG version? Configuring the FFMPEG like this: ./configure --prefix=/usr --enable-pthreads Does not produce ffplay anymore. I tried to add --enable-ffplay, without any result. Any idea what I'm missing? Thanks.

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How do I encode DVB Subtitles?
by Wagner Patriota 02 May '17

02 May '17

*QUESTION 1:* How do I add a STR file to a MPEG-TS file with the "DVB subtitle codec"? If so, could you give me a sample command line? Thank. I am trying: ffmpeg -i video.ts *-i subtitle.srt* -vcodec copy -acodec copy *-scodec dvbsub* output.ts The stream is created because I can see it with MediaInfo, but the subtitles don't appear! *QUESTION 2:* How do I use the command *slang*? I try to use it, but ffmpeg says the command doesn't exist. Thank you.

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How to use global_header, fflags, rtpflags parameter.
by Ly Tran Cong 06 Dec '16

06 Dec '16

Hello, Can anyone tell me how to use flags +global_header, fflags +latm, rtpflags +latm Which parameter using with video encode and which parameter using with audio encode. Thanks -- Regards, Tran Ly Vega Corporation 98 Hoang Quoc Viet Str, Hanoi, Vietnam Tel: 84 4 755 4190 Fax: 84 4 755 4190 Mobile: 84 91 487 1115 www.vega.com.vn; www.clip.vn; www.chacha.vn; www.ringring.vn

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Flip in sws_scale
by romulo.fernandes＠gmail.com 11 Mar '16

11 Mar '16

Hello, I'm using libswscale to scale and convert pixel format from frames that I capture from the screen. In my windows build the frames that I capture are in DIB format. I'm using FFMPEG SVN rev 16437. Calling sws_scale normally I get the image correctly, except that it's vertically flipped. If I change the srcSliceY from 0 to srcHeight and srcSliceH from srcHeight to 0, I get a grey image (with correct dimensions). If I keep the srcSliceY in 0 and change srcSliceH from srcHeight to -srcHeight, I also get that grey image. Is there a way to make sws_scale flip the image for me ?

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Howto compile ffplay.c in seperate directory with my own svn-version of ffmpeg
by tps＠netmaster.dk 23 Feb '16

23 Feb '16

Dear All I have compiled svn-version of ffmpeg. I only do configure and make. I do not do make install because I don't want this svn-version of ffmpeg to overwrite the 'official' Ubuntu version of ffpeg that is also on the host. Now I want to compile my own version of ffplay.c so I create a directory ffplay at the same level as the svn ffmpeg-directory. I copy ffplay.c from ffmpeg to ffplay. Now what gcc options should I use to compile ffplay.c? I tried with this but got tons of errors: gcc -o ffplay ffplay.c -I../ffmpeg -L../ffmpeg/libavcodec -L../ffmpeg/libavformt -L../ffmpeg/libswscale -L../ffmpeg/libavutil -lavcodec -lavformat -lswscale -lavutil -lmp3lame -lx264 `sdl-config --cflags --libs` I note that when I do the make in the svn ffmpeg-directory there are a lot of options on the gcc command. I hardly know any of the options. Thanks in advance

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timestamp calculation
by Chau Pham 24 Jun '15

24 Jun '15

Hi Everybody, Suppose that I know AVPacket element: pts, dts, duration How could I calculate the timestamp (hh:mm:ss:MMM)? Thank You,Chau Pham.

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Use build_libstagefright in ffmpeg to build file .so for android
by hsu tom 26 Nov '14

26 Nov '14

In order to use ffmpeg in android , I tried to build ffmpeg by using build_libstagefright in ffmpeg/tools/lib_stagefright but it alwalys appear a error which i coundn't solve it. the message in the bottom of ffmpeg/config.log is arm-linux-androideabi-g++ -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC -I../android-source/frameworks/base/include -I../android-source/system/core/include -I../android-source/frameworks/base/media/libstagefright -I../android-source/frameworks/base/include/media/stagefright/openmax -I/home/user/android-ndk/sources/cxx-stl/gnu-libstdc++/include -I/home/user/android-ndk/sources/cxx-stl/gnu-libstdc++/libs/armeabi-v7a/include -march=armv7-a -mfloat-abi=softfp -mfpu=neon -march=armv7-a -std=c99 -fomit-frame-pointer -fPIC -marm -Wno-multichar -fno-exceptions -fno-rtti -D__STDC_CONSTANT_MACROS -c -o /tmp/ffconf.6ZzEMs87.o /tmp/ffconf.5jbfvJxK.cpp ./configure: line 701: /home/user/android-ndk/toolchains/arm-linux-androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-g++: cannot execute binary file ERROR: libstagefright_h264 not found my Build Environment is 1. Window 7 2. Cygwin 3. android-NDK-r8b linux version Tom

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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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