September 2010 - ffmpeg-user - lists.ffmpeg.org

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 24 Feb '16

24 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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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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UDP Multicast streaming
by omar.gomez＠gmail.com 05 Oct '13

05 Oct '13

Hi!, I've been using the FFmpeg + Mplayer combo for a college project I'm working on. The idea is to stream an Opengl app's buffer is real-time in order to have thin-clients interact remotely with the app. I've succeed at building an scenario where the buffer is pulled out of the app and injected to a pipe using the following command: cat <fifo> | ppmtoy4m -S 420jpeg | ffmpeg -y -f yuv4mpegpipe -i - -f avi udp:127.0.0.1:1234 Clients then use the following command to view the stream: mplayer -demuxer mpeg4es udp://@127.0.0.1:5000 This has been working all right, but I want to achieve the following: 1) Work in a multicast scenario. ======================== I though it was as easy as to change the address for a multicasting one like the following: cat <fifo> | ppmtoy4m -S 420jpeg | ffmpeg -y -f yuv4mpegpipe -i - -f avi udp:224.0.0.1:5000?ttl=1 Using the ?ttl parameter was something I found by googling, but I'm not sure if that works. At the same time MPLayer clients should work by using: mplayer -demuxer mpeg4es udp://@224.0.0.1:5000 Which works, but I'm wondering why I *have* to add a port (being a multicast address) and why it doesn't work if I open a second client (I believe is still working as unicast) 2) Using VLC ========== I've not been able to make this work with VLC. That "demuxer" parameter seen to be the problem. 3) Use different containers,codecs ================== This seems to be working using mpeg4+yuv420p. I would like to experiment with other codes as well, like h263p or h262. Have any of you have had one of these problems? I appreciate any help --Omar -- Follow me at: Twitter: http://twitter.com/omargomez Buzz: http://www.google.com/profiles/108165850309051561506#buzz Blog: blog.pixagora.com

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Extracting MXF Metadata
by hsomob1999＠yahoo.com 03 Sep '13

03 Sep '13

Hello, Is it possible to retrieve MXF metadata with ffmpeg? Is there way to extract metadata from any format? I see that there is a -metadata flag but this only allows one to set metadata. Thanks

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iMovie created variable frame rates cause a/v sync problems
by jdrukman＠gmail.com 22 Jul '13

22 Jul '13

I have a file generated from Apple's "iMovie HD". It has a variable frame rate, according to mediainfo: Frame rate mode : Variable Frame rate : 29.402 fps Minimum frame rate : 0.303 fps Maximum frame rate : 29.970 fps Original frame rate : 29.970 fps I am trying to get ffmpeg to create a transcoded version suitable for web & iphone streaming, but the audio & video are out of sync. I used ffmpeg -i original.m4v -acodec copy -vcodec libx264 output.mp4 I also tried with -acodec libfaac. Same result. Oddly enough, mediainfo on the ffmpeg converted file shows the same frame rate information, but the soundtrack is clearly not in sync: Frame rate mode : Variable Frame rate : 29.402 fps Minimum frame rate : 0.303 fps Maximum frame rate : 29.970 fps Original frame rate : 29.970 fps Is there some simple command line option that I'm missing to keep the two in sync? I can provide the original file if needed. The current version of iMovie does not have the same problem because it outputs at constant frame rate by default. -jsd-

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Frame rate and movies from images
by josh＠uncommonprojects.com 22 Jul '13

22 Jul '13

I am using ffmpeg to create movies from a directory of images. This is the command I use: ffmpeg -i imagedir/%05d.jpg -b 1800 video.mp4 This works great except that I want to be able to slow down the playback to say 6 frames per second. I tried this: ffmpeg -i imagedir/%05d.jpg -r 6 -b 1800 video.mp4 Which does playback at 6 frames per second, but what seems to be happening is that ffmpeg is also skipping frames. So if I have say 506 images in the images directory, the resulting movie uses only 123 frames. I can see where this something you might want, if you don't want the duration of the output movie to change with different frame rates, but is there a way to force it to use all the frames and extend the duration? I tried using the -vframes flag but it didn't seem to do anything: ffmpeg -i imagedir/%05d.jpg -r 6 -vframes 506 -b 1800 video.mp4 thanks, josh

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change frame rate
by info＠danielerossi.net 22 Jul '13

22 Jul '13

> I would like to convert my video source, in lenght 07 min 40 sec frame rate 30 per second to a frame rate of 60 per second without adding new frames, that means the duration will be the half. I did try ffmpeg -r 30 -i input_mjpg.avi -r 60 -vcodec copy -an test.avi and does not change the frame rate. Can anybody suggest me the correct command line? thank you D

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"error, non monotone timestamps"
by kshawkeye＠gmail.com 28 May '13

28 May '13

Hi, when trying to encode from a .mov with MPEG-4 video and PCM audio into .mp4 with MPEG-4 video and AAC audio I get: [avi @ 0x956cda0]st:0 error, non monotone timestamps 1028 >= 102889.9kbits/s av_interleaved_write_frame(): Operation not permitted I tried to demux and then mux them: Demux: ffmpeg -i 'Infile.mov' -vn -acodec libfaac -ab 320k -ar 48000 -ac 2 'Outfile.aac' and ffmpeg -i 'Infile.mov' -vcodec copy -an 'Outfilet.m2v' Mux: ffmpeg -i 'Outfile.m2v' -i 'Outfile.aac' -vcodec copy -acodec copy 'Final.mp4' While I finally got the audio to convert this way, the resulting Final.mp4 was out of sync. I tried to do a normal encode: ffmpeg -i 'Infile.mov' -vcodec copy -acodec libfaac -ab 320k -ar 48000 -ac 2 'Outfile.mp4' Which lead to my "error, non monotone timestamps" problem.

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Changing aspect ratio flag without re-encoding
by mi.lists.ffmpeg＠alma.ch 16 May '13

16 May '13

I tried to use FFmpeg to change the aspect ratio flag in an MPEG file, but it didn't. Am I doing it wrong, or is it not possible without re-encoding? Here, the Output part correctly shows "DAR 16:9", but the resulting file is still "DAR 4:3" : $ ffmpeg -i INPUT.m2v -aspect 16:9 -vcodec copy OUTPUT.m2v FFmpeg version SVN-r18021, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-libxvid libavutil 50. 0. 0 / 50. 0. 0 libavcodec 52.21. 0 / 52.21. 0 libavformat 52.31. 1 / 52.31. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 18 2009 14:40:53, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1) Input #0, mpegvideo, from 'INPUT.m2v': Duration: 00:00:11.33, bitrate: 8000 kb/s Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 8000 kb/s, 25 tbr, 1200k tbn, 50 tbc Output #0, mpeg2video, to 'OUTPUT.m2v': Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 8000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 282 fps= 0 q=-1.0 Lsize= 11072kB time=11.28 bitrate=8041.3kbits/s video:11072kB audio:0kB global headers:0kB muxing overhead 0.000000% Looking at the result: $ ffmpeg -i OUTPUT.m2v FFmpeg version SVN-r18021, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libx264 --enable-libxvid libavutil 50. 0. 0 / 50. 0. 0 libavcodec 52.21. 0 / 52.21. 0 libavformat 52.31. 1 / 52.31. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 18 2009 14:40:53, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 25.00 (25/1) Input #0, mpegvideo, from 'OUTPUT.m2v': Duration: 00:00:11.33, bitrate: 8000 kb/s Stream #0.0: Video: mpeg2video, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 8000 kb/s, 25 tbr, 1200k tbn, 50 tbc

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