May 2009 - 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 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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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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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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convert a sequence of images to a video file
by joshjdevl＠gmail.com 04 Oct '12

04 Oct '12

Hi, I would like to convert a series of images (png files) to a video. Would something like this work? ffmpeg -r 29.97 -i /images/picture%.png out.avi When I try the above, I get picture%d.jpg: I/O error occurred Usually that means that input file is truncated and/or corrupted. Thanks, Josh

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named pipes - > x264
by joshjdevl＠gmail.com 29 May '12

29 May '12

Hi, I was following these instructions to do a named pipe to x264 http://ffmpeg.x264.googlepages.com/piping mkfifo fifo.y4m ffmpeg -i test.mp4 -an -f yuv4mpegpipe - > fifo.y4m & x264 --quiet --crf 18 -o video.264 fifo.y4m For some reason, ffmpeg just sits here Stream mapping: Stream #0.1 -> #0.0 Press [q] to stop encoding It doesn't do any processing of test.mp4 and basically sits as a blank console. If I do something like ffmpeg -i test.mp4 -an -f yuv4mpegpipe - > test.y4m & Where test.y4m is not a named pipe, then test.y4m will end up being the yuv4 data... So I guess my question is, is there some extra option I need to compile ffmpeg or x264 with? Or is there another option I need to pass in? I tested this on Ubuntu 8.04 and centos 5.2 Thanks, Josh

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.m4a to .flv - image/video fails
by nreddig＠ptd.net 31 Mar '12

31 Mar '12

Hi all, I have greatly appreciated the information I have already gleaned from reading this forum. However I still can not solve my current problem. I am trying to convert from .m4a to .flv. These are podcast files (.m4a) built in garage band. Pictures were added as well and we would like to use this format to create weekly announcements because Garage Band is so easy for our non-technically oriented staff. Using ffmpeg the audio converts just fine but the "video" is not right. I guess it's because these files include chapter data not actual video. However when I convert it to any format the images fly through at different rates, then it seems to get stuck on a picture in the middle. The format I would like to convert to is flash so that I can easy display multiple announcement files using a flash player. I can convert .mov to .flv no problem. Converting .m4a to .mov and then .flv doesn't produce any better results. Here are the before and after files: http://www.warwicksd.org/warwick.m4a http://www.warwicksd.org/output.flv I used the following to get my conversion: ffmpeg -i warwick.m4a -b 300k -r 12 -s 300x300 -ar 22050 -y -ab 96k output.flv FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-libfaac --enable-libfaad --enable- libmp3lame --disable-vhook --enable-shared libavutil version: 49.5.0 libavcodec version: 51.44.0 libavformat version: 51.14.0 built on Oct 22 2007 15:31:25, gcc: 3.3 20030304 (Apple Computer, Inc. build 1640) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'warwick.m4a': Duration: 00:00:55.9, start: 0.000000, bitrate: 100 kb/s Stream #0.0(eng): Audio: mpeg4aac, 44100 Hz, stereo Stream #0.1(eng): Video: mjpeg, yuvj420p, 300x300, 44100.00 fps(r) Stream #0.2(eng): Data: tx3g / 0x67337874 Stream #0.3(eng): Data: text / 0x74786574 Output #0, flv, to 'output.flv': Stream #0.0(eng): Video: flv, yuv420p, 300x300, q=2-31, 300 kb/s, 12.00 fps (c) Stream #0.1(eng): Audio: libmp3lame, 22050 Hz, stereo, 96 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 83 fps= 21 q=2.0 Lsize= 1099kB time=6.9 bitrate=1301.6kbits/s video:70kB audio:657kB global headers:0kB muxing overhead 51.296550% .mov files convert nicely to .flv but these .m4as are just not working. Our current staff is very eager to use these Garage Band files to communicate announcements. I have been so desperate to get this particular file type to convert that I have been experimenting with grabbing the frames every 5 secs. and then rebuilding the images into video. This produces strange results and seems to be more programming then should be necessary. Thanks in advanced for any solutions you can provide. Nancy Reddig Web Programmer Warwick SD

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milliseconds to mpeg-ts pts/dts
by steuer＠unixsystems.de 07 Sep '11

07 Sep '11

guys, i try to create an mpegts file from H.264 and ACC raw frames that i receive from a network connection. I create AVPacket structures, set data, size, pts and dts etc. manually. For the stream that I receive, PTS and DTS seem to be equal, so I just calculate the packet pts and dts with incoming pts / 1000.0 * 90000.0 I do that for audio and video. Unfortunately that seems to be wrong as ffmpeg states "dts < pcr, TS is invalid" after a couple of seconds. I seems that the timestamps run out of sync with the internal pcr reference clock. What would be the correct way to calculate the pts and dts of the mpegts stream? Thanks alot for your help! Cheers, Heri

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