May 2011 - 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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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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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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UDP Multicast streaming
by omar.gomez＠gmail.com 04 Oct '13

04 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 02 Sep '13

02 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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