June 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.

6 9

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.

7 15

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

2 1

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 ?

8 11

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

5 5

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

3 2

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

2 1

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

6 6

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

7 10

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

2 1