May 2005 - ffmpeg-user - lists.ffmpeg.org

reading a series backwards
by bailey＠utias.utoronto.ca 17 May '05

17 May '05

Hello, I have had really good results so far with FFMPEG, so thanks to all the developers! I don't know how I can accomplish my latest task, so any help would be appreciated. I want to convert a two series of gifs into a mpeg. Both series run from 0 to 100, but I also want each series to run backwards from 100 to 0. Therefore, the complete mpeg will contain the four series as given below: 0 -> 100 first series of gifs 100 -> 0 first series 0 -> 100 second series 100 -> 0 second series >From the documentation, I can include multiple inputs with multiple '-i', but is there a way to read the series backwards? Thanks, Brent

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precompiled windows dlls?
by dallen＠sequencedesign.com 16 May '05

16 May '05

Folks, This must be a frequently asked question, but I can't find the answer in the FAQ, or in the recent archive of this list, or by using google with the keywords I thought of. I would like to use libavcodec.dll and libavformat.dll in my Windows XP Visual C++ 6 project. However, I would prefer not to download a whole new compiler environment and compile ffmpeg from sources, just to get the dlls. Is there anyplace I can find relatively recent precompiled dll's and a matching header file? Thanks in advance, - Dave Allen: dallen at sequencedesign.com

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Gop
by runlevel_4＠passagen.se 16 May '05

16 May '05

Dearest dudes... How do i create a STRICT (hm) VideoCD mpeg1 a/v file with a Gop like this, IBBPBBPBBPBBPBBPBB Me not like IPPPPP... _______________________________________________________ Skicka gratis SMS! http://www.passagen.se

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Unable to stream from .asf file
by ferlin＠plink.it 16 May '05

16 May '05

I have ffserver running on Openbsd. Config file is Port 8090 BindAddress 0.0.0.0 MaxClients 1000 MaxBandwidth 1000 CustomLog - NoDaemon <Stream file.asf> File "/usr/jpeg/200ms/video.asf" NoAudio Author "It's me" Copyright "Myself" Title "ASF file test" Comment "Trial" </Stream> If I try http://ip_add/fils.asf Windows Media Player starts, I can see all descriptions (Author, Copyright and so on) but does nothing and after a while it produce an error windows "Connection aborted (Error=C00D0029). It seems it does not stream anything..... Please help!!! Andrea

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3gp / amr
by manu＠cappelleri.net 16 May '05

16 May '05

Hi, sorry for the question... but Ive spent the last few days searching for a reply somewhere with no luck... That's the point: I am trying to encode a 3gp compliant file... compiled ffmpeg with amr support I run this command: ffmpeg -i source.mpg -acodec amr_nb-ar 8000 -ac 1 -ab 32 -vcodec h263 -s qcif -r 10 target.3gp it seems to encode correctly, but the file I get is not quite good as it should, for example: a) it crashes windows media player; b) it doesn't reproduce in 3G terminal (expecially NEC e606) while trying to check it out with mplayer under linux, that is what I see: *************************** Playing Ev.3gp. ISO: File Type Major Brand: 3GPP Profile 4 QuickTime/MOV file format detected. -------------- MOV track #0: 397 chunks, 397 samples MOV: Found unknown movie atom d263 (15)! Image size: 176 x 144 (24 bpp) Display size: 180 x 144 Fourcc: s263 Codec: 'h263' -------------- MOV track #1: 1976 chunks, 0 samples Audio bits: 8 chans: 1 rate: 8000 MOV: Found unknown audio atom damr (17)! Fourcc: samr -------------- MOV: longest streams: A: #1 (1976 samples) V: #0 (397 samples) VIDEO: [s263] 176x144 24bpp 10.000 fps 0.0 kbps ( 0.0 kbyte/s) ************************* Please, take a look at the audio part: 1976 chunks, 0 samples... isn't it strange? A valid 3gp file, for example, says: MOV track #1: 127 chunks, 127 samples Audio bits: 8 chans: 1 rate: 8000 MOV: Found unknown audio atom damr (17)! Fourcc: samr chunks and samples report the same numbers Could someone help me out? ffmpeg version 0.4.9-pre1, build 4754, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-mp3lame --enable-vorbis --enable-faadbin --disable-faac --enable-xvid --enable-a52bin --disable-dts --enable-pp --enable-shared-pp --enable-amr_nb --enable-amr_nb-fixed --disable-sunmlib --enable-pthreads --enable-dc1394 --enable-gpl --disable-strip (Sorry, I know almost nothing specifically about audio/video ... :) Manuele

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[Ffmpeg-user-old] problems compiling on mingw
by pgiacon＠ing.unife.it 16 May '05

16 May '05

I m trying to compile version 0.4.7 with mingw32 on windows OS. I know that u do not support old versions of mmpeg but in order to try and use the RTP patch for h263/RFC2190 i must use this version. i've got problem with undefined reference to ff_pw_3 and ff_pw_2, so it doesn't compile. any suggestions? thanks for your help paolo -- C.N.I.T Consorzio Nazionale Interuniversitario per le Telecomunicazioni c/o Unit? di Ricerca di Ferrara Dipartimento di Ingegneria, Universita' di Ferrara Via Saragat 1, Ferrara tel 349/7549790 email:gamma2002 at virgilio.it ------------------------------------------------------- This SF.Net email is sponsored by Oracle Space Sweepstakes Want to be the first software developer in space? Enter now for the Oracle Space Sweepstakes! http://ads.osdn.com/?ad_ids93&alloc_id281&op=click _______________________________________________ Ffmpeg-user mailing list Ffmpeg-user at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/ffmpeg-user

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Unable to send feed from ffmpeg to ffserver
by slava＠mutatorr.net 16 May '05

16 May '05

I just compiled ffmpeg/ffserver on a FreeBSD 5.3 machine from the CVS, and am attempting to send ffserver a stream from a dv camera writting to a named pipe read by ffmpeg the set up is as follows: mkfifo camera.dv fwcontrol -R camera.dv ffserver -f /usr/local/etc/ffserver.conf ffmpeg -i camera.dv http://localhost:8090/feed1.ffm The output I get from ffmpeg: ffmpeg version 0.4.9-pre1, build 4754, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-memalign-hack built on May 16 2005 06:03:36, gcc: 3.4.2 [FreeBSD] 20040728 Input #0, dv, from '/home/slava/camera.dv': Duration: N/A, start: 0.000000, bitrate: 28771 kb/s Stream #0.0: Video: dvvideo, yuv411p, 720x480, 29.97 fps, 25000 kb/s Stream #0.1: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Stream #0.0: Audio: mp2, 44100 Hz, mono, 32 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, 3.00 fps, q=3-31, 64 kb/s Stream #0.2: Audio: mp2, 22050 Hz, mono, 64 kb/s Stream #0.3: Video: msmpeg4, yuv420p, 352x240, 15.00 fps, q=3-31, 256 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Stream #0.1 -> #0.2 Stream #0.0 -> #0.3 [mpeg1video @ 0x82ec9fc]removing common factors from framerate [mpeg1video @ 0x82ec9fc]MPEG1/2 does not support 3/1 fps Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height Is there anything I can do to resolve this problem? The ffserver.conf looks like this, it's more or less the sample config exactly: # Port on which the server is listening. You must select a different # port from your standard http web server if it is running on the same # computer. Port 8090 # Address on which the server is bound. Only useful if you have # several network interfaces. BindAddress 0.0.0.0 # Number of simultaneous requests that can be handled. Since FFServer # is very fast, it is more likely that you will want to leave this high # and use MaxBandwidth, below. MaxClients 1000 # This the maximum amount of kbit/sec that you are prepared to # consume when streaming to clients MaxBandwidth 1000 # Access Log file (uses standard Apache log file format) # '-' is the standard output CustomLog - # Suppress that if you want to launch ffserver as a daemon NoDaemon ################################################################## # Definition of the live feeds. Each live feed contains one video # and/or audio sequence coming from an ffmpeg encoder or another # ffserver. This sequence may be encoded simultaneously with several # codecs at several resolutions. <Feed feed1.ffm> # You must use 'ffmpeg' to send a live feed to ffserver. In this # example, you can type: # # ffmpeg http://localhost:8090/feed1.ffm # ffserver can also do time shifting. It means that it can stream any # previously recorded live stream. The request should contain: # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify # a path where the feed is stored on disk. You also specify the # maximum size of the feed (100M bytes here). Default: # File=/tmp/feed_name.ffm FileMaxSize=5M File /tmp/feed1.ffm FileMaxSize 10M # You could specify # ReadOnlyFile /saved/specialvideo.ffm # This marks the file as readonly and it will not be deleted or updated # Specify launch in order to start ffmpeg automatically #Launch # Only allow connections from localhost to the feed ACL allow 127.0.0.1 </Feed> ################################################################## # Now you can define each stream which will be generated from the # original audio and video stream. Each format has a filename (here # 'test1.mpg'). FFServer will send this stream when answering a # request containing this filename. <Stream test1.mpg> # coming from live feed 'feed1' Feed feed1.ffm # Format of the stream : you can choose among: # mpeg : MPEG1 multiplexed video and audio # mpegvideo : only MPEG1 video # mp2 : MPEG2 audio (use AudioCodec to select layer 2 and 3 codec) # ogg : Ogg format (Vorbis audio codec) # rm : Real Networks compatible stream. Multiplexed audio and video. # ra : Real Networks compatible stream. Audio only. # mpjpeg : Multipart JPEG (works with Netscape without any plugin) # jpeg : Generate a single JPEG image. # asf : ASF compatible streaming (Windows Media Player format). # swf : Macromedia flash(tm) compatible stream # avi : AVI format (open divx video, mpeg audio sound) # master : special ffmpeg stream used to duplicate a server Format mpeg # Bitrate for the audio stream. Codecs usually support only a few # different bitrates. AudioBitRate 32 # Number of audio channels : 1 = mono, 2 = stereo AudioChannels 1 # Sampling frequency for audio. When using low bitrates, you should # lower this frequency to 22050 or 11025. The supported frequencies # depend on the selected audio codec. AudioSampleRate 44100 # Bitrate for the video stream. VideoBitRate 64 # Ratecontrol buffer size VideoBufferSize 40 # Number of frames per second VideoFrameRate 3 # Size of the video frame : WxH (default: 160x128) # The following abbreviation are defined : sqcif, qcif, cif, 4cif VideoSize 160x128 # transmit only intra frames (useful for low bitrates, but kills frame rate) #VideoIntraOnly # If non intra only, an intra frame is transmitted every VideoGopSize # frames Video synchronization can only begin at an I frames. VideoGopSize 12 # More MPEG4 parameters # VideoHighQuality # Video4MotionVector # Choose your codecs: #AudioCodec mp2 #VideoCodec mpeg1video # Suppress audio #NoAudio # Suppress video #NoVideo #VideoQMin 3 #VideoQMax 31 # Set this to the number of seconds backwards in time to start. Note that # most players will buffer 5-10 seconds of video, and also you need to allow # for a key frame to appear in the data stream. #PreRoll 15 # ACL: # You can allow ranges of addresses (or single addresses) #ACL ALLOW <first address> # You can deny ranges of addresses (or single addresses) #ACL DENY <first address> # You can repeat the ACL allow/deny as often as you like. It is on a per # stream basis. The first match defines the action. If there are no matches, # then the default is the inverse of the last ACL statement. # # Thus 'ACL allow localhost' only allows access from localhost. # 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and # allow everybody else. </Stream> ################################################################## # Example streams # Multipart JPEG #<Stream test.mjpg> #Feed feed1.ffm #Format mpjpeg #VideoFrameRate 2 #VideoIntraOnly #NoAudio #</Stream> # Single JPEG #<Stream test.jpg> #Feed feed1.ffm #Format jpeg #VideoFrameRate 2 #VideoIntraOnly ##VideoSize 352x240 #NoAudio #</Stream> # Flash #<Stream test.swf> #Feed feed1.ffm #Format swf #VideoFrameRate 2 #VideoIntraOnly #NoAudio #</Stream> # ASF compatible <Stream test.asf> Feed feed1.ffm Format asf VideoFrameRate 15 VideoSize 352x240 VideoBitRate 256 VideoBufferSize 40 VideoGopSize 30 AudioBitRate 64 StartSendOnKey </Stream> # MP3 audio #<Stream test.mp3> #Feed feed1.ffm #Format mp2 #AudioCodec mp3 #AudioBitRate 64 #AudioChannels 1 #AudioSampleRate 44100 #NoVideo #</Stream> # Ogg Vorbis audio #<Stream test.ogg> #Feed feed1.ffm #Title "Stream title" #AudioBitRate 64 #AudioChannels 2 #AudioSampleRate 44100 #NoVideo #</Stream> # Real with audio only at 32 kbits #<Stream test.ra> #Feed feed1.ffm #Format rm #AudioBitRate 32 #NoVideo #NoAudio #</Stream> # Real with audio and video at 64 kbits #<Stream test.rm> #Feed feed1.ffm #Format rm #AudioBitRate 32 #VideoBitRate 128 #VideoFrameRate 25 #VideoGopSize 25 #NoAudio #</Stream> ################################################################## # A stream coming from a file : you only need to set the input # filename and optionnally a new format. Supported conversions: # avi -> asf #<Stream file.rm> #File "/usr/local/httpd/htdocs/tlive.rm" #NoAudio #</Stream> #<Stream file.asf> #File "/usr/local/httpd/htdocs/test.asf" #NoAudio #Author "Me" #Copyright "Super MegaCorp" #Title "Test stream from disk" #Comment "Test comment" #</Stream> ################################################################## # RTSP examples # # You can access to this stream with the RTSP URL: # rtsp://localhost:5454/test1-rtsp.mpg # # A non standard RTSP redirector is also created. Its URL is: # http://localhost:8090/test1-rtsp.rtsp #<Stream test1-rtsp.mpg> #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" #</Stream> ################################################################## # SDP/multicast examples # # If you want to send your stream in multicast, you must set the # multicast address with MulticastAddress. The port and the TTL can # also be set. # # An SDP file is automatically generated by ffserver by adding the # 'sdp' extension to the stream name (here # http://localhost:8090/test1-sdp.sdp) You should usually give this # file to your player to play the stream. # # The 'NoLoop' option can be used to avoid looping when the stream is # terminated. #<Stream test1-sdp.mpg> #Format rtp #File "/usr/local/httpd/htdocs/test1.mpg" #MulticastAddress 224.124.0.1 #MulticastPort 5000 #MulticastTTL 16 #NoLoop #</Stream> ################################################################## # Special streams # Server status <Stream stat.html> Format status # Only allow local people to get to the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 #FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico </Stream> # Redirect index.html to the appropriate site <Redirect index.html> URL http://ffmpeg.sourceforge.net/ </Redirect> Thank you, -- Slava Shirokov slava at mutatorr.net

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decoding quicktime 6 audio
by gavinj＠fnarg.net 14 May '05

14 May '05

Hello, I found that I can't decode quicktime 6 audio. [gavinj at www dvd]$ ffmpeg -i /home/project/video/mov/switzler084d_dl.mov -acodec mp2 -vn /tmp/switzler084d_dl.mp2 ffmpeg version 0.4.9-pre1, build 4752, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-mp3lame built on Apr 15 2005 20:11:42, gcc: 3.3.3 20040412 (Red Hat Linux 3.3.3-7) Input #0, mov,mp4,m4a,3gp,3g2, from '/home/project/video/mov/switzler084d_dl.mov': Duration: 00:01:47.6, start: 0.000000, bitrate: 667 kb/s Stream #0.0: Video: svq3, 480x206, 15.01 fps Stream #0.1: Audio: 0x324d4451, 48000 Hz, stereo Output #0, mp2, to '/tmp/switzler084d_dl.mp2': Stream #0.0: Audio: mp2, 48000 Hz, stereo, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Unsupported codec (id=0) for input stream #0.1 (I think this is QT6, it's the star wars trailer. The fourcc is QDM2). I prolly need to reconfigure ffmpeg with some extra --enables. But I'm not sure which ones. Anyone? Gavin

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4:2:0 to 4:2:2 conversion
by mangeshsk＠gmail.com 14 May '05

14 May '05

Hello, What should be the change in following command to convert 4:2:0 planar to 4:2:2 packed? ffmpeg -s 720x576 -f rawvideo -pix_fmt yuv420p -i src.yuv -s 720x576 -f rawvideo -pix_fmt yuv422p dst.yuv. Thanks Mangesh

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[Ffmpeg-user-old] ffmpeg does not compile with x264
by alvarez＠ac.upc.edu 13 May '05

13 May '05

hi I trying to compile the lastest cvs ffmpeg with x264 support on a linux pentium-m machine but I obtain some errors: I am using x264 core:24 svn-228 the configuration of ffmpeg is like that: ./configure --enable-x264 --enable-xvid --enable-gpl --disable-ffserver and the compile error is (using gcc version 3.3.5 (Debian 1:3.3.5-8ubuntu2)) $make make -C libavcodec all make[1]: Entering directory `/home/alvarez/multimedia/h264/tools/ffmpeg/libavcodec' gcc -O3 -g -Wall -Wno-switch -DHAVE_AV_CONFIG_H -I.. -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_GNU_SOURCE -c -o x264.o x264.c x264.c: In function `X264_init': x264.c:136: error: structure has no member named `i_rc_buffer_size' It seems that there is a change in the X264Context structure from the last version of X264... bye Mauricio ------------------------------------------------------- This SF.Net email is sponsored by Oracle Space Sweepstakes Want to be the first software developer in space? Enter now for the Oracle Space Sweepstakes! http://ads.osdn.com/?ad_id=7393&alloc_id=16281&op=click _______________________________________________ Ffmpeg-user mailing list Ffmpeg-user at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/ffmpeg-user

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