March 2010 - ffmpeg-user - lists.ffmpeg.org

MPEG transport stream with h264 video and aac audio
by visuall＠gmail.com 12 Mar '10

12 Mar '10

Hello, I am trying to encapsulate h264 video and aac audio in MPEG transport stream container using ffmpeg via re-encoding of various input media files. The encoding process runs OK with progress in console and so on but at the end the output is damaged: no audio when playing, video starts to stutter after the first several seconds. At the same time when doing the above but producing h264 video and aac audio in mp4 container all is OK and the output sounds fine when played back. Example command lines I use: ffmpeg -i g:\testmedia\weddingdate.wmv -f mp4 -acodec aac -ar 48000 -ab 64k -s 320?240 -vcodec libx264 -b 96k -flags +loop -cmp +chroma -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -async 2 g:\testmedia\output.mp4 This one works OK! ffmpeg -i g:\testmedia\weddingdate.wmv -f mpegts -acodec aac -ar 48000 -ab 64k -s 320?240 -vcodec libx264 -b 96k -flags +loop -cmp +chroma -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -async 2 g:\testmedia\output.mp4 This one causes the playback to stutter. For the tests I use ffmpeg revision 21736 and latest x264 library. Is there something specific that should be added to command line or the problem is elsewhere? Regards, Nick

5 8

How to use ffmpeg to encode -vcodec libxvid and send the rtp protocol
by shawnlrs＠gmail.com 12 Mar '10

12 Mar '10

Hi all, who can help me to use ffmpeg to encode -vcodec libxvid and send the rtp protocol, now i test use this command it's work, -vcodec mpeg4 -f rtp ffmpeg.exe -f vfwcap -r 5 -i 0 -vcodec mpeg4 -s 640x480 -r 5 -b 400kb -f rtp rtp://192.168.209.224:5004 and it's work too, -vcodec libxvid ffmpeg -v 9 -f vfwcap -r 5 -i 0 -y -vcodec libxvid -s 640x480 -aspect 4:3 -b 400kb rtp://192.168.209.224:5004/?pkt_size=1400;file=.avi but if i want to encode -vcodec libxvid and send the rtp protocol is wrong. ffmpeg.exe -f vfwcap -r 5 -i 0 -y -vcodec libxvid -s 640x480 -aspect 4:3 -b 400kb -f rtp rtp://192.168.209.224:5004/?pkt_size=1400;file=.avi there are some error message ffmepg version SVN-r22426, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 10 2010 06:11:03 with gcc 4.4.2 configuration: --enable-shared --disable-static --enable-memalign-hack --cross -prefix=i686-mingw32- --cc=ccache-i686-mingw32-gcc --arch=i686 --target-os=mingw 32 --enable-runtime-cpudetect --enable-avisynth --enable-gpl --enable-version3 - -enable-bzlib --enable-libgsm --enable-libfaad --enable-pthreads --enable-libvor bis --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpe g --enable-libxvid --enable-libschroedinger --enable-libx264 --enable-libopencor e_amrwb --enable-libopencore_amrnb libavutil 50.11. 0 / 50.11. 0 libavcodec 52.58. 0 / 52.58. 0 libavformat 52.55. 0 / 52.55. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 [vfwcap @ 004edb80]Estimating duration from bitrate, this may be inaccurate Input #0, vfwcap, from '0': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0: Video: rawvideo, bgr24, 640x480, 5 tbr, 1k tbn, 5 tbc [rtp @ 00f88cf0]Unsupported codec 3f Output #0, rtp, to 'rtp://192.168.209.224:5004/?pkt_size=1400;file=.avi': Metadata: encoder : Lavf52.55.0 Stream #0.0: Video: libxvid, yuv420p, 640x480 [PAR 1:1 DAR 4:3], q=2-31, 400 kb/s, 90k tbn, 5 tbc Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?)

1 0

compile error libx264 not found (windows)
by alexeybbb＠gmail.com 11 Mar '10

11 Mar '10

Hi, how compile ffmpeg whith --enable-libx264? I do: 1. MSYS c:\msys\1.0\ 2. MinGW(GCC 4) c:\MinGW\ 3. Get and compile x264: ---------------------------------------------------------- cd /c/x264 git clone git://git.videolan.org/x264.git ./ ./configure make make install ---------------------------------------------------------- 4. Get and compile FFmpeg ---------------------------------------------------------- cd /c/ffmpeg git clone git://git.ffmpeg.org/ffmpeg/ ./ git clone git://git.ffmpeg.org/libswscale/ $ ./configure --enable-libx264 --enable-gpl --enable-memalign-hack --extra-cflags='-I/usr/local/include' --extra-ldflags='-L/usr/local/lib' ERROR: libx264 not found ---------------------------------------------------------- What I do wrong? (config.err in attach)

2 2

Help Need to provide transitions between videos
by arinhere＠gmail.com 11 Mar '10

11 Mar '10

Hello, In my application I want to use various kind of transitions between 2 videos? I have FFMPEG VHOOK [IMLIB2] build FFMPEG. I successfully implemented effects on videos. But unable to provide transition effects between them. Could you please help me out? Thanks - Arin -- ----------------------- Thanks & Regards -- Arin

1 0

need help on DV video decoding
by pcuser.mails＠gmail.com 11 Mar '10

11 Mar '10

If i get a dv file(DIF sequences) wrapped in a avi/quicktime/mxf container. can ffmpeg demuxer or any other commercial/opensource demuxer produce orginal DIF sequence without any modification to the original DIF sequence. My requirement is to develop DV decoder for IEC 61834 & SMPTE 341M conformant bitstreams. But mostly, I will be getting DIF sequences(dv file) from avi/mxf file after demuxing. so the question is, does both mxf and avi container produce same DIF sequences after demuxing (or) DIF sequences depends on demuxer? My question may be out of scope of DV decoder,but I need to define the capability of my decoder. any help in this would be greatly appreciated! > On 3/9/10, Dave Rice <dave at avpreserve.com> wrote: >> In DV (speaking particularly of the .dv file) each DIF sequence has its >> own >> header and there is no file specific file header. If you split a .dv file >> every 120000 or 144000 bytes (NTSC or PAL), you'd have a bunch of valid >> little dv streams one per frame. >> For closer inspection of the DV frame I usually run >> xxd -ps -c 80 file.dv >> in order to generate an output that aligns with the DIF block structure >> that >> aligns with SMPTE 314m. >> If it helps, I wrote some articles on DV at >> http://www.avpreserve.com/dvanalyzer/ along with an analysis tool for DV >> media. >> Dave >> >> On Mar 9, 2010, at 2:19 AM, pcuser p wrote: >> >>> Dave, >>> Thanks for reply.I've gone through dv video standards (SMPTE-314 and IEC >>> 61834). >>> Both of the documents explain about how video data compression is done >>> (DCT,VLC,etc.,) >>> but none defines about file format of dv video file. >>> >>> For example if i have a file with .dv extension,there must be some >>> header which identifies that it's a dv file and also info about where >>> AAUX,VAUX, & DIFdata in the dv file. >>> This info is not provided in the standard doc. >>> >>> PS: >>> I've read through IEC 61834-2 and SMPTE 314M document. >>> Is this info is available in IEC 61834-4 document? >>> >>> expecting some help & suggestions.. >>> >>> On 3/9/10, pcuser p <pcuser.mails at gmail.com> wrote: >>>> Dave, >>>> Thanks for reply.I've gone through dv video standards (SMPTE-314 and >>>> IEC >>>> 61834). >>>> Both of the documents explain about how video data compression is done >>>> (DCT,VLC,etc.,) >>>> but none defines about file format of dv video file. >>>> >>>> For example if i have a file with .dv extension,there must be some >>>> header which identifies that it's a dv file and also info about where >>>> AAUX,VAUX, & DIFdata in the dv file. >>>> This info is not provided in the standard doc. >>>> >>>> PS: >>>> I've read through IEC 61834-2 and SMPTE 314M document. >>>> Is this info is available in IEC 61834-4 document? >>>> >>>> expecting some help & suggestions.. >>>> >>>> >>>> >>>> >>>> On 3/5/10, Dave Rice <dave at avpreserve.com> wrote: >>>>> On Mar 5, 2010, at 10:42 AM, Walter Werner wrote: >>>>> >>>>>> >>>>>> >>>>>> Am 05.03.2010 09:54, schrieb pcuser p: >>>>>>> I am trying to write a wrapper for dv dec(dv,dvcpro25/50) by reading >>>>>>> input from a dv encoded video file.I have some progressive encoded >>>>>>> dv >>>>>>> video files in my pc. >>>>>>> >>>>>>> But I don't know how to interpret the (.dv) file,I dont have the >>>>>>> file >>>>>>> format spec for dv. >>>>>>> can somebody provide me insight on dv file format, or atleast >>>>>>> provide >>>>>>> me the link for the file format spec of dv. >>>>>>> >>>>>>> I have seen that ffmpeg can decode all types of dv files. >>>>>>> "on what file format ffmpeg dv decoder is implemented?how it >>>>>>> interprets dv files and extract header and metadata?" >>>>>>> >>>>>>> If dv is wrapped in avi container,will there be any change in dv >>>>>>> headers and meta data. >>>>>>> can we extract .dv file from avi container without any change to >>>>>>> header and metadata of .dv files >>>>>>> >>>>>>> PS:I have looked into SMPTE314M standard,it has info about data >>>>>>> compression but not on dv file format >>>>>>> >>>>>>> I really got struck at the dv decoder wrapper for sometime without >>>>>>> have dv file format specification... >>>>>>> Any help would be greatly appreciated! >>>>>>> >>>>>>> Regards, >>>>>>> RJ >>>>>>> >>>>>> Why don't you use ffmpeg to convert to raw video and use the output >>>>>> (eventuelly through a pipe) in your program? >>>>>> There is some information on dv structure (also within avi) in >>>>>> wikipedia, >>>>>> with links to the details >>>>>> Walter. >>>>> >>>>> I don't think this would work. When I use ffmpeg to convert a >>>>> Quicktime >>>>> wrapped file to a dv stream using -vcodec copy, the resulting dv does >>>>> not >>>>> contain the VAUX and AAUX metadata of the stream wrapped in the >>>>> original >>>>> file. >>>>> For instance with >>>>> ffmpeg -i test.mov -vcodec copy test.dv >>>>> the resulting dv stream does not have the embedded recording >>>>> timestamps, >>>>> recording start and stop flags, or closed captioning text of the >>>>> original. >>>>> ffmpeg output below, >>>>> >>>>> David Rice >>>>> AudioVisual Preservation Solutions >>>>> 350 7th Avenue, Room 1603 >>>>> New York, NY 10001 >>>>> >>>>> office: 212-564-2140 >>>>> cell: 347-213-3517 >>>>> dave at avpreserve.com >>>>> www.avpreserve.com >>>>> >>>>> >>>>> u815:A7600 daverice$ ffmpeg -i test.mov -vcodec copy test.dv >>>>> FFmpeg version FFmbc-0.3, Copyright (c) 2000-2009 Fabrice Bellard, et >>>>> al. >>>>> built on Feb 15 2010 16:01:14 with gcc 4.2.1 (Apple Inc. build 5646) >>>>> (dot >>>>> 1) >>>>> configuration: --enable-libopenjpeg >>>>> libavutil 50. 4. 0 / 50. 4. 0 >>>>> libavcodec 52.41. 0 / 52.41. 0 >>>>> libavformat 52.39. 2 / 52.39. 2 >>>>> libavdevice 52. 2. 0 / 52. 2. 0 >>>>> libswscale 0. 7. 1 / 0. 7. 1 >>>>> [dvvideo @ 0x101014600]AC EOB marker is absent pos=66 >>>>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': >>>>> Metadata: >>>>> major_brand: qt >>>>> minor_version: 537199360 >>>>> compatible_brands: qt >>>>> year(eng): Thursday, May 25, 2017 10:26:25 PM >>>>> Duration: 00:00:30.76, start: 0.000000, bitrate: 30308 kb/s >>>>> Stream #0.0(eng): Video: dvvideo, yuv411p, 720x480, tbc 1001/30000, >>>>> 28771 kb/s, PAR 10:11 DAR 15:11, fps 29.970, tbf 1/2997 >>>>> Stream #0.1(eng): Data: tmcd / 0x64636D74, 0 kb/s >>>>> Stream #0.2(eng): Audio: pcm_s16be, 48000 Hz, 2 channels, s16, 1536 >>>>> kb/s >>>>> Output #0, dv, to 'test.dv': >>>>> Stream #0.0(eng): Video: dvvideo, yuv411p, 720x480 [PAR 10:11 DAR >>>>> 15:11], tbc 1001/30000, q=2-31, 28771 kb/s, tbf 1/90000 >>>>> Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 >>>>> kb/s >>>>> Stream mapping: >>>>> Stream #0.0 -> #0.0 >>>>> Stream #0.2 -> #0.1 >>>>> Press [q] to stop encoding >>>>> frame= 922 fps= 97 q=-1.0 Lsize= 107930kB time=30.76 >>>>> bitrate=28740.7kbits/s >>>>> video:108047kB audio:5768kB global headers:0kB muxing overhead >>>>> -5.170946% >>>>> >>>>> >>>>> >>>>>>> _______________________________________________ >>>>>>> ffmpeg-user mailing list >>>>>>> ffmpeg-user at mplayerhq.hu >>>>>>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> ffmpeg-user mailing list >>>>>> ffmpeg-user at mplayerhq.hu >>>>>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >>>>> >>>>> _______________________________________________ >>>>> ffmpeg-user mailing list >>>>> ffmpeg-user at mplayerhq.hu >>>>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >>>>> >>>> >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at mplayerhq.hu >>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at mplayerhq.hu >> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >> >

2 2

encoding video from a scratched DVD-RAM
by tv＠fuzzy.cz 11 Mar '10

11 Mar '10

Hi everyone, I do have a scratched DVD-RAM with video on it (single scratch, so it's not heavily scratched). When I play it (e.g. in mplayer) the corruption is noticeable for about one minute (random blocks, etc.), but when I try to recode it using ffmpeg - e.g. like this $ ffmpeg -i /mnt/cdrom/DVD_RTAV/VR_MOVIE.VRO -deinterlace \ -s 600x450 -sws_flags bicubic -vcodec libxvid \ -acodec libmp3lame -vb 500000 -ab 128000 -ar 44100 \ ~/recoded.avi; the corruption is much longer (several minutes instead of one), which is unacceptable. Are there any options I've missed in the man page to fix this? I've tried to recode it using mencoder, and the results are much better (equal to what I've seen when playing the VRO file in mplayer), but I have other problems with mencoder (audio out of sync). thanks for any help Tomas

2 1

Webcam access from remote server
by slyttleton＠mac.com 11 Mar '10

11 Mar '10

I have been trying to figure this out for some time and thought I would throw it out here for the intelligentsia. I have a remote server where my web site is run and it is a Linux server with ffmpeg and ffserver installed. I am looking for a way to have the server pull video from my local webcam and have it streamed to my web site. The issue is that the ffmpeg commands I have found look for a local webcam (/dev/video0). However, the webcam is not on my server but rather on my Mac at home. Can this be done? I cannot find a way to have ffmpeg look to my Mac at home for the video source. Any help is greatly appreciated. ------------------------------- Stephen Lyttleton Sent from my iPhone

2 1

svn 22406: Missing dependency for muxer: mpegts and for decoder: twinvq
by tfoucu＠gmail.com 10 Mar '10

10 Mar '10

Using SVN 22406: ./configure --disable-demuxers --disable-muxers --disable-decoders --disable-encoders --disable-parsers --disable-indevs --disable-outdevs --disable-bsfs --disable-doc --disable-ffplay --disable-ffserver --enable-muxer=mpegts make Error: /ffmpeg_svn/ffmpeg/libavformat/libavformat.a(adtsenc.o): In function `ff_adts_decode_extradata': /ffmpeg_svn/ffmpeg/libavformat/adtsenc.c:63: undefined reference to `ff_copy_pce_data' ------------------------------------------------------------------- ./configure --disable-demuxers --disable-muxers --disable-decoders --disable-encoders --disable-parsers --disable-indevs --disable-outdevs --disable-bsfs --disable-doc --disable-ffplay --disable-ffserver --enable-decoder=twinvq make Error: /ffmpeg_svn/ffmpeg/libavcodec/libavcodec.a(lsp.o): In function `ff_acelp_lsf2lsp': /ffmpeg_svn/ffmpeg/libavcodec/lsp.c:65: undefined reference to `ff_cos' -------------------------------------------------------------------------- If you want, i can open a bug, just not sure to whom I should assign them. Thanks.

1 0

DTS issue!!!help plz
by faizan.ahmed39＠gmail.com 10 Mar '10

10 Mar '10

i have worked thru the structures AVFormatContext,streams and packets but i am not completely familiar with pkt->dts and st->cur_dts i need to know why is that st->cur_dts get larger than pkt->dts and so compute_pkt_fields2() gives return -1 and the packet doesn't get written no matter what the st->codec_type is. i have been working on it quite some time and i could sombody make me understand and hopefully give me a fix of this issue this is where the error generates from if(st->cur_dts && st->cur_dts != AV_NOPTS_VALUE && st->cur_dts >= pkt->dts){ av_log(s, AV_LOG_ERROR,"st:%d error, non monotone timestamps %"PRId64" >= %"PRId64"\n",st->index, st->cur_dts, pkt->dts); return -1; and thaats why my frame/pkt doesn't get written thru av_interleaved_write_frame(); -- View this message in context: http://n4.nabble.com/DTS-issue-help-plz-tp1587411p1587411.html Sent from the FFmpeg-users mailing list archive at Nabble.com.

1 0

Setup problems - Streaming over RTP with webcam video encodec in H264
by smutje.sanji＠googlemail.com 10 Mar '10

10 Mar '10

Hello, any hints about the setup would be great. My aim is to stream with ffserver webcam video which is delivered and encodey by ffmpeg. Iam setting up ffserver with this config: Port 8090 BindAddress 0.0.0.0 RTSPPort 5004 RTSPBindAddress 0.0.0.0 MaxClients 10 MaxBandwidth 40000 NoDaemon <Feed feed1.ffm> File /tmp/feed1.ffm FileMaxSize 5M ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Feed> <stream webcamvideo.mp4> Feed feed1.ffm Format rtp VideoCodec libx264 VideoFrameRate 15 VideoBufferSize 40 VideoBitRate 500 VideoSize 640x480 # which value? VideoGopSize 15 NoAudio # i had problems with videostreaming with file. player said that header were corrupted, so i used this config, but it doesnt help. i still let it in. AVOptionVideo flags +global_header </Stream> <Stream stat.html> Format status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> <Redirect index.html> URL http://www.ffmpeg.org/ </Redirect> Starting the server with, which is working ffserver -d -f ffserverwebcam.conf Now I want to use ffmpeg to deliver the webcamvideo with ffmpeg -f video4linux2 -s 640x480 -r 15 -i /dev/video0 -an -vcodec libx264 -vpre default -bf 5 http://localhost:8090/feed1.ffm ffmpeg doesnt encode the video and ends with this output FFmpeg version SVN-r22390, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 10 2010 11:30:16 with gcc 4.4.1 configuration: --prefix=/usr/local --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libmp3lame --enable-libvorbis --enable-libschroedinger --enable-libspeex --enable-libx264 --enable-libxvid --enable-zlib --enable-pthreads --enable-postproc --enable-avfilter --enable-avfilter-lavf --enable-gpl --enable-version3 --enable-nonfree --enable-x11grab --arch=x86_64 --disable-stripping --disable-optimizations --enable-debug libavutil 50.11. 0 / 50.11. 0 libavcodec 52.58. 0 / 52.58. 0 libavformat 52.55. 0 / 52.55. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.17. 0 / 1.17. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [video4linux2 @ 0x2800050][3]Capabilities: 5000001 [video4linux2 @ 0x2800050]Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2, from '/dev/video0': Duration: N/A, start: 1268222505.539241, bitrate: 55296 kb/s Stream #0.0: Video: rawvideo, yuv420p, 640x480, 55296 kb/s, 15 tbr, 1000k tbn, 15 tbc [libx264 @ 0x2807500]broken ffmpeg default settings detected [libx264 @ 0x2807500]use an encoding preset (vpre) Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Stream #0.0: Video: libx264, yuv420p, 640x480, q=3-31, 500 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.0 -> #0.0 Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height ioctl(VIDIOC_QBUF) Thats it... ffmpeg is always telling me [libx264 @ 0x2807500]broken ffmpeg default settings detected [libx264 @ 0x2807500]use an encoding preset (vpre) I put all libx264-*.ffpreset to [home]/[user]/.ffmpeg so ffmpeg should find the presets. He will lookup to .ffmpeg-folder concatenate the string of the ffmpeg params -vcodec with -vpre and with ".ffpreset", so it should find a "libx264-default.ffpreset". But its still complaining that it doesnt find it. But it runs, so I think its ok ?! The next message is more problematic Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height ioctl(VIDIOC_QBUF Any hints and help would be appreciated... Thank you, Ali Ayhan D.

1 0

avfilter patching errors
by alexeybbb＠gmail.com 10 Mar '10

10 Mar '10

Hi, I try to build ffmpeg with "--enable-avfilter". What I do: 1. Get libavfilter from?svn://svn.ffmpeg.org/soc/libavfilter 2. Execute the checkout.sh script (as write in readme) And have patching errors: ---------------------------------------------------------------------------------- Updated to revision 30856. patching ffmpeg patching file ffmpeg.c Hunk #1 FAILED at 42. Hunk #2 FAILED at 156. Hunk #3 FAILED at 318. Hunk #4 FAILED at 340. Hunk #5 FAILED at 662. Hunk #6 FAILED at 1098. Hunk #7 FAILED at 1146. Hunk #8 FAILED at 1160. Hunk #9 FAILED at 1187. Hunk #10 FAILED at 1235. Hunk #11 FAILED at 1501. Hunk #12 FAILED at 1632. Hunk #13 FAILED at 1662. Hunk #14 FAILED at 1691. Hunk #15 FAILED at 1762. Hunk #16 FAILED at 2209. Hunk #17 FAILED at 2220. Hunk #18 FAILED at 2614. Hunk #19 FAILED at 4209. Hunk #20 FAILED at 4272. 20 out of 20 hunks FAILED -- saving rejects to file ffmpeg.c.rej ---------------------------------------------------------------------------------- What I do wrong?

2 2

Can't seem to pipe from ffmpeg
by robertk54＠aol.com 10 Mar '10

10 Mar '10

Well, I've been struggling with my application for a number of days now. The application is essentially a proxy server which intercepts media and attempts to pipe the media through ffmpeg. The application is one process, when media is detected it spawns a new process for ffmpeg using popen. If the input to ffmpeg is a named pipe and the output is a file, thethe application is able to write to the fifo, the ffmpeg process is able to read from the fifo, process the stream, and write the modified stream to file just fine. However, if the output from ffmpeg is a pipe (using '-' for stdout) or a named pipe (using the same file name as before but the application calls mkfifo prior to the popen call), I am unable to read from the pipe to obtain the data. I am spawning a reader thread to handle the reading of either pipe. In the case where the output is '-', the reader thread's call to fread does not block but never reads any data. In the case where I am using a named pipe, the fopen of the fifo for reading (done at the beginning of my reader thread) appears to block. It's as if ffmpeg is unable to open the pipes for writing. Any suggestions??

2 3

Problem with ffmpeg compiled from svn or git
by abu_hurayrah＠hidayahonline.org 09 Mar '10

09 Mar '10

I've recently tried compiling ffmpeg from git as well as from svn, and I am experiencing the following errors whenever I attempt to run any ffmpeg executable: > [basil.gohar at escapdv07 Video]$ ffmpeg -i balcony-2-vga.ogv > ffmpeg: /usr/lib/i686/libpostproc.so.51: no version information > available (required by ffmpeg) > ffmpeg: /usr/lib/i686/libavdevice.so.52: no version information > available (required by ffmpeg) > ffmpeg: /usr/lib/i686/libavcodec.so.52: no version information > available (required by ffmpeg) > ffmpeg: /usr/lib/i686/libswscale.so.0: no version information > available (required by ffmpeg) > ffmpeg: /usr/lib/i686/libavformat.so.52: no version information > available (required by ffmpeg) > ffmpeg: /usr/lib/i686/libavcodec.so.52: no version information > available (required by /usr/local/lib/libavfilter.so.1) > ffmpeg: /usr/lib/i686/libswscale.so.0: no version information > available (required by /usr/local/lib/libavfilter.so.1) > ffmpeg: relocation error: /usr/local/lib/libavfilter.so.1: symbol > sws_isSupportedInput, version LIBSWSCALE_0 not defined in file > libswscale.so.0 with link time reference My configure looks like this: > ./configure --prefix=/usr/local --enable-shared --enable-gpl > --enable-version3 --enable-nonfree --enable-postproc --enable-avfilter > --enable-avfilter-lavf --enable-pthreads --enable-x11grab > --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad > --enable-libfaadbin --enable-libgsm --enable-libmp3lame > --enable-libopenjpeg --enable-libschroedinger --enable-libspeex > --enable-libtheora --enable-libvorbis --enable-libx264 > --enable-libxvid --enable-zlib --extra-ldflags=-L/usr/local/lib > --extra-cflags=-I/usr/local/include I was able to configure and make without any errors. Can anyone identify why I am getting the above-mentioned error? Thanks in advance!

1 0

need help on DV video decoding
by pcuser.mails＠gmail.com 09 Mar '10

09 Mar '10

Dave, Thanks for reply.I've gone through dv video standards (SMPTE-314 and IEC 61834). Both of the documents explain about how video data compression is done (DCT,VLC,etc.,) but none defines about file format of dv video file. For example if i have a file with .dv extension,there must be some header which identifies that it's a dv file and also info about where AAUX,VAUX, & DIFdata in the dv file. This info is not provided in the standard doc. PS: I've read through IEC 61834-2 and SMPTE 314M document. Is this info is available in IEC 61834-4 document? expecting some help & suggestions.. On 3/9/10, pcuser p <pcuser.mails at gmail.com> wrote: > Dave, > Thanks for reply.I've gone through dv video standards (SMPTE-314 and IEC > 61834). > Both of the documents explain about how video data compression is done > (DCT,VLC,etc.,) > but none defines about file format of dv video file. > > For example if i have a file with .dv extension,there must be some > header which identifies that it's a dv file and also info about where > AAUX,VAUX, & DIFdata in the dv file. > This info is not provided in the standard doc. > > PS: > I've read through IEC 61834-2 and SMPTE 314M document. > Is this info is available in IEC 61834-4 document? > > expecting some help & suggestions.. > > > > > On 3/5/10, Dave Rice <dave at avpreserve.com> wrote: >> On Mar 5, 2010, at 10:42 AM, Walter Werner wrote: >> >>> >>> >>> Am 05.03.2010 09:54, schrieb pcuser p: >>>> I am trying to write a wrapper for dv dec(dv,dvcpro25/50) by reading >>>> input from a dv encoded video file.I have some progressive encoded dv >>>> video files in my pc. >>>> >>>> But I don't know how to interpret the (.dv) file,I dont have the file >>>> format spec for dv. >>>> can somebody provide me insight on dv file format, or atleast provide >>>> me the link for the file format spec of dv. >>>> >>>> I have seen that ffmpeg can decode all types of dv files. >>>> "on what file format ffmpeg dv decoder is implemented?how it >>>> interprets dv files and extract header and metadata?" >>>> >>>> If dv is wrapped in avi container,will there be any change in dv >>>> headers and meta data. >>>> can we extract .dv file from avi container without any change to >>>> header and metadata of .dv files >>>> >>>> PS:I have looked into SMPTE314M standard,it has info about data >>>> compression but not on dv file format >>>> >>>> I really got struck at the dv decoder wrapper for sometime without >>>> have dv file format specification... >>>> Any help would be greatly appreciated! >>>> >>>> Regards, >>>> RJ >>>> >>> Why don't you use ffmpeg to convert to raw video and use the output >>> (eventuelly through a pipe) in your program? >>> There is some information on dv structure (also within avi) in >>> wikipedia, >>> with links to the details >>> Walter. >> >> I don't think this would work. When I use ffmpeg to convert a Quicktime >> wrapped file to a dv stream using -vcodec copy, the resulting dv does not >> contain the VAUX and AAUX metadata of the stream wrapped in the original >> file. >> For instance with >> ffmpeg -i test.mov -vcodec copy test.dv >> the resulting dv stream does not have the embedded recording timestamps, >> recording start and stop flags, or closed captioning text of the >> original. >> ffmpeg output below, >> >> David Rice >> AudioVisual Preservation Solutions >> 350 7th Avenue, Room 1603 >> New York, NY 10001 >> >> office: 212-564-2140 >> cell: 347-213-3517 >> dave at avpreserve.com >> www.avpreserve.com >> >> >> u815:A7600 daverice$ ffmpeg -i test.mov -vcodec copy test.dv >> FFmpeg version FFmbc-0.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. >> built on Feb 15 2010 16:01:14 with gcc 4.2.1 (Apple Inc. build 5646) >> (dot >> 1) >> configuration: --enable-libopenjpeg >> libavutil 50. 4. 0 / 50. 4. 0 >> libavcodec 52.41. 0 / 52.41. 0 >> libavformat 52.39. 2 / 52.39. 2 >> libavdevice 52. 2. 0 / 52. 2. 0 >> libswscale 0. 7. 1 / 0. 7. 1 >> [dvvideo @ 0x101014600]AC EOB marker is absent pos=66 >> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': >> Metadata: >> major_brand: qt >> minor_version: 537199360 >> compatible_brands: qt >> year(eng): Thursday, May 25, 2017 10:26:25 PM >> Duration: 00:00:30.76, start: 0.000000, bitrate: 30308 kb/s >> Stream #0.0(eng): Video: dvvideo, yuv411p, 720x480, tbc 1001/30000, >> 28771 kb/s, PAR 10:11 DAR 15:11, fps 29.970, tbf 1/2997 >> Stream #0.1(eng): Data: tmcd / 0x64636D74, 0 kb/s >> Stream #0.2(eng): Audio: pcm_s16be, 48000 Hz, 2 channels, s16, 1536 >> kb/s >> Output #0, dv, to 'test.dv': >> Stream #0.0(eng): Video: dvvideo, yuv411p, 720x480 [PAR 10:11 DAR >> 15:11], tbc 1001/30000, q=2-31, 28771 kb/s, tbf 1/90000 >> Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 >> kb/s >> Stream mapping: >> Stream #0.0 -> #0.0 >> Stream #0.2 -> #0.1 >> Press [q] to stop encoding >> frame= 922 fps= 97 q=-1.0 Lsize= 107930kB time=30.76 >> bitrate=28740.7kbits/s >> video:108047kB audio:5768kB global headers:0kB muxing overhead -5.170946% >> >> >> >>>> _______________________________________________ >>>> ffmpeg-user mailing list >>>> ffmpeg-user at mplayerhq.hu >>>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >>>> >>>> >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at mplayerhq.hu >>> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at mplayerhq.hu >> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user >> >

2 1

HTTP/RTSP as webcam simulator
by enrico＠pasqualotto.org 09 Mar '10

09 Mar '10

Hi all, is there a way to bridge an HTTP or RTSP stream from IP camera to /dev/video0 (on Linux) or the corresponding in Windows? The goal is to use my IP camera with Skype or similar client. Thanks Pasqu. -- Pasqualotto Enrico cell. +39 3473702179 skype://epasqualotto :: http://www.linkedin.com/in/epasqualotto

1 0

log ffplay error
by m.louvel＠gmail.com 09 Mar '10

09 Mar '10

Hi, I'm wondering if there is a way to log errors encountered while playing a stream with ffplay. I would like to build a script that plays a list of stream and I don't want to spend my whole life in front of a screen to see what happen :) cheers, maxime -- Maxime Louvel +33 6 85 91 05 30 3 rue charreton 38000 Grenoble France

2 2

Buffer underflow during re-encoding
by francesco.andrisani＠acotel.com 09 Mar '10

09 Mar '10

Hi all, this is my first post. I'm a newbe on FFMPEG, but i've a lot of experience with linux. I want explain my problem. I've read a lot of mails from this mailing list, but i don't understand the solution of my problem. I've a webcam (/dev/video0) on my linux ubuntu 9.04. I'm able to view video streaming (mjpeg encoding) fron this webcam, but when i try to re encoding it to flv i've a n "rc buffer underflow". Below my ffserver.conf: Port 8090 BindAddress 0.0.0.0 MaxHTTPConnections 200 MaxClients 10 MaxBandwidth 100000 CustomLog - NoDaemon ################################################################## <Feed feed1.ffm> File /tmp/feed1.ffm File /tmp/feed1.ffm FileMaxSize 5M ACL allow 127.0.0.1 ACL allow 10.8.0.0 10.8.0.255 ACL allow 192.168.2.0 192.168.2.255 </Feed> ################################################################## ################################################################## <Stream test.flv> Feed feed1.ffm Format flv VideoCodec flv VideoFrameRate 15 VideoBufferSize 80000 VideoBitRate 200 VideoQMin 10 VideoQMax 15 VideoSize 352x288 PreRoll 0 Noaudio </Stream> FFSERVER run fine. Then i start ffmpeg taking the video flow directly from /dev/video0, but i've the follow errors: # /usr/local/bin/ffmpeg -r 15 -s 352x288 -f video4linux2 -i /dev/video0 http://localhost:8090/feed1.ffm FFmpeg version SVN-r22168, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 11:21:00 with gcc 4.3.3 configuration: --enable-cross-compile --arch=arm --target-os=linux --cc=gcc --prefix=/usr/local --enable-armv5te --disable-armv6 --disable-mmx --enable-shared --disable-optimizations libavutil 50. 9. 0 / 50. 9. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 [video4linux2 @ 0x45cc0][3]Capabilities: 4000001 [video4linux2 @ 0x45cc0]Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2, from '/dev/video0': Duration: N/A, start: 1267632564.627815, bitrate: 24330 kb/s Stream #0.0: Video: rawvideo, yuyv422, 352x288, 24330 kb/s, 15 tbr, 1000k tbn, 15 tbc Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : Lavf52.54.0 Stream #0.0: Video: flv, yuv420p, 352x288, q=15-15, 200 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [flv @ 0x4c760]rc buffer underflow av_interleaved_write_frame(): Error while opening file If i try to rec a video put it into a file, instead ffserver....it work fine. Can someone help me please. Regards

3 2

-color_primaries -color_trc -colorspace -color_range working?
by dev＠rarevision.com 09 Mar '10

09 Mar '10

I still can't get these options to do anything. Outputting to PNG is the same no matter what combination of values I use for these options: -colorspace 1 -color_primaries 1 -color_trc 1 -color_range 2 Furthermore, I'm not even sure what exactly they change. Here is my hypothesis: -colorspace: YUV or RGB? -color_primaries: 601, 709, etc.? -color_trc: don't know what this is -color_range: 16-235 or 0-255? The problem I am still having is that FFmpeg is clipping the blacks off my input, because it assumes that because it's YUV material that the color range is 16-235. It isn't. It's 0-255, which I admit is non standard but it utilizes the full 8 bit range for color data instead of 16-235, which is limited. I need to be able to explicitly tell FFmpeg that the input is 0-255, so it doesn't ignore the values below 16 and above 235. Any info would be greatly appreciated.

1 0

Making a video of lower frame rate?
by hinacomp＠gmail.com 09 Mar '10

09 Mar '10

Hello everyone, I have a sequence of images and I wish to create a video with relatively low frame rate - like 2 fps or 1 fps. I am typing in the following command in the command line: ffmpeg -i %04d.ppm -r 2 out.avi The video uses only 12 frames and the whole thing is not created correctly. Does this lack something? Thanks

7 11

getting sharper video
by kosta＠brazzers.com 08 Mar '10

08 Mar '10

hello, these are my commands: ffmpeg -i input/%1.mp4 -f mp4 -pass 1 -vcodec libx264 -vpre ./ffpresets/libx264-fastfirstpass.ffpreset -b 8000k -bt 1000k -maxrate 10000k -s 1280x720 -aspect 16:9 -acodec libfaac -ab 192k -ac 2 -threads 0 -y NUL ffmpeg -i input/%1.mp4 -f mp4 -pass 2 -vcodec libx264 -vpre ./ffpresets/libx264-hq.ffpreset -b 8000k -bt 1000k -maxrate 10000k -s 1280x720 -aspect 16:9 -acodec libfaac -ab 192k -ac 2 -threads 0 output/%1_1280_8000k.mp4 is there anything else i can do to make the output better looking? it's a little pixelated right now and i would like to make it look better

1 0

(no subject)
by marcos_ourica＠yahoo.com.br 08 Mar '10

08 Mar '10

My name is Marcos Flavio, I have a problem with ffmpeg. When I convert a video with the extension. MPG to. FLV, the time bar does not work, the video plays but the time bar is stopped. Can someone help me on this? ____________________________________________________________________________________ Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

2 1

Problem when Interchanging audio and video streams
by gihanchamara＠gmail.com 08 Mar '10

08 Mar '10

Hi I want to put audio stream first and video stream secondly in 3gp video (0:0 -audio stream and 0:1 video stream). I tried to do it passing following parameters ./ffmpeg -i test.3gp -acodec libopencore_amrnb -ab 4.75k -ac 1 -ar 8000 -s 176x144 -vcodec mpeg4 -r 25 -b 128k -map 0:1 -map 0:0 out.3gp but it gives error "Codec type mismatch for mapping #0.1 -> #0.0" I tried and ffmpeg always put video stream first. But Apple quicktime pro can do that. I want to know how i can do this using ffmpeg. Please I would very much grateful if you can give me a solution soon. Thanks, Gihan

1 0

problem in compute_pkt_fields2()
by faizan.ahmed39＠gmail.com 08 Mar '10

08 Mar '10

during av_interleaved_write_frame() when the program reaches compute_pkt_fields2() it returns -1 bcz of : if(st->cur_dts && st->cur_dts != AV_NOPTS_VALUE && st->cur_dts >= pkt->dts){} can anybody tell me how to fix it i am muxing 3 streams and only 2 streams get done and the 3rd has about 90% error return from av_interleaved_write_frame() bcz of st->cur_dts>=pkt->dts i have checked and st->cur_dts does have a larger value than pkt->dts help please -- View this message in context: http://n4.nabble.com/problem-in-compute-pkt-fields2-tp1584456p1584456.html Sent from the FFmpeg-users mailing list archive at Nabble.com.

1 0

Compilation Failure: 'No rule to make target `install-headers''
by andrew.david.45＠gmail.com 07 Mar '10

07 Mar '10

Hi, I am having some trouble with compiling the current svn FFmpeg on Slackware 13 and gcc 4.3.3 with the following error message: make: *** No rule to make target `install-headers'. Stop. Please let me know if more information is required. Andrew -- Do you think that's air you're breathing?

1 1

Video size to fit in screen
by dav19＠inbox.lv 07 Mar '10

07 Mar '10

Hi! I am using ffmpeg to convert videos for MP3 player and phone. They have screen sizes X by Y and I want to convert and resize videos to fit on that screen. If video doesn't have the same aspect ratio as MP3 player/phone then it should be smaller in one dimension (to maintain aspect ratio). Does ffmpeg support such option or should I write a script to do this? BR, Davis

3 3

Build with --enable-static
by mark＠mdsh.com 07 Mar '10

07 Mar '10

I'm confused, again, with compiling a static monolithic ffmpeg. Only this time I'm compiling for Linux. When I ./configure with --enable-static --disable-shared the resultant file is still dependent on many libraries. Only if I include --extra-libs=-static --extra-cflags=--static in the ./configure options do I get a truly static binary. Is that what I should expect? Because, if so, I'm very confused by what the --enable-static switch is actually for! Here are two examples: without --extra-(libs|cflags) mdsh at revo01:~/src/ffmpeg$ ./ffmpeg FFmpeg version SVN-r22244, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 6 2010 18:58:00 with gcc 4.4.1 configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-static --disable-shared --enable-pthreads --enable-runtime-cpudetect libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... Use -h to get full help or, even better, run 'man ffmpeg' mdsh at revo01:~/src/ffmpeg$ ldd ./ffmpeg linux-gate.so.1 => (0x00813000) libz.so.1 => /lib/libz.so.1 (0x00e4f000) libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0x00bd8000) libasound.so.2 => /usr/lib/libasound.so.2 (0x0055a000) libjack.so.0 => /usr/lib/libjack.so.0 (0x00993000) libpthread.so.0 => /lib/tls/i686/cmov/libpthread.so.0 (0x00110000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x0063c000) /lib/ld-linux.so.2 (0x00f1d000) libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0x007a8000) librt.so.1 => /lib/tls/i686/cmov/librt.so.1 (0x00dc3000) libsamplerate.so.0 => /usr/lib/libsamplerate.so.0 (0x00129000) libcelt.so.0 => /usr/lib/libcelt.so.0 (0x00466000) mdsh at revo01:~/src/ffmpeg$ file ./ffmpeg ./ffmpeg: ELF 32-bit LSB executable, Intel 80386, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.15, stripped and with --extra-(libs|cflags) mdsh at revo01:~/src/ffmpeg$ ./ffmpeg FFmpeg version SVN-r22244, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 6 2010 18:38:37 with gcc 4.4.1 configuration: --prefix=/usr/local --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-static --disable-shared --enable-pthreads --enable-runtime-cpudetect --extra-libs='-static -L/usr/lib' --extra-cflags=--static libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}... Use -h to get full help or, even better, run 'man ffmpeg' mdsh at revo01:~/src/ffmpeg$ ldd ./ffmpeg not a dynamic executable mdsh at revo01:~/src/ffmpeg$ file ./ffmpeg ./ffmpeg: ELF 32-bit LSB executable, Intel 80386, version 1 (SYSV), statically linked, for GNU/Linux 2.6.15, stripped Thanks. -- Mark

2 3

help with RTP out
by nash.tsai＠gmail.com 07 Mar '10

07 Mar '10

Hi, May I get help with constructing the ffmpeg command arguments that encoded video in xvid codec and use RTP as its output stream. Thanks in advance. Nash

1 0

Compiling on Mac G3 PPC
by tome＠QX.Net 06 Mar '10

06 Mar '10

Thanks David, Programmingkid also suggested disabling Altivec, which got me going a few days ago. I very much appreciate your reply. Best regards, Tom ------- Original Message -------

3 2

Compiling on Mac G3 PPC
by tome＠QX.Net 06 Mar '10

06 Mar '10

Hello, I have been attempting to install a download of ffmpeg-0.5.tar.bz2 for the past week or so, with no success. I have tried various configure options, based on internet information I have read, again with no success. The configure and make seem to run without error, but with many warnings. When I execute the resulting ffmpeg I get illegal instruction. My system is 500 MHz PPC 750 G3 (pismo) running OS X 10.4.11. The gcc is 4.0.0 GNU make is 3.81 built for powerpc-apple-darwin8.11.0 I am not a computer science person and do not understand much of what I have read about ffmpeg installation options. If anyone has experience with or can provide any guidance for ffmpeg build for ppc, your help will be greatly appreciated! Best regards to all, Tom

2 1

Provide input to ffmpeg from another program
by dmccallum9＠gmail.com 05 Mar '10

05 Mar '10

Hello, I would like to perform some changes to a webcam's video stream before I compress it with ffmpeg. I'm considering using a named pipe - my program would output video frames into the named pipe, and ffmpeg would read them. What data format should I use here? I'm looking for the simplest format that ffmpeg understands, preferably an endless stream of single images. Any help is much appreciated!

1 0

need help on DV video decoding
by pcuser.mails＠gmail.com 05 Mar '10

05 Mar '10

I am trying to write a wrapper for dv dec(dv,dvcpro25/50) by reading input from a dv encoded video file.I have some progressive encoded dv video files in my pc. But I don't know how to interpret the (.dv) file,I dont have the file format spec for dv. can somebody provide me insight on dv file format, or atleast provide me the link for the file format spec of dv. I have seen that ffmpeg can decode all types of dv files. "on what file format ffmpeg dv decoder is implemented?how it interprets dv files and extract header and metadata?" If dv is wrapped in avi container,will there be any change in dv headers and meta data. can we extract .dv file from avi container without any change to header and metadata of .dv files PS:I have looked into SMPTE314M standard,it has info about data compression but not on dv file format I really got struck at the dv decoder wrapper for sometime without have dv file format specification... Any help would be greatly appreciated! Regards, RJ

3 2

Help with Threads
by slyttleton＠mac.com 05 Mar '10

05 Mar '10

I am trying to improve the encoding performance of my script by using -threads. However, I see no difference when I run with or without this option, it takes the same amount of time to work. I have compiled ffmpeg with the --enable-pthreads option. Can someone enlighten me as to how to properly utilize -threads? Thanks in advance. Here is the input: # ffmpeg -i igetit.mov -acodec aac -vcodec mpeg4 -threads 2 igetit1.mp4 Here is the output FFmpeg version SVN-r22008, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 4 2010 19:36:10 with gcc 4.1.2 20080704 (Red Hat 4.1.2-46) configuration: --prefix= --libdir= --bindir= --incdir= --extra-cflags= --extra-ldflags= libavutil 50. 9. 0 / 50. 9. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 1 codec frame rate differs from container frame rate: 59940.00 (59940/1) -> 29.97 (2997/100) Seems stream 3 codec frame rate differs from container frame rate: 59940.00 (59940/1) -> 29.97 (2997/100) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'igetit.mov': Duration: 00:04:30.46, start: 0.000000, bitrate: 1323 kb/s Stream #0.0(eng): Audio: aac, 44100 Hz, stereo, s16, 123 kb/s Stream #0.1(eng): Video: h264, yuv420p, 480x270, 1187 kb/s, 29.97 fps, 29.97 tbr, 299 70 tbn, 59940 tbc Stream #0.2(eng): Audio: aac, 44100 Hz, stereo, s16, 123 kb/s Stream #0.3(eng): Video: h264, yuv420p, 480x270, 1187 kb/s, 29.97 fps, 29.97 tbr, 299 70 tbn, 59940 tbc Output #0, mp4, to 'igetit1.mp4': Metadata: encoder : Lavf52.54.0 Stream #0.0(eng): Video: mpeg4, yuv420p, 480x270, q=2-31, 200 kb/s, 2997 tbn, 29.97 t bc Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 8102 fps=166 q=3.6 Lsize= 8918kB time=270.37 bitrate= 270.2kbits/s dup=0 drop=2 video:6703kB audio:2056kB global headers:0kB muxing overhead 1.808421%

2 3

mpegts that can't be decoded
by mark＠mdsh.com 05 Mar '10

05 Mar '10

I have an mpegts that is an off-air recording - here is the first 20MB of it: http://mdsh.com/tmp/weatherview.ts Mplayer plays it with a large-ish tsprobe: $ mplayer -tsprobe 10240000 weatherview.ts VLC plays it. But FFmpeg does not find enough details in the stream to decode it - I want stream 0 and 1 from below: I hoped that -probesize would do the same as -tsprobe on mplayer but it made no difference. What incantation do I need to convince ffmpeg to probe far enough through the file so it can decode it? $ ffmpeg -i weatherview.ts FFmpeg version SVN-r22221, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 5 2010 12:03:14 with gcc 4.0.1 (Apple Inc. build 5490) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --mandir=/opt/local/share/man --enable-shared --disable-shared --disable-indev=jack --enable-pthreads --cc=gcc-4.0 libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.17. 0 / 1.17. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 0x1002600]max_analyze_duration reached [NULL @ 0x1007000]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1007400]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1007c00]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1008000]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1008400]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1008800]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1008c00]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1009000]start time is not set in av_estimate_timings_from_pts [NULL @ 0x1009400]start time is not set in av_estimate_timings_from_pts Input #0, mpegts, from 'weatherview.ts': Duration: 14:39:47.92, start: 52754.286722, bitrate: 3 kb/s Program 4164 BBC ONE Metadata: name : BBC ONE provider_name : BBC Stream #0.0[0x258]: Video: mpeg2video, 90k tbr, 90k tbn, 90k tbc Stream #0.1[0x259](eng): Audio: mp3, 0 channels, s16 Stream #0.2[0x25a](eng): Audio: mp2, 48000 Hz, 1 channels, s16, 64 kb/s Stream #0.3[0x25d](eng): Subtitle: dvbsub Stream #0.4[0x28a]: Data: 0x000b Stream #0.5[0x28b]: Data: 0x000b Stream #0.6[0x28c]: Data: 0x000b Stream #0.7[0x3ed]: Data: 0x000b Stream #0.8[0x3ee]: Data: 0x000b Stream #0.9[0x3ef]: Data: 0x000b Program 4351 BBC THREE Metadata: name : BBC THREE provider_name : BBC Program 4479 BBC Red Button Metadata: name : BBC Red Button provider_name : BBC Program 4671 CBBC Channel Metadata: name : CBBC Channel provider_name : BBC Program 4415 BBC NEWS Metadata: name : BBC NEWS provider_name : BBC Program 4228 BBC TWO Metadata: name : BBC TWO provider_name : BBC At least one output file must be specified -- Mark

1 0

ffmpeg export snapshot
by pupilla＠libero.it 05 Mar '10

05 Mar '10

They aren't updated since 2010 03 02. Is there any problem?

1 0

avcodec_encode_audio doesn't return
by rowen＠protoven.com 04 Mar '10

04 Mar '10

I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context->frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. Interestingly, if I replace my source audio with all 0's, encoding works. Also, if I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { AVPacket pkt; int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); } Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated!

1 0

Multithreading help
by slyttleton＠mac.com 04 Mar '10

04 Mar '10

I am trying to speed up my encoding time by using the -threads command. I am on a server with 16 processors and want to maximize their usage. However, when I run the command below with the threads option, I get the error at the bottom of the output. When I omit the -threads, it encodes fine (but rather slow). Any thoughts? Thanks in advance. Steve ***Input/output*** ffmpeg -async 1 -i igetit.mov -acodec libfaac -ac 2 -ar 22050 -ab 96k -vcodec libx264 -fpre /home/readyg5/.ffmpeg/libx264-default.ffpreset -s 640x360 -threads 0 igetit.mp4 FFmpeg version SVN-r22008, Copyright (c) 2000-2010 the FFmpeg developers built on Feb 26 2010 10:03:37 with gcc 4.1.2 20080704 (Red Hat 4.1.2-46) configuration: --prefix= --libdir=/home/readyg5/lib --bindir=/home/readyg5/bin --incdir =/home/readyg5/include --enable-shared --enable-gpl --enable-nonfree --extra-cflags=-I/ho me/readyg5/include --extra-ldflags=-L/home/readyg5/lib --enable-libx264 --enable-libvorbi s --enable-libfaad --enable-libfaac --enable-postproc --enable-pthreads --enable-libmp3la me --enable-libtheora libavutil 50. 9. 0 / 50. 9. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 1 codec frame rate differs from container frame rate: 59940.00 (59940/1) -> 29.97 (2997/100) Seems stream 3 codec frame rate differs from container frame rate: 59940.00 (59940/1) -> 29.97 (2997/100) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'igetit.mov': Duration: 00:04:30.46, start: 0.000000, bitrate: 1323 kb/s Stream #0.0(eng): Audio: aac, 44100 Hz, stereo, s16, 123 kb/s Stream #0.1(eng): Video: h264, yuv420p, 480x270, 1187 kb/s, 29.97 fps, 29.97 tbr, 299 70 tbn, 59940 tbc Stream #0.2(eng): Audio: aac, 44100 Hz, stereo, s16, 123 kb/s Stream #0.3(eng): Video: h264, yuv420p, 480x270, 1187 kb/s, 29.97 fps, 29.97 tbr, 299 70 tbn, 59940 tbc [libx264 @ 0xb8d0c80]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0xb8d0c80]profile High, level 3.0 Output #0, mp4, to 'igetit.mp4': Metadata: encoder : Lavf52.54.0 Stream #0.0(eng): Video: libx264, yuv420p, 640x360, q=10-51, 200 kb/s, 2997 tbn, 29.9 7 tbc Stream #0.1(eng): Audio: aac, 22050 Hz, stereo, s16, 96 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding Video encoding failed

1 0

Looking for pragmatic help - 1:1 Vido chat
by d＠panaton.com 04 Mar '10

04 Mar '10

Hello, I am looking for some help - we need a "simple" 1:1 web based Video Chat in on of our browser apps. Would anyone be available for a little consulting or can you recommend a best practice approach? Dejan Nenov

2 1

What coders are being used?
by robertk54＠aol.com 04 Mar '10

04 Mar '10

If I process a file as follows . . . ffmpeg -i file1.flv -acodec copy -b 350k file2.flv What coders are being used? If file1.flv contains H.264 / AAC, and my ffmpeg supports libx264, I was expecting fle2.flv to contain H.264 / AAC where the H.264 was decoded with the built-in H.264 decoder and encoded using the libx264 encoder (at 350k bps). However, when I play back using VLC, it is identifying FLV1 as the video type. Does the FLV1 encoder use libx264 under the hood or is this another codec? Thanks, Bob

2 1

Frames from video
by vlad.sub＠borovcov.ru 04 Mar '10

04 Mar '10

Hi all! I'm beginer in linux... And.. I'm russian, sorry for my broken English. I can't understand, how to extract frames (images) from video using ffmpeg ? what means message "Unsupported pixel format requested"? Thanks.

3 3

Enc: Help FFMPEG
by marcos_ourica＠yahoo.com.br 04 Mar '10

04 Mar '10

Good Morning, My name is Marcos Flavio, I have a problem with ffmpeg. When I convert a video with the extension. MPG to. FLV, the time bar does not work, the video plays but the time bar is stopped. Can someone help me on this? Regards, Marcos Fl?vio marcos_ourica at yahoo.com.br ________________________________ Veja quais s?o os assuntos do momento no Yahoo! + Buscados: Top 10 - Celebridades - M?sica - Esportes ____________________________________________________________________________________ Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

2 2

Need some advice
by robertk54＠aol.com 04 Mar '10

04 Mar '10

I am relatively new to ffmpeg but I am a a very experienced software engineer (mainly real-time embedded). I am investigating to see if the ffmpeg framework can be used for a media converter project I am working on. The product will be sort of a proxy server which will transcode media requests in order to control bandwidth. It will sit between various clients (not ours) such as browsers and media servers (also not ours). I think ffmpeg can do this but I am not sure exactly how to get started. It seems that in order to this as part of an application I would need to duplicate much of the code in ffmpeg and ffserver (the application code not the libraries). I do not like this approach. I was hoping I could just use ffmpeg directly such that it can transcode the media as it streams through the system. If anyone could provide some insight it would be greatly appreciated. Thanks, Bob

3 15

ffmpeg option request -force-cfr for libx264
by kla960＠icqmail.com 03 Mar '10

03 Mar '10

I need the option -force-cfr for the libx264 video codec. Please implement that option to the ffmpeg tool.

3 2

urgent help: wowza + xuggle + ffmpeg
by wrvishnu＠gmail.com 03 Mar '10

03 Mar '10

Hi I am using wowza videochat sample to broadcast the webcam , and using xuggler ffmpeg command to convert the stream inti iphone compaitable format. While using wowza sample am able to encode stream to the Iphone compaitable format . but when i use some other sample to broadcast the video like FMS or red5 sample stream is not encoding . I tried by open the wowza sample source and save it in adobe flas cs4 without any modification stream is not encoding. Can someone help me what could be the issue with the other samples. Below is command line with java code used to encode .(works only with wowza sample) Java Code package com.globalwebhq.transcoder; import java.io.IOException; import java.sql.Connection; import java.sql.DriverManager; import java.sql.ResultSet; import java.sql.Statement; import com.wowza.wms.amf.AMFPacket; import com.wowza.wms.logging.WMSLogger; import com.wowza.wms.stream.IMediaStream; import com.wowza.wms.stream.IMediaStreamActionNotify2; public class StreamListener implements IMediaStreamActionNotify2 { private WMSLogger logger = null; Connection conn = null; public StreamListener(WMSLogger logger2) { this.logger = logger2; } @Override public void onMetaData(IMediaStream stream, AMFPacket metaDataPacket) { // TODO Auto-generated method stub } @Override public void onPauseRaw(IMediaStream stream, boolean isPause, double location) { // TODO Auto-generated method stub } @Override public void onPause(IMediaStream stream, boolean isPause, double location) { // TODO Auto-generated method stub } @Override public void onPlay(IMediaStream stream, String streamName, double playStart, double playLen, int playReset) { // TODO Auto-generated method stub } @Override public void onPublish(IMediaStream stream, String streamName, boolean isRecord, boolean isAppend) { // TODO Auto-generated method stub //TransCoder tc = new TransCoder(stream.getName(), logger); //tc.start(); logger.info("onStreamCreate: " + stream.getSrc()); //String command = "C:\\test.bat"; //Poor working Quality /* String command = "ffmpeg -i rtmp://192.168.30.145:1936/live/ "+ streamName + " -re -acodec libfaac -ar 48000 -s 320?240 -b 512k -bufsize 384 - coder 0 -level 13 -vcodec libx264 -vpre hq -vpre" + " baseline3 -f flv rtmp://192.168.30.145:1936/live2/ "+streamName+"trans"; */ try { Statement stmt; ResultSet rs; System.out.println("....1"); String userName = "root"; String password = "welcome"; String url = "jdbc:mysql://192.168.30.145/newdesign"; logger.info("Before Register"); Class.forName ("com.mysql.jdbc.Driver").newInstance (); logger.info("After Register"); conn = DriverManager.getConnection (url, userName, password); logger.info("After Driver Manager"); System.out.println("....4"); logger.info("Database Connected"); stmt = conn.createStatement(); stmt.executeUpdate("INSERT INTO TblStreams(StreamName) VALUES('"+streamName+"')"); conn.commit(); } catch (Exception e) { logger.info("Cannont connect to the database : "+e); } finally { if (conn != null) { try { conn.close (); logger.info("Connection Closed"); } catch (Exception e) { /* ignore close errors */ } } } //better working Quality String command = "ffmpeg -i rtmp://192.168.30.145:1936/live/ "+ streamName + " -re -acodec libfaac -crf 24 -bt 256k -ab 128k -ar 48000 -s 320?240 -b 512k -bufsize 384 -coder 0 -level 13 -vcodec libx264 -vpre hq -vpre" + " baseline3 -f flv rtmp://192.168.30.145:1936/live2/ "+streamName+"trans"; Runtime rt= Runtime.getRuntime(); logger.info("onStreamCreate: " + stream.getSrc()); try { logger.info("Before Conversion"); rt.exec(command); //URL url = new URL(" http://192.168.30.145/testpost1.cfm?<http://www.google.com/url?sa=D&q=http://192.168.30.145/testpost1.cfm%3F&usg…> st="+streamName); //URLConnection conn = url.openConnection(); // conn.setDoOutput(true); // OutputStreamWriter wr = new OutputStreamWriter(conn.getOutputStream()); // String data = "test"; // wr.write(data); // wr.flush(); // wr.close(); logger.info("After Conversion"+ command); } catch (IOException e) { System.out.println(e); } } @Override public void onSeek(IMediaStream stream, double location) { // TODO Auto-generated method stub } @Override public void onStop(IMediaStream stream) { // TODO Auto-generated method stub } @Override public void onUnPublish(IMediaStream stream, String streamName, boolean isRecord, boolean isAppend) { // TODO Auto-generated method stub try { Statement stmt; ResultSet rs; System.out.println("....1"); String userName = "root"; String password = "welcome"; String url = "jdbc:mysql://192.168.30.145/newdesign"; logger.info("Before Register"); Class.forName ("com.mysql.jdbc.Driver").newInstance (); logger.info("After Register"); conn = DriverManager.getConnection (url, userName, password); logger.info("After Driver Manager"); System.out.println("....4"); logger.info("Database Connected"); stmt = conn.createStatement(); stmt.executeUpdate("delete from TblStreams where StreamName ='"+streamName+"'"); conn.commit(); } catch (Exception e) { logger.info("Cannont connect to the database : "+e); } finally { if (conn != null) { try { conn.close (); logger.info("Connection Closed"); } catch (Exception e) { /* ignore close errors */ } } } } -- Thks and Regards W.R.Vishnu Prasad Mobile :+65-83060528

1 0

swscale and AVFrame use of const
by kaspar.bumke＠gmail.com 02 Mar '10

02 Mar '10

Hi, I have been trying to follow a simple tutorial on writing a player for ffmpeg: http://dranger.com/ffmpeg/tutorial01.html It is a little outdated as I am using ffmpeg SVN-r21688 but I have replaced img_convert with sws_scale as follows: --- tutorial01_old.c 2010-02-27 22:06:43.000000000 +0000 +++ tutorial01.c 2010-02-27 22:17:35.000000000 +0000 (...) @@ -124,17 +125,23 @@ // Is this a packet from the video stream? if(packet.stream_index==videoStream) { // Decode video frame - avcodec_decode_video(pCodecCtx, pFrame, &frameFinished, - packet.data, packet.size); + avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, + &packet); // Did we get a video frame? if(frameFinished) { // Convert the image from its native format to RGB - img_convert((AVPicture *)pFrameRGB, PIX_FMT_RGB24, - (AVPicture*)pFrame, pCodecCtx->pix_fmt, pCodecCtx->width, - pCodecCtx->height); - - // Save the frame to disk + + convert_ctx = sws_getContext( pCodecCtx->width, + pCodecCtx->height, + pCodecCtx->pix_fmt, + pCodecCtx->width, + pCodecCtx->height, + PIX_FMT_RGB24, + SWS_BILINEAR, 0, 0, 0); + + sws_scale(convert_ctx, pFrame->data, pFrame->linesize, 0, pCodecCtx->height, pFrameRGB->data, pFrameRGB->linesize); + if(++i<=5) SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height, i); I have attached the whole modified file which should be compiled with: gcc -o tutorial01 tutorial01.c -lavformat -lavcodec -lz The problem is during compilation it says sws_scale expexts a ?const uint8_t * const*? but argument is of type ?uint8_t **? in the second field. I have been lookign at the doxygen documentation and other peoples players and trying all sorts of things but I cannot figure out how to change and AVFrame->data to a const or what what I should really be passing to sws_scale in the second field. Any help at all would be greatly appreciated. --Kaspar P.S. This similar implementation has exactly the same problem: http://web.me.com/dhoerl/Home/Tech_Blog/Entries/2009/1/22_Revised_avcodec_s…

1 2

Split a video for Youtube
by tokland＠gmail.com 02 Mar '10

02 Mar '10

Hi, I am writing a script to split videos in order to match Youtube restrictions (<100Mb, <10'). For now I have something like that: $ ffmpeg -i in.avi -fs 100000000 -t 600 -s $OFFSET out-$INDEX.avi $OFFSET starts at 0 and it's incremented with the duration of the previous chunk. Finally, when $OFFSET+$DURATION_OF_CHUNK >= $TOTAL_DURATION, I know I have to break. Problem is that sometimes, depending on the input file, the sum of chunks does not reach the input video duration; then, the script does not know this was the last chunk and continues to loop. I know I could check the chunk size and time duration and compare against my restrictions (if none is meet, then it was the last chunk), but maxfilesize is actually never reached. Any ideas? thanks

1 0

Jerky output with when playing (ffplay) or transcoding (ffmpeg) MPEG-2 TV recording
by jaakko.peltonen＠nbl.fi 02 Mar '10

02 Mar '10

Hello, I would like to use FFmpeg to transcode videos that I have recorded off Finnish digital TV. Ultimately, I want to convert the videos to another format that will work on an Archos 3 player device. However, for some of my recorded .mpg files, I have encountered a problem when using ffmpeg to encode the file, and even when using ffplay to play the file. Basically, the video looks like it stops very briefly about every 0.5 seconds, which makes any originally continuous motion look jerky. I think this behavior has something to do with how FFplay and FFmpeg decode the input video. - The original .mpg file plays without problems using Windows Media Player, MPlayer or my DVD software (InterVideo WinDVD). - However, if I play it with FFplay the motion looks jerky. - If I transcode it to AVI with FFmpeg, the output video looks jerky in both InterVideo WinDVD and in MPlayer; Windows Media Player cannot play it at all. - If I transcode it simply to .mpg with FFmpeg, the output video looks jerky in InterVideo WinDVD. In MPlayer the output video does not look jerky but there is an audio screech at the start. Windows Media Player again cannot play the output video at all. I guess it is possible that the input .mpg file is partly damaged, e.g. due to poor TV reception when I recorded it (is there some way I could check this?). However, if that is the case, then Windows Media Player, MPlayer and InterVideo WinDVD seem to be able to tolerate the damage better. It would be really great if FFmpeg could avoid the jerkiness too! I experienced this behavior using both the old FFmpeg/FFplay build that comes with WinFF 1.2, and also using the recent autobuild downloaded on February 28 from http://ffmpeg.arrozcru.org/autobuilds/ Here are example commandlines: ffplay testinput.mpg (This shows jerky motion.) ffmpeg -i testinput.mpg -vcodec copy -acodec copy testoutput.mpg (The result is jerky in ffplay and in InterVideo WinDVD; mplayer gives an audio screech at the start; and Windows Media Player cannot play it at all.) Thanks for any help. Best regards, Jaakko Peltonen

1 0

Audio mp2, video blurry
by jmmckee＠flinthills.com 02 Mar '10

02 Mar '10

-----Original message----- From: Christian Ebert blacktrash at gmx.net Date: Tue, 02 Mar 2010 03:32:19 -0600 To: ffmpeg-user at mplayerhq.hu Subject: Re: [FFmpeg-user] Audio mp2, video blurry > * jmmckee on Sunday, February 28, 2010 at 11:41:37 -0600 > >> * Jarred Nicholls on Monday, February 22, 2010 at 22:39:37 -0500 > >>> You need -acodec libmp3lame, thus: > >>> > >>> ffmpeg -i -vcodec msmpeg4v2 -b 200 *-acodec libmp3lame *-ab 64 -ar 22050 -s 320x240 /tmp/video.avi > >> ^^^^^^ > >> I guess that should be: -ab 64k > > As Robert noted, also: -b 200k > > > It works great now. However, I get a message stating that > > audio encoding has failed from lame. This is confusing, as the > > encoded file has great video and audio. Any ideas what that > > message is referring to? > > And what is the message? Please post exact commandline and > output. > > You never said whether/how compilation against lame went. It may > be that your ffmpeg is still compiled without lame. > > c > -- > \black\trash movie _C O W B O Y_ _C A N O E_ _C O M A_ > Ein deutscher Western/A German Western > > --->> http://www.blacktrash.org/underdogma/ccc.php > _______________________________________________ This is the command used to encode: 2>&1 ffmpeg -i "$(find ~/.mozilla -regex '.*Cache.*' -a -not -regex \ '.*_CACHE_.*' -printf '%T+ %p\n' | \ sort -n | awk '{ print $2 }' |xargs file | \ grep -i "Video" | tail -1 | awk -F : '{ print $1 }')" \ -vcodec msmpeg4v2 -b 200k -acodec libmp3lame \ -ab 64k -ar 22050 -s 320x240 \ /tmp/video.avi |tee dwnload.log And, the bottom of dwnload.logThis is a portion from mplayer . The error from lame was not included in dwnload.log, just directed to stderr: frame= 3077 fps=113 q=14.4 size= 3579kB time=102.77 bitrate= 285.3kbits/s dup frame= 3130 fps=113 q=10.7 size= 3639kB time=104.59 bitrate= 285.0kbits/s dup [libmp3lame @ 0xa261ac0]lame: output buffer too small (buffer index: 9405, free bytes: 387) Audio encoding failed How is buffer size for lame set? A compile option? This is a portion from mplayer -identify: Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 22050 Hz, 2 ch, s16le, 64.0 kbit/9.07% (ratio: 8000->88200) ID_AUDIO_BITRATE=64000 ID_AUDIO_RATE=22050 ID_AUDIO_NCH=2 Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== AO: [pulse] 22050Hz 2ch s16le (2 bytes per sample) ID_AUDIO_CODEC=mad Forced audio codec: mad Opening audio decoder: [libmad] libmad mpeg audio decoder AUDIO: 22050 Hz, 2 ch, s16le, 64.0 kbit/9.07% (ratio: 8000->88200) ID_AUDIO_BITRATE=64000 ID_AUDIO_RATE=22050 ID_AUDIO_NCH=2 Selected audio codec: [mad] afm: libmad (libMAD MPEG layer 1-2-3) ========================================================================== AO: [pulse] 22050Hz 2ch s16le (2 bytes per sample) ID_AUDIO_CODEC=mad I hear the audio. Is it correct to assume that audio is NOT mp3? How do I show if ffmpeg has lame compiled? I thought pulse was not mp3. My guess is that lame failed and left audioas pulse. But, it is something I can hear now, as opposed to before attempting to use lame. Thanks for the patience. John McKee

1 0

Audio mp2, video blurry
by jmmckee＠flinthills.com 02 Mar '10

02 Mar '10

-----Original message----- From: Christian Ebert blacktrash at gmx.net Date: Tue, 23 Feb 2010 03:46:50 -0600 To: ffmpeg-user at mplayerhq.hu Subject: Re: [FFmpeg-user] Audio mp2, video blurry > * Jarred Nicholls on Monday, February 22, 2010 at 22:39:37 -0500 > > You need -acodec libmp3lame, thus: > > > > ffmpeg -i -vcodec msmpeg4v2 -b 200 *-acodec libmp3lame *-ab 64 -ar 22050 -s 320x240 /tmp/video.avi > ^^^^^^ > I guess that should be: -ab 64k > > c > -- > \black\trash movie _S A M E_ _T I M E_ _S A M E_ _P L A C E_ > --->> http://www.blacktrash.org/underdogma/stsp.php > \black\trash audio _A N O T H E R_ _T I M E_ _A N O T H E R_ _P L A C E_ > --->> http://www.blacktrash.org/underdogma/atap.html > _______________________________________________ It works great now. However, I get a message stating that audio encoding has failed from lame. This is confusing, as the encoded file has great video and audio. Any ideas what that message is referring to? John McKee

2 1