October 2006 - ffmpeg-user - lists.ffmpeg.org

How to convert .3g2 .mp4 files to SWF format
by shamjuj＠yahoo.com 21 Mar '07

21 Mar '07

Dear all I'm getting errors when I tried to convert some .3g2, .mp4 files to SWF format. The output listings are given below. Please suggest me some solution. Thanks Shamju Joseph ---------------------------------------------------- bash-2.05b$ ./ffmpeg -sameq -i 0616060922.3g2 -f swf 0616060922.swf FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --extra-cflags=-I/home/shamju/Demo1/lame/include --extra-ldflags=-L/home/shamju/Demo1/lame/lib --enable-mp3lame --prefix=/home/shamju/Demo1/ffmpegsvn --cc=gcc32 libavutil version: 49.0.0 libavcodec version: 51.11.0 libavformat version: 50.5.0 built on Aug 30 2006 11:32:41, gcc: 3.2.3 20030422 (Red Hat Linux 3.2.3-6) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '0616060922.3g2': Duration: 00:00:14.8, start: 0.000000, bitrate: 118 kb/s Stream #0.0(eng): Video: h263, yuv420p, 176x144, 29.97 fps(r) Stream #0.1(eng): Audio: mp4a / 0x6134706D, 8000 Hz, stereo Output #0, swf, to '0616060922.swf': Stream #0.0: Video: flv, yuv420p, 176x144, q=2-31, 200 kb/s, 29.97 fps(c) Stream #0.1: Audio: mp3, 8000 Hz, stereo, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec (id=86043) for input stream #0.1 ---------------------------------------------------- bash-2.05b$ ./ffmpeg -sameq -i GetTech.mov -f swf GetTech.swf FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --extra-cflags=-I/home/shamju/Demo1/lame/include --extra-ldflags=-L/home/shamju/Demo1/lame/lib --enable-mp3lame --prefix=/home/shamju/Demo1/ffmpegsvn --cc=gcc32 libavutil version: 49.0.0 libavcodec version: 51.11.0 libavformat version: 50.5.0 built on Aug 30 2006 11:32:41, gcc: 3.2.3 20030422 (Red Hat Linux 3.2.3-6) [mov,mp4,m4a,3gp,3g2,mj2 @ 0x8339b5c]negative ctts, ignoring Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'GetTech.mov': Duration: 00:00:31.2, start: 0.000000, bitrate: 152 kb/s Stream #0.0(eng): Audio: mp4a / 0x6134706D, 32000 Hz, mono Stream #0.1(eng): Video: h264, yuv420p, 160x120, 9.99 fps(r) Output #0, swf, to 'GetTech.swf': Stream #0.0: Video: flv, yuv420p, 160x120, q=2-31, 200 kb/s, 9.99 fps(c) Stream #0.1: Audio: mp3, 32000 Hz, mono, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 [flv @ 0x83c180c]removing common factors from framerate Unsupported codec (id=86018) for input stream #0.0 ------------------------------------------------- bash-2.05b$ ./ffmpeg -sameq -i 2863.wmv -f swf 2863.swf FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --extra-cflags=-I/home/shamju/Demo1/lame/include --extra-ldflags=-L/home/shamju/Demo1/lame/lib --enable-mp3lame --prefix=/home/shamju/Demo1/ffmpegsvn --cc=gcc32 libavutil version: 49.0.0 libavcodec version: 51.11.0 libavformat version: 50.5.0 built on Aug 30 2006 11:32:41, gcc: 3.2.3 20030422 (Red Hat Linux 3.2.3-6) Seems that stream 1 comes from film source: 1000.00 (1000/1) -> 24.00 (24/1) Input #0, asf, from '2863.wmv': Duration: 00:00:41.0, start: 3.000000, bitrate: 408 kb/s Stream #0.0: Audio: wmav2, 32000 Hz, stereo, 48 kb/s Stream #0.1: Video: wmv2, yuv420p, 320x240, 24.00 fps(r) Output #0, swf, to '2863.swf': Stream #0.0: Video: flv, yuv420p, 320x240, q=2-31, 200 kb/s, 24.00 fps(c) Stream #0.1: Audio: mp3, 32000 Hz, stereo, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Could not write header for output file #0 (incorrect codec parameters ?) ---------------------------------------------------- bash-2.05b$ ./ffmpeg -sameq -i MP4sample_movie.mp4 -f swf MP4sample_movie.swf FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --extra-cflags=-I/home/shamju/Demo1/lame/include --extra-ldflags=-L/home/shamju/Demo1/lame/lib --enable-mp3lame --prefix=/home/shamju/Demo1/ffmpegsvn --cc=gcc32 libavutil version: 49.0.0 libavcodec version: 51.11.0 libavformat version: 50.5.0 built on Aug 30 2006 11:32:41, gcc: 3.2.3 20030422 (Red Hat Linux 3.2.3-6) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'MP4sample_movie.mp4': Duration: 00:01:14.3, start: 0.000000, bitrate: 902 kb/s Stream #0.0(eng): Data: mp4s / 0x7334706D Stream #0.1(eng): Data: mp4s / 0x7334706D Stream #0.2(eng): Video: mpeg4, yuv420p, 240x176, 29.97 fps(r) Stream #0.3(eng): Audio: mp4a / 0x6134706D, 44100 Hz, stereo Output #0, swf, to 'MP4sample_movie.swf': Stream #0.0: Video: flv, yuv420p, 240x176, q=2-31, 200 kb/s, 29.97 fps(c) Stream #0.1: Audio: mp3, 44100 Hz, stereo, 64 kb/s Stream mapping: Stream #0.2 -> #0.0 Stream #0.3 -> #0.1 Unsupported codec (id=86018) for input stream #0.3 ------------------------------------------------- __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Seg fault 3gp->flv
by incubateur2＠wanadoo.fr 17 Feb '07

17 Feb '07

Hello I am running ffmpeg on a debian. I just compile ffmpeg with amr_nb support but now when I try to encode a 3gp file I get a segmentation fault : # ffmpeg -y -i 173.3gp -vcodec flv -s 550x400 -acodec mp3 173.flv FFmpeg version SVN-r5970, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-mp3lame --enable-vorbis --enable-gpl --enable-libogg --enable-xvid --enable-dts --enable-amr_nb libavutil version: 49.0.0 libavcodec version: 51.11.0 libavformat version: 50.5.0 built on Aug 10 2006 12:44:24, gcc: 3.3.5 (Debian 1:3.3.5-13) Seems that stream 0 comes from film source: 250.00 (1000/4) -> 10.00 (10/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '173.3gp': Duration: 00:00:20.8, start: 0.000000, bitrate: 81 kb/s Stream #0.0(jpn): Video: h263, yuv420p, 176x144, 10.00 fps(r) Stream #0.1(jpn): Audio: amr_nb, 8000 Hz, mono Output #0, flv, to '173.flv': Stream #0.0: Video: flv, yuv420p, 550x400, q=2-31, 200 kb/s, 10.00 fps(c) Stream #0.1: Audio: mp3, 8000 Hz, mono, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding Segmentation fault I get the same error with -an instead of -acodec mp3 I get the same error regardless the video codec I use. thanks for any help !

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Video into Premiere Pro
by jdboyd＠jdboyd.net 09 Nov '06

09 Nov '06

I need to get assorted internet downloaded videos into Premiere Pro 1.5 for editing and output to DV. Premiere Pro will work most quickly if it is importing DV/AVI files. Has anyone done this? I tried doing this: ffmpeg -i in_file.m4v -f avi -r 29.97 -ar 48000 -s 720x480 -ac 2 -vcodec \ dvvideo -acodec pcm_s16le out_file.avi When I try to import out_file.avi into Premiere, it claims that it is an invalid file. I took a file output from Premiere Pro and did to it: ffmpeg -i Sample_NTSC.avi /dev/null to look at what ffmpeg thinks that file is and it says: Input #0, avi, from 'Sample_NTSC.avi': Duration: 00:00:43.5, start: 0.000000, bitrate: 30313 kb/s Stream #0.0: Video: dvvideo, yuv411p, 720x480, 29.97 fps Stream #0.1: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s When I do the same with the out_file.avi generated above, it shows the same Stream formats and mapping. Has anyone sucessfully done this sort of conversion? I was able to work out something using ffmpeg to extract audio and avisynth to get video, and premiere to realign the two and render them out to DV/AVI, but it is slow, it includes more steps than I want, and it ties up the Premiere Pro machine doing transcoding when I'd rather do the transcoding on a seperate (faster) linux machine. -- Joshua D. Boyd jdboyd at jdboyd.net http://www.jdboyd.net/ http://www.joshuaboyd.org/

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Overlaying audio back-to-back over the same video
by joelr1＠gmail.com 06 Nov '06

06 Nov '06

Folks, Is there a way to overlay several audio tracks over the same video sequentially? Should concatenate the audio files outside of ffmpeg and then lay the audio over? Thanks, Joel -- http://wagerlabs.com/

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Framerate, Duration and Keyframes
by Aya＠animes.de 02 Nov '06

02 Nov '06

Hi, I'm new to FFMpeg and am just trying to grab Frames out of an AVI. I followed the instructions of this tutorial: http://www.inb.uni-luebeck.de/~boehme/using_libavcodec.html Anything works fine, I am able to capture the frames etc.. But, how do I determine the Framerate and Framecount? This function returns 0 all the time: double av_q2d(pFormatCtx->streams[videoStream]->codec->time_base) the only way I found to get the Framerate is this, but.. isn't there another way (via Codec) too? double frameRate = av_q2d(pFormatCtx->streams[videoStream]->r_frame_rate); and, no matter what I do.. I am unable to calculate the total Framecount... pFormatCtx->streams[videoStream]->nb_frames // this returns 0 Could someone help me with this? Also, when I'm following the tutorial showed above and Playback a Video with this method.. the first frame is allways gray as if there is no first keyframe... does anyone know, why? (in any other Video player it looks ok) Kind regards, Aya~

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MP3 + JPG = video?
by alkemiman＠yahoo.co.uk 31 Oct '06

31 Oct '06

Hello, I'm currently working on a project whereby I need to create a video file from an mp3 audio track and one jpg image. I've been researching Transcode to see whether this is possible (as well as ffmpeg), and have attempted a few Demuxing techniques, but to no avail. Any help would be greatly appreciated. Regards Paul Paul Bawon alkemiman at yahoo.co.uk

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ffserver conf parms for asf stream
by sanjay＠upsnap.com 31 Oct '06

31 Oct '06

Hello everybody, I'm trying to setup a streaming server using ffserver/ffmpeg and I'm getting the live feed from my tv card. It's a Hauppauge pvr 350 (uses ivtv driver) card and outputs mpeg 2 video. I've used the default ffserver conf to generate asf stream and none of the players that I used can play the stream. I've googled and played around with the asf stream parms in the ffserver.conf with no success so far. I've used wmv1 & wmv2 as codecs with no success. I've attached my ffserver.conf just for reference. And here's the ffmpeg output $ cat /dev/video0 | ffmpeg -i - http://localhost:8090/feed1.ffm FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-amr_nb --enable-faac --enable-mp3lame --enable-x264 --enable-gpl libavutil version: 49.0.0 libavcodec version: 51.8.0 libavformat version: 50.4.0 built on Sep 22 2006 22:43:21, gcc: 4.0.4 20060507 (prerelease) (Debian 4.0.3-3) Input #0, mpeg, from 'pipe:': Duration: N/A, bitrate: 9984 kb/s Stream #0.0[0x1e0], 29.97 fps(r): Video: mpeg2video, yuv420p, 720x480, 9600 kb/s Stream #0.1[0x1c0]: Audio: mp2, 48000 Hz, stereo, 384 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Stream #0.0: Audio: mp3, 22050 Hz, mono, 64 kb/s Stream #0.1, 15.00 fps(c): Video: wmv2, yuv420p, 352x240, q=3-31, 256 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 [wmv2 @ 0x83a1888]removing common factors from framerate [wmv2 @ 0x83a1888]rc buffer underflow [wmv2 @ 0x83a1888]rc buffer underflow frame= 7117 q=31.0 Lsize= 18808kB time=474.1 bitrate= 325.0kbits/s Anybody who has successfully generated asf streams using ffserver here? Could anyone let me know what I'm missing ?? Appreciate any help.

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Skipping frames gives wrong timing of video
by tomas.stenlund＠telia.com 30 Oct '06

30 Oct '06

Hi, I'm trying to create a very low bitrate H.263 video sequence which works really good when using two passes to get a fixed bitrate. But I would also like to emit a video frame only when the change from the previous is "large" enough, so I found a -skip_threshold parameter for ffmpeg. When experimenting with this found I found out that: 1) It does not seem to work with two pass (assertion failed) due to the reason that not enough frames are located in the intermediare statistics file. Or should you emit it in the first pass and only use the switch in the second pass ? 2) The temporal reference for the frames are correct, i.e. in the H.236 picture headers. But the duration for each frame as shown in the quicktime sample table is not expanded to "fit" the "hole" until the next frame is emitted. And it seems that the trak information is calculated on the basis that no frames have been removed. This is causing the video to be much faster than the audio. So, now to my question. Is this two pass thing supposed to work together with skip_threshold (i.e. am I using it correcly)? If so it seems that the calculation of the trak duration and frame duration in the quicktime file is wrong, or ? If so I can provide with more data if anyone is interested. Thanks, Tomas

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buffer underflow errors
by otrcomm＠isp-systems.net 30 Oct '06

30 Oct '06

hello, when i use: ffmpeg -i soderberg.vob -f mp4 -vcodec mpeg4 -maxrate 1500 -b 1500 -qmin 3 -qmax 5 -bufsize 4096 -g 300 -acodec aac -ar 44100 -ab 192 -s 480x360 -aspect 4:3 soderberg-480x360.mp4 i get: [root at ripper soderberg]# ffmpeg -i soderberg.vob -f mp4 -vcodec mpeg4 -maxrate 1500 -b 1500 -qmin 3 -qmax 5 -bufsize 4096 -g 300 -acodec aac -ar 44100 -ab 192 -s 480x360 -aspect 4:3 soderberg-480x360.mp4 FFmpeg version SVN-r6830, Copyright (c) 2000-2006 Fabrice Bellard, et al. configuration: --prefix=/usr --enable-mp3lame --enable-faac --enable-x264 - -enable-a52 --enable-shared --enable-pthreads --enable-xvid --enable-gpl libavutil version: 49.0.2 libavcodec version: 51.22.0 libavformat version: 50.6.0 built on Oct 29 2006 01:11:29, gcc: 4.1.1 20060525 (Red Hat 4.1.1-1) Input #0, mpeg, from 'soderberg.vob': Duration: 00:33:36.2, start: 0.218678, bitrate: 7610 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x480, 7482 kb/s, 29.97 fps(r) Stream #0.1[0xa0]: Audio: pcm_s16be, 48000 Hz, stereo, 1536 kb/s Output #0, mp4, to 'soderberg-480x360.mp4': Stream #0.0: Video: mpeg4, yuv420p, 480x360, q=3-5, 1 kb/s, 29.97 fps(c) Stream #0.1: Audio: aac, 44100 Hz, stereo, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflowme=0.5 bitrate= 0.6kbits/s [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow [mpeg4 @ 0x714704]rc buffer underflow ... ... the buffer underflow stream goes on "forever" what am i doing wrong here. i created soderberg.vob with mplayer dvd://1 -dumpstream -dumpfile soderberg.vob any ideas what is causing the buffer underflows? the mp4 plays okay, but the buffer underflow stream is a pain. thanks, murrah boswell

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itsoffset having no effect
by mike＠redtux.org.uk 30 Oct '06

30 Oct '06

I am using variations of the following to try to correct sync ffmpeg -i /home/vid/dvd_files/vcr2ben2en.vob -itsoffset 24 -i /home/vid/dvd_files/vcr2ben2en.vob -sameq vcr2b.avi -map 0:0 -map 1:1 and it does not seem to have any effect (at all the amount out of sync is unchanged) anyone any ideas

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