April 2010 - ffmpeg-user - lists.ffmpeg.org

Binary Build of FFMpeg
by ambedkumar＠gmail.com 13 Apr '10

13 Apr '10

Hi Friends, Few Queries related to FFMpeg:- 1.) Does FFMpeg provides its binary build? 2.) What is the latest release of FFMpeg, as per the http://ffmpeg.org/index.html it seems 0.5.1, but on other site http://rpmfind.net/linux/rpm2html/search.php?query=ffmpeg&submit=Search+... I can find even 0.6. Is it the same FFMpeg and supported by its community. 3.) Is there any document which can be referred to find out all other third-party/dependency library of FFMpeg, For e.g.:- LAME. Any help on the above will be appreciated. -- Thanks and Regards, Ambed Kumar

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iPhone audio noise problems
by ferenc.deak＠adactus.no 13 Apr '10

13 Apr '10

Hi all, I have the following strange situation here: I'm converting a file to be played on the iphone, here is the command line: ffmpeg -i 33_500-00008.ts -vcodec libx264 -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 82k -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 -acodec libfaac -b 90000 -ab 22050 -ar 11000 -f mpegts -vbsf h264_mp4toannexb -s 480x270 ok33_500-0008.ts it starts doing its job, then at some time it gives: ..... [aac @ 0x1d19380]channel element 1.10 is not allocated Error while decoding stream #0.1 [aac @ 0x1d19380]Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. If you heard an audible artifact, please submit the sample to the FFmpeg developers. ..... but it finishes, and creates a result file. and now comes the strange thing: if I play the resulting TS file with ffplay it plays nicely and the audio is correct. If I play it with iPhone, it's pretty noisy, at a regular stage there are "peaking noises" (the best way I can describe it is like hearing the audio over a really old radio with lot of interference at regulated intervals). If required I can provide the sample file. And: If I extract the audio from the TS file and play it on an iphone, it plays nicely, no artefacts. Have you ever experienced anything like this? Thanks, Ferenc

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encoding a video with watermark
by cmptuomp3＠gmail.com 13 Apr '10

13 Apr '10

Hello guys, I want to encode a video and place an image (or text) as watermark. Is this possible with the latest (svn) ffmpeg? I'm currently using ffmpeg 0.5 because of the vhook support, and I know this was dropped at some point. Is there a way to do this with the latest ffmpeg? Thanks a lot, Ivo

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pts/dts jumps in clip boundaries
by sun_radha＠yahoo.com 13 Apr '10

13 Apr '10

Hi All, I am writing an application to join parts of a mpeg file, that is open a MPEG file seek to a position <x> and get <a> sec content , then seek to <y> get <b> sec content and so on. Put the clips to a single MPEG file. There is no encoding involved, re-encoding is done only for the first few frames until we hit a keyframe and from keyframe we just write packets to the output file. I wonder if someone can give ideas on how to handle packet pts/dts, ie modify time stamps so that they are continuous. Thanks Kris

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Should I avoid copying codecs when cropping padding, ... ?
by jamiesgreen＠gmail.com 12 Apr '10

12 Apr '10

Hi, Are my speculations correct that I'd better avoid copying the original video codec of a file when cropping and padding and avoid both codecs for changing the duration and start point?

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ffmpeg skips first imput frame
by jaymin＠gmx.net 11 Apr '10

11 Apr '10

I have a folder of 20 images (00000000.png to 00000019.png) and wanted to convert them into a DV video file. The problem here is that ffmpeg skips image 00000000.png. From the output below, I can see that ffmpeg recognizes the input to be 100s (00:01:40.00) in length but only encodes 95 seconds. [johann at johann committee]$ ffmpeg -f image2 -r 1:5 -i %08d.png -y -f dv -vcodec dvvideo -r 25 -s pal -aspect 4:3 -an test.dv FFmpeg version SVN-r20372, Copyright (c) 2000-2009 Fabrice Bellard, et al. built on Nov 7 2009 10:57:27 with gcc 4.4.2 20091027 (Red Hat 4.4.2-7) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --extra-version=rpmfusion --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avfilter-lavf --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 50. 3. 0 / 50. 3. 0 libavcodec 52.37. 1 / 52.37. 1 libavformat 52.39. 2 / 52.39. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1. 4. 1 / 1. 4. 1 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 [image2 @ 0x13b45e0]max_analyze_duration reached [png @ 0x13b5c40]Found 1 unreleased buffers! Seems stream 0 codec frame rate differs from container frame rate: 0.20 (1/5) -> 1.00 (1/1) Input #0, image2, from '%08d.png': Duration: 00:01:40.00, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb24, 720x576, 1 tbr, 0.20 tbn, 0.20 tbc Output #0, dv, to 'test.dv': Stream #0.0: Video: dvvideo, yuv411p, 720x576 [PAR 16:15 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 2375 fps=145 q=0.0 Lsize= 333984kB time=95.00 bitrate=28800.0kbits/s video:333984kB audio:0kB global headers:0kB muxing overhead 0.000000% [png @ 0x13b5c40]Found 2 unreleased buffers! [johann at johann committee]$

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"Insufficient audio data or severe sync problem."
by kshawkeye＠gmail.com 11 Apr '10

11 Apr '10

Just thought that I would be nice and update this, I fixed the issue by configuring FFmpeg for NTSC-DV myself. Here is my command line: ffmpeg -i "Invideo" -vcodec dvvideo -pix_fmt yuv411p -s 720x480 -aspect 4:3 -r 29.970 -acodec pcm_s16le -ar 48000 -ac 2 "Outvideo" I hope that if anyone else happens to have this issue this helps them.

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DVD created with ffmpeg has very low volume on DVD player but regular volume on computer
by odedrim2＠yahoo.com 10 Apr '10

10 Apr '10

Hello! I have an AVI video on my computer which I created using mencoder (recorded from a VHS tape using a TV card). This AVI plays very well on my computer. I've used different programs (mencoder, avidemux2) to encode this video to MPEG format, using different video and audio configurations (MP2 / AC3 audio and different bitrates). All of the resulting MPEGs look and sound well when I play them on my machine, and *all* have very very low sound volume (almost silent) when I create a DVD out of them and play it using an LG DVD player. The DVDs play fine on the computer, and the DVD player plays other DVD just fine. I'm using ffmpeg-0.5-3 on Fedora 11 with kernel v2.6.30, avidemux-2.5.2 and mencoder-1.0. Can anyone help? (if this is not the right forum, please tell me what is... thanx) Thank You, O.R.

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3d Video
by sangprasert＠reigntec.com 10 Apr '10

10 Apr '10

Is there a way to interlace 2 different 30fps video streams into 1 60fps video stream? I am asking because if it is possible, I will be able to create 3d movies that can be viewed using shutter glasses or is there another piece of software that already does this. I don't like using the red/blue glasses for a 3d effect. It just doesn't look good. Thanks.

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-acodec not correctly documented
by Cecil＠decebal.nl 10 Apr '10

10 Apr '10

I tried to convert a video with: ffmpeg -i input.mov -f avi -vcodec mpeg4 -b 400k -g 150 -bf 2 -acodec libmp3lame -ab 64k output.avi At input I get: Stream #0.1(eng): Audio: pcm_u8, 7875 Hz, 1 channels, u8, 63 kb/s and at output I get: Stream #0.1(eng): Audio: libmp3lame, 7875 Hz, 1 channels, s16, 64 kb/s So far so good, but I also get: Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height You have to define the audio frequency with for example '-ar 8000'. So it becomes: ffmpeg -i input.mov -f avi -vcodec mpeg4 -b 400k -g 150 -bf 2 -acodec libmp3lame -ar 8000 -ab 64k output.avi Would be nice if the documentation would be updated. -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof

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