January 2008 - ffmpeg-user - lists.ffmpeg.org

.m4a to .flv - image/video fails
by nreddig＠ptd.net 31 Mar '12

31 Mar '12

Hi all, I have greatly appreciated the information I have already gleaned from reading this forum. However I still can not solve my current problem. I am trying to convert from .m4a to .flv. These are podcast files (.m4a) built in garage band. Pictures were added as well and we would like to use this format to create weekly announcements because Garage Band is so easy for our non-technically oriented staff. Using ffmpeg the audio converts just fine but the "video" is not right. I guess it's because these files include chapter data not actual video. However when I convert it to any format the images fly through at different rates, then it seems to get stuck on a picture in the middle. The format I would like to convert to is flash so that I can easy display multiple announcement files using a flash player. I can convert .mov to .flv no problem. Converting .m4a to .mov and then .flv doesn't produce any better results. Here are the before and after files: http://www.warwicksd.org/warwick.m4a http://www.warwicksd.org/output.flv I used the following to get my conversion: ffmpeg -i warwick.m4a -b 300k -r 12 -s 300x300 -ar 22050 -y -ab 96k output.flv FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-libfaac --enable-libfaad --enable- libmp3lame --disable-vhook --enable-shared libavutil version: 49.5.0 libavcodec version: 51.44.0 libavformat version: 51.14.0 built on Oct 22 2007 15:31:25, gcc: 3.3 20030304 (Apple Computer, Inc. build 1640) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'warwick.m4a': Duration: 00:00:55.9, start: 0.000000, bitrate: 100 kb/s Stream #0.0(eng): Audio: mpeg4aac, 44100 Hz, stereo Stream #0.1(eng): Video: mjpeg, yuvj420p, 300x300, 44100.00 fps(r) Stream #0.2(eng): Data: tx3g / 0x67337874 Stream #0.3(eng): Data: text / 0x74786574 Output #0, flv, to 'output.flv': Stream #0.0(eng): Video: flv, yuv420p, 300x300, q=2-31, 300 kb/s, 12.00 fps (c) Stream #0.1(eng): Audio: libmp3lame, 22050 Hz, stereo, 96 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 83 fps= 21 q=2.0 Lsize= 1099kB time=6.9 bitrate=1301.6kbits/s video:70kB audio:657kB global headers:0kB muxing overhead 51.296550% .mov files convert nicely to .flv but these .m4as are just not working. Our current staff is very eager to use these Garage Band files to communicate announcements. I have been so desperate to get this particular file type to convert that I have been experimenting with grabbing the frames every 5 secs. and then rebuilding the images into video. This produces strange results and seems to be more programming then should be necessary. Thanks in advanced for any solutions you can provide. Nancy Reddig Web Programmer Warwick SD

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resize and imlib2 vhook in one command ?
by code＠u-turnmedia.com 02 Oct '08

02 Oct '08

hi, i would like to resize video and overlay with image. when i resize video, e.g. from 624x352 to 320x240, and use imlib2 vhook in one line, i get 320x240 video with small picture in the corner: ffmpeg -i infile.avi -cropright 52 -cropleft 52 -s 320x240 -aspect 1.333 -vhook './imlib2.so -x 0 -y 0 -i image.png' outfile.avi i know that options for imlib2 are -x -y for coordinates, and i would like to know how to use """ Expressions are functions of these variables: N frame number (starting at zero) H frame height W frame width h image height w image width X previous x coordinate of text or image Y previous y coordinate of text or image """ precisely, how do i tell to the image to keep the same size ? do i have to use two commands ? like in first command resize video, and in second use vhook ? thanks Co.De. --

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DVCPRO HD?
by stefan＠sfendt.de 17 Mar '08

17 Mar '08

Hi, if I understand http://www.dcine.com/sightings.html correctly, there allready is DVCPRO HD support in ffmpeg since march 2007? As my current SVN-checkout does not like it nonetheless, do I need to issue any special configure options to enable that codec? Or doesn't ffmpeg support it? cu Stefan

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av_interleaved_write_frame(): Error while opening file
by gfrankliu＠gmail.com 04 Mar '08

04 Mar '08

ffmpeg -i 08112007007.mp4 -ar 22050 -f flv -y frank.flv FFmpeg version SVN-r10986, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include/ffmpeg --enable-libmp3lame --enable-libogg --enable-libvorbis --enable-libfaad --enable-libfaac --enable-pp --enable-shared --enable-pthreads --enable-gpl --enable-libamr-nb --enable-libamr-wb --disable-strip libavutil version: 49.5.0 libavcodec version: 51.48.0 libavformat version: 51.18.0 built on Nov 9 2007 17:42:28, gcc: 4.1.1 20070105 (Red Hat 4.1.1-52) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '08112007007.mp4': Duration: 00:00:07.2, start: 0.000000, bitrate: 2504 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 640x480 [PAR 1:1 DAR 4:3], 30000.00 fps(r) Stream #0.1(und): Audio: mpeg4aac, 48000 Hz, stereo Output #0, flv, to 'frank.flv': Stream #0.0(und): Video: flv, yuv420p, 640x480 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 30000.00 fps(c) Stream #0.1(und): Audio: libmp3lame, 22050 Hz, stereo, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding error, non monotone timestamps 1 >= 1 av_interleaved_write_frame(): Error while opening file Test file can be found at http://timepassindia.com/08112007007.mp4 Any thoughts? Thanks! Frank

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AVCodecContext->frame_size & Interleave Encoding
by kyle.mallory＠utah.edu 16 Feb '08

16 Feb '08

I'm trying to write a generic encoder (generic = encode multiple formats, codecs), but am using dv/dvvideo as my current test codec. Borrowing code from ffmpeg.c and apiexample.c, I am generating a sample buffer of size avctx->frame_size * sample_size * numchannels. However, pcm_encode_init():107 sets avctx->frame_size = 1. What should the frame-size for dv/pcm_s16le be, and how can I calculate the correct frame-size for different codecs? Also, are there any good examples/tutorials/documentation on how to write interleaved frames when encoding? I've search everywhere and can only seem to find the occasional example of encoding video or encoding audio, but nothing that encodes both, and uses libavformat to interleave them together. thanks, Kyle

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Pts problem with variable framerate mpeg4 video in mpegts stream
by M.P.vanGalen＠student.tudelft.nl 16 Feb '08

16 Feb '08

On 2008.01.05 01:31, ffmpeg-user-bounces at mplayerhq.hu wrote: > Your mail to 'ffmpeg-user' with the subject > > Pts problem with variable framerate mpeg4 video in mpegts stream > > Is being held until the list moderator can review it for approval. > > The reason it is being held: > > Message body is too big: 48950 bytes with a limit of 40 KB > Well this was more than 2 weeks ago, so I am resending this mail with a few attachments left out. Though I wonder if there is anybody reading the list-moderator email... Hello list, I am trying to get a mpegts stream which contains a variable framerate mpeg4 stream working, but have trouble playing it back. To demonstrate the problem I am encountering (or my misunderstanding of the API), I made a few adjustments to output_example.c from the ffmpeg svn version of today (diff attached). I am trying to achieve the following: I want to create a mpegts stream, with an audiostream, with constant sample-rate and a mpeg4 videostream with an average framerate of 25fps. For my application this videostream needs to have ms-accuracy timestamps which correlate with the timestamps of the audio-frame-packets. Therefore I changed time_base to 1/1000 and set r_frame_rate to 25. I also added a printf to show that the pkt.pts values are as I would expect. (I increment the video.pts with 45 instead of 40 to simulate an slower than average stream) Now if I play the generated file with ffplay, the results are not as I would expect. The video plays to fast, stutters and the bitrate has automagically changed (suddenly a lot of compression artifacts). I also added some printf's to ffplay (diff also attached) to check what is going on and allthough the video-pts values are correct, most of the pkt.pts values are AV_NOPTS_VALUE, which seems incorrect to me. output_example.c + output_example.orig.c.diff -> ffplay.orig.output output_example.c + output_example.c.diff -> output_example.output & ffplay.output Could someone provide me with some pointers about what is going on, am I doing something wrong or have I hit a bug in ffmpeg? Thanks in advance, Marijn

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[qt-faststart] Ever get "could not allocate 0x14 byte for ftyp atom"?
by kevin.t.campbell＠gmail.com 12 Feb '08

12 Feb '08

Hello, I am checking with people to see if anyone has ever gotten this error after running qt-faststart. I am pulling my hair out trying to figure out what the problem may be and wondered if anyone knew of a starting point. I'll take anything! *Error:* *could not allocate 0x14 byte for ftyp atom* *System: *Linux (centos), gcc ~3.4, latest ffmpeg, latest qt-faststart, basically all codec libraries (the movie comes out great with FFMPEG, it just wont stream because I fear the MOOV atom is out of place). *Example FFMPEG: First Pass* ffmpeg -i input.avi -an -pass 1 -threads 2 -vcodec libx264 -b 512k -bf 3 -subq 6 -cmp 256 -refs 5 -qmin 10 -qmax 51 -qdiff 4 -coder 1 -loop 1 -me hex -me_range 16 -trellis 1 -flags +mv4 -flags2 +bpyramid+wpred+mixed_refs+brdo+8x8dct -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 output.mp4 *Second Pass:* ffmpeg -i input.avi -acodec libfaac -ar 44100 -ab 160k -pass 2 -threads 2 -vcodec libx264 -b 512k -bf 3 -subq 6 -cmp 256 -refs 5 -qmin 10 -qmax 51 -qdiff 4 -coder 1 -loop 1 -me hex -me_range 16 -trellis 1 -flags +mv4 -flags2 +bpyramid+wpred+mixed_refs+brdo+8x8dct -partitions parti4x4+parti8x8+partp4x4+partp8x8+partb8x8 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 output.mp4 *QT-FASTSTART: *qt-faststart output.mp4 newOutput.mp4 Please let me know if anything else is needed. Kevin

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Terrible framerates (muxing overhead?) for x11grab + oss
by chris.lasher＠gmail.com 09 Feb '08

09 Feb '08

I'm getting really lousy framerates for my screen capture using x11grab in conjunction with capture from /dev/dsp. chris at feathers:~/downloads/ffmpeg$ ffmpeg -f x11grab -s svga -i :0.0 -f oss -i /dev/dsp -qscale 2 -vcodec mpeg4 -ab 32000 -ar 22050 /tmp/out.mp4 FFmpeg version SVN-r11163, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-x11grab --enable-pp --enable-pthreads --enable-libfaac --enable-libfaad --enable-liba52 --enable-libx264 --enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis libavutil version: 49.5.0 libavcodec version: 51.48.0 libavformat version: 52.1.0 built on Dec 4 2007 14:34:36, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2) [x11grab @ 0x8425814]device: :0.0 -> display: :0.0 x: 0 y: 0 width: 800 height: 600 [x11grab @ 0x8425814]shared memory extension found Input #0, x11grab, from ':0.0': Duration: N/A, start: 1196798006.303876, bitrate: 384000 kb/s Stream #0.0: Video: rawvideo, rgb32, 800x600 [PAR 0:1 DAR 0:1], 384000 kb/s, 25.00 tb(r) Input #1, oss, from '/dev/dsp': Duration: N/A, start: 1196798006.325148, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 44100 Hz, mono, 705 kb/s File '/tmp/out.mp4' already exists. Overwrite ? [y/N] y Output #0, mp4, to '/tmp/out.mp4': Stream #0.0: Video: mpeg4, yuv420p, 800x600 [PAR 0:1 DAR 0:1], q=2-31, 200 kb/s, 25.00 tb(c) Stream #0.1: Audio: libfaac, 22050 Hz, mono, 32 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 276 fps= 29 q=2.0 Lsize= 5283kB time=8.7 bitrate=4983.9kbits/s video:417kB audio:34kB global headers:0kB muxing overhead 1072.708355% Compare this to without capturing from /dev/dsp: chris at feathers:~/downloads/ffmpeg$ ffmpeg -f x11grab -s svga -i :0.0 -qscale 2 -vcodec mpeg4 /tmp/out.mp4 FFmpeg version SVN-r11163, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-x11grab --enable-pp --enable-pthreads --enable-libfaac --enable-libfaad --enable-liba52 --enable-libx264 --enable-libdc1394 --enable-libgsm --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis libavutil version: 49.5.0 libavcodec version: 51.48.0 libavformat version: 52.1.0 built on Dec 4 2007 14:34:36, gcc: 4.1.3 20070929 (prerelease) (Ubuntu 4.1.2-16ubuntu2) [x11grab @ 0x8425814]device: :0.0 -> display: :0.0 x: 0 y: 0 width: 800 height: 600 [x11grab @ 0x8425814]shared memory extension found Input #0, x11grab, from ':0.0': Duration: N/A, start: 1196797081.323056, bitrate: 384000 kb/s Stream #0.0: Video: rawvideo, rgb32, 800x600 [PAR 0:1 DAR 0:1], 384000 kb/s, 25.00 tb(r) File '/tmp/out.mp4' already exists. Overwrite ? [y/N] y Output #0, mp4, to '/tmp/out.mp4': Stream #0.0: Video: mpeg4, yuv420p, 800x600 [PAR 0:1 DAR 0:1], q=2-31, 200 kb/s, 25.00 tb(c) Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 370 fps= 32 q=2.0 Lsize= 20575kB time=14.8 bitrate=11388.6kbits/s video:18733kB audio:0kB global headers:0kB muxing overhead 9.830883% I notice a huge difference in the muxing overhead, which has probably caused this choppiness. How can I reduce the muxing overhead to a sane level? Thanks in advance, Chris

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length of quicktime displays as "1193:02:47" on ipod
by olav.morkrid＠gmail.com 06 Feb '08

06 Feb '08

hello i use ffmpeg to convert some video for ipod use. the video plays fine, but the ipod shows the time length of the video as "1193:02:47", which converted to milliseconds equals $FFFFFFFF in hexadecimal. my command line is as follows: ffmpeg -y -i "in.avi" -f mov -vcodec mpeg4 -b 1000k -acodec aac -ar 44100 -ab 128 "out.m4v" can anyone tell me how to make the ipod show the correct length?

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MPEG-2 transport stream and PCR
by linux.kernel＠free.fr 05 Feb '08

05 Feb '08

Hello, I'm trying to get ffmpeg to compress a 440 Hz tone into an MPEG-2 transport stream. I generated a raw PCM tone with the following code. (Based on the examples provided with ffmpeg.) => 48 kHz sampling frequency, 16 bits per sample per channel, same samples on left and right channels. #define _GNU_SOURCE #include <unistd.h> #include <stdint.h> #include <fcntl.h> #include <math.h> int main(void) { const double pi = 3.14159265358979323846; const double incr = 2 * pi * 440 / 48000; double x = 0; int i, j, fd; int16_t samples[1152 * 2]; if ((fd = open("tone.sw", O_WRONLY | O_CREAT, 0644)) < 0) return 1; for (i=0; i < 2000; ++i) { for (j=0; j < 1152; ++j) { int16_t temp = lrint(12000*sin(x)); samples[2*j+0] = temp; samples[2*j+1] = temp; x += incr; } write(fd, samples, sizeof samples); } return 0; } My first test was to encode to MPEG-1 Audio layer II. $ ffmpeg -ar 48000 -ac 2 -i tone.sw -ab 128000 foo.mp2 FFmpeg version SVN-r11555 configuration: --enable-gpl --cpu=pentium3 --enable-debug libavutil version: 49.6.0 libavcodec version: 51.49.0 libavformat version: 52.4.0 libavdevice version: 52.0.0 built on Jan 24 2008 13:46:39, gcc: 3.4.6 Input #0, s16le, from 'tone.sw': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Output #0, mp2, to 'foo.mp2': Stream #0.0: Audio: mp2, 48000 Hz, stereo, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 750kB time=48.0 bitrate= 128.0kbits/s video:0kB audio:750kB global headers:0kB muxing overhead 0.000000% (Did I provide the appropriate command line?) This seems to work as I can decode the file e.g. in Winamp. Then I tried to wrap the elementary stream in a transport stream. $ ffmpeg -ar 48000 -ac 2 -i tone.sw -ab 128000 foo.ts [...] Input #0, s16le, from 'tone.sw': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0.0: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Output #0, mpegts, to 'foo.ts': Stream #0.0: Audio: mp2, 48000 Hz, stereo, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 1043kB time=48.0 bitrate= 178.0kbits/s video:0kB audio:750kB global headers:0kB muxing overhead 39.041667% But when I load the resulting file in Tektronix's Transport Stream Compliance Analyzer (TSCA) it cannot even find the PAT... (Looking at the individual TS packets.) Packet 1: 47 40 00 11 00 00 b0 0d 00 01 c1 00 00 00 01 ef ff 36 90 e2 3d ff ff ff ff ff ff ff [...] I count 21 non-padding bytes (not a multiple of 4, hmmm). After the TS header (4 bytes) there should be 0x00 and the next nibble should be 1011 i.e. 0xb but in the above packet, it's 0x00 0x00 0xb... (i.e. an extra 0x00) I suppose I'm reading something wrong? What am I missing? Regards.

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