November 2007 - ffmpeg-user - lists.ffmpeg.org

.m4a to .flv - image/video fails
by nreddig＠ptd.net 30 Mar '12

30 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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DVCPRO HD?
by stefan＠sfendt.de 16 Mar '08

16 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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Re-indexing wmv's via libavcodec/libavformat
by stas.oskin＠gmail.com 09 Dec '07

09 Dec '07

Hi. Is it possible to re-index WMV's via libavcode/format? After creating wmv's file, I'm sometimes getting them with broken seeking (unable to seek in windows media player), and after inspection I noticed the index is missing or corrupted. Thanks in advance, Stas. P.S. Sorry for the incorrect post to ffmpeg-develop.

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How to mixer wave to media (video+audio) file??
by quoctoanpham＠gmail.com 03 Dec '07

03 Dec '07

I have question: i have one media (video is mpeg and audio is mp3 ) and one wave file. The sound of media is s1 and sound of wav file is s2 Now , i want to mixer sound s2 with s1. It is said that (extracted from ffmpeg-devel) i can do that by: Demux the audio from 'media' to wav. Use sox to audiomix the 2 wav (something ffmpeg cant do now). Use ffmpeg to remux the video with the final audio. But unlucky, i didnot see how to use (or command lines ) Demux , sox and remux by ffmpeg? Anyone can help me how to use these ?

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3GP video settings for a real phone.
by P.J.Campbell＠kainos.com 03 Dec '07

03 Dec '07

Morning/Afternoon/Evening all, I am doing some R&D involving 3G video applications. The platform I am using makes use of FFMpeg to encode the videos either on the fly or in batch in advance. The problem I am having is that any reception glitch at all causes massive distortion of the video which can really destroy what would otherwise be a good video. The settings I am using are usually something like: -vcodec h263 -s qcif -r=7.5 -qscale 12 -g 30 -ps 400 -b 50k -acodec pcm_mulaw -ac 1 -a 8000 -scplx_mask 0.1 -tcplx_mask 0.1 -qmin 4 -qmax 31 -i_qfactor 0.01 I have also used "start masking" on the video generation to only render a quarter frame for the first 4 frames to prevent bitrate spikes on the first key frame. The parameter that seems to effect the bandwidth a lot is "qscale", but my current project has a lot more full motion stuff and a lot more break up and adjusting qscale doesn't seem to be enough. I've been told the -i_qfactor determines the balance between key frames and intermediary frames, but I don't understand how it works or if it will help me. I am aware that more key frames will mean the break up is likely to last a shorter period at a cost of more bandwidth. With only 64kbit to play with for both audio and video, it's not easy this stuff. I am also concerned with the audio settings. As I understand it pcm_ulaw is not compressed and at 8bit, 1 channel, 8000Hz results in 64kbit, which should be my whole channel gone.... But yet audio works for the most part. Should I try with "libamr_nb" and -ab 12000 ? Another problem is determining if the breakup is me pushing the channel too far, or just reception errors. Is there a 3G emulator about I could use instead of wasting 50pence (GBP) a minute testing? Is there anyone around on the list with experience of live 3G video applications... ie. Through a real live media gateway with a dial in number? Someone who has tested working settings? Thanks for any advice, hints, suggestions. Paul

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Help with compiling ffmpeg
by wkunen＠hotmail.com 03 Dec '07

03 Dec '07

Hi,howdy! i compiled ffmpeg in cygwin,but faild,i'v atteched the process report about results of "./configure,make"in email,plz help me analyze,appreciate! warm regards! _________________________________________________________________ Windows Live Photo gallery ????????????????????????????? http://get.live.cn/product/photo.html

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Unable to convert from this au file to other formats (ac3, mp4 ... etc)
by richcollins＠gmail.com 03 Dec '07

03 Dec '07

I get Unsupported codec (id=0) for input stream when I use this windows build: http://arrozcru.no-ip.org/ffmpeg_builds/bin/ffmpeg-r10461-gpl-static- win32.tar.bz2 With the attached au file. Any ideas??

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Question what is not answered by FAQ
by rwimmer＠praktika.de 03 Dec '07

03 Dec '07

hello, I am working with FFMPEG and I have a problem, which is not answered by the FAQ. My Problem is I have a (one!) picture (JPG) and a audio stream (format = wave) and my job is to create a video file (format = avi). This video file will be created but the picture will be shown only one secound and not the whole length of the audio stream. So I watched the FFMPEG-Documentation (http://ffmpeg.mplayerhq.hu/ffmpeg-doc.html) and the FAQ (http://ffmpeg.mplayerhq.hu/faq.html#SEC3) but I do not find how to solve the problem. At the moment the ffmpeg-arguemnts looks like: ffmpeg -i PicturePath -i WavePath -ar 44100 -s 320x240 -ac 2 -r 25 -y VideoPathAVI; This line is written in C# Visual Studio .Net 2. I hope you could understand my english and hope that you could help my with my problem. Thanks Ronny Wimmer

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