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• Delphi: Error when starting MCI

  - by marco92w

  I use the TMediaPlayer component for playing music. It works fine with most of my tracks. But it doesn't work with some tracks. When I want to play them, the following error message is shown: Which is German but roughly means that: In the project pMusicPlayer.exe an exception of the class EMCIDeviceError occurred. Message: "Error when starting MCI.". Process was stopped. Continue with "Single Command/Statement" or "Start". The program quits directly after calling the procedure "Play" of TMediaPlayer. This error occurred with the following file for example: file size: 7.40 MB duration: 4:02 minutes bitrate: 256 kBit/s I've encoded this file with a bitrate of 128 kBit/s and thus a file size of 3.70 MB: It works fine! What's wrong with the first file? Windows Media Player or other programs can play it without any problems. Is it possible that Delphi's TMediaPlayer cannot handle big files (e.g. 5 MB) or files with a high bitrate (e.g. 128 kBit/s)? What can I do to solve the problem?

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• Convert swf file to mp4 file using FFMPEG

  - by user1624004

  I now want to show an html5 video on a html page. Now I have an sample.swf file, I want to convert it to .mp4 or .ogg or .webm file. I have tried: ffmpeg -i sample.swf sample.mp4 But I got this error: [swf @ 0000000001feef40] Could not find codec parameters for stream 0 (Audio: pcm_s16le, 5512 Hz, 1 channels, 88 kb/s): unspecified sample format Consider increasing the value for the 'analyzeduration' and 'probesize' options [swf @ 0000000001feef40] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.0 : mono Input #0, swf, from 'sample.swf': Duration: N/A, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 5512 Hz, mono, 88 kb/s Stream #0:1: Video: mjpeg, yuvj444p, 1024x768 [SAR 100:100 DAR 4:3], 16 fps, 16 tbr, 16 tbn File 'sample.mp4' already exists. Overwrite ? [y/N] y Invalid sample format '(null)' Error opening filters!

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• ffmpeg video4linux2 at specified resolution

  - by wim

  When I'm trying to record a clip from my webcam, using: ffmpeg -f video4linux2 -s 640x480 -i /dev/video0 /tmp/spam.avi I get annoying problem with very low resolution video, and there is a message from ffmpeg saying: [video4linux2,v4l2 @ 0x2bff3e0] The V4L2 driver changed the video from 800x600 to 176x144 I have tried not specifying -s, or trying other sizes like 800x600, and always it forces me back to 176x144. Why is this and how can I prevent it? My webcam is one of those Logitech 9000 Pro, I know it supports better resolutions than this and I can see with v4l2-ctl --list-formats-ext that it goes up to at least 800x600. edit: complete console output follows wim@wim-desktop:~$ ffmpeg -f video4linux2 -s 640x480 -i /dev/video0 /tmp/spam.avi ffmpeg version git-2012-11-20-70c0f13 Copyright (c) 2000-2012 the FFmpeg developers built on Nov 21 2012 00:09:36 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-gpl --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 8.100 / 52. 8.100 libavcodec 54. 73.100 / 54. 73.100 libavformat 54. 37.100 / 54. 37.100 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 23.101 / 3. 23.101 libswscale 2. 1.102 / 2. 1.102 libswresample 0. 17.100 / 0. 17.100 libpostproc 52. 2.100 / 52. 2.100 [video4linux2,v4l2 @ 0x37a33e0] The V4L2 driver changed the video from 640x480 to 176x144 [video4linux2,v4l2 @ 0x37a33e0] Estimating duration from bitrate, this may be inaccurate Input #0, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 37066.740548, bitrate: 6082 kb/s Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 176x144, 6082 kb/s, 15 tbr, 1000k tbn, 15 tbc File '/tmp/spam.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to '/tmp/spam.avi': Metadata: ISFT : Lavf54.37.100 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 176x144, q=2-31, 200 kb/s, 15 tbn, 15 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> mpeg4) Press [q] to stop, [?] for help frame= 95 fps= 22 q=2.0 Lsize= 88kB time=00:00:13.86 bitrate= 51.8kbits/s video:77kB audio:0kB subtitle:0 global headers:0kB muxing overhead 13.553706%

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• wmp12 refuses to convert files when syncing

  - by Carbonara

  I have quite a large music collection of MP3s at 320kbps and some WMA files at various bitrates. I'm trying to sync some of them to my HTC Desire and am quickly running out of space. WMP12 has options to set, per device you wish to sync, to auto convert to a lower bitrate whilst syncing. I have set this to auto convert files to a maximum bitrate of 192kbps, that way I can fit more music on the device but keep the files on my PC at the higher rate. See these screens to see that it's set up correctly. Only problem is, surprise surprise for a Microsoft product, it doesn't actually work. Any file that is greater than 192kbps, MP3 or WMA simply fails, doesn't get converted or copied to the device. The message in the sync log displays the rather unhelpful message "error" and that's it. Any help would be appreciated. I'm not really looking for alternative software solutions I'd like to get this working since that's what it's supposed to do.

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• FFMPEG settings for Youtube and facebook video uploads

  - by eco_bach

  Can any FFMPEG experts share their preferred settings for video conversion to both Youtube and Facebook? For youtube I am following these guidelines and my video size is 480P @ 24 fps Audio Codec: AAC-LC Channels: Stereo or Stereo + 5.1<br> Sample rate 96khz or 48 khz<br> Video Codec: H.264 Progressive scan (no interlacing)<br> High Profile<br> 2 consecutive B frames<br> Closed GOP. GOP of half the frame rate.<br> CABAC<br> Variable bitrate. No bitrate limit required Color Space: 4.2.0 http://support.google.com/youtube/bin/static.py?hl=en&topic=1728573&guide=1728585&page=guide.cs

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• When spliting MP4s with ffmpeg how do I include metadata?

  - by Josh

  I have a few MP4s that i want to upload to my flickr account but they have a maximum size of 500mb as mine is only about 550 i was planing to simply split them in half then upload them, but i want to make sure all the meta data is included but it does not seem to be. I have tried each of the following with no luck, (at the end of this post i have the original and the new ffprobe outputs): ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_meta_data SANY0069.MP4:SANY0069A.MP4 SANY0069A.MP4 with the this one I manually produced the individual meta tags that i took from this command ffmpeg -i SANY0069A.MP4 -f ffmetadata meta.txt ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -metadata major_brand="mp42" -metadata minor_version="1" -metadata compatible_brands="mp42avc1" -metadata creation_time="2012-09-29 09:05:50" -metadata comment="SANYO DIGITAL CAMERA CA9" -metadata comment-eng="SANYO DIGITAL CAMERA CA9" SANY0069A.MP4 using the output of the former command i also tried this: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -f ffmetadata -i meta.txt SANY0069A.MP4 Output: sample output from my first command: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg version 0.8.12, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) 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' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 File 'SANY0069A.MP4' already exists. Overwrite ? [y/N] y Output #0, mp4, to 'SANY0069A.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 encoder : Lavf53.5.0 Stream #0.0(eng): Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 9007 kb/s, 30k tbn, 29.97 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help frame= 7773 fps=4644 q=-1.0 Lsize= 289607kB time=00:04:19.35 bitrate=9147.4kbits/s video:285416kB audio:4033kB global headers:0kB muxing overhead 0.054571% and finaly, when i compare the ffprobe of the original and the first split part i get the 2 following outputs: original ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) 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' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Split ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) 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' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069A.MP4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.5.0 comment : SANYO DIGITAL CAMERA CA9 Duration: 00:04:19.37, start: 0.000000, bitrate: 9146 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9015 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 1970-01-01 00:00:00 I know this is incredibly long but its actually a quite simple question. I thought it would be best to provide as much detail as possible. any advice here would be great, Thanks

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• Why can't I convert FLV to MP4 format using FFmpeg when MP3 works?

  - by hugemeow

  In fact I have succeeded to convert FLV to MP3: D:\tmp\ffmpeg-20121005-git-d9dfe9a-win64-static\ffmpeg-20121005-git-d9dfe9a-win 4-static\bin>ffmpeg.exe -i a.flv -acodec mp3 a.mp3 ffmpeg version N-45080-gd9dfe9a Copyright (c) 2000-2012 the FFmpeg developers built on Oct 5 2012 16:49:01 with gcc 4.7.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-run ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass -enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable- ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopen peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libthe ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-l bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --en ble-zlib libavutil 51. 73.102 / 51. 73.102 libavcodec 54. 63.100 / 54. 63.100 libavformat 54. 29.105 / 54. 29.105 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 19.102 / 3. 19.102 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 Input #0, flv, from 'a.flv': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 Duration: 00:08:49.73, start: 0.000000, bitrate: 255 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 448x336 [SAR 1:1 DAR 4:3], 218 kb s, 15 tbr, 1k tbn, 30 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 32 kb/s File 'a.mp3' already exists. Overwrite ? [y/N] y Output #0, mp3, to 'a.mp3': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 TSSE : Lavf54.29.105 Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16 Stream mapping: Stream #0:1 -> #0:0 (aac -> libmp3lame) Press [q] to stop, [?] for help size= 8279kB time=00:08:49.78 bitrate= 128.0kbits/s video:0kB audio:8278kB subtitle:0 global headers:0kB muxing overhead 0.006842% But I failed to convert FLV to MP4. Why is the encoder 'mp4' unknown? What's more, how can I find the codecs which are already supported by my FFmpeg? D:\tmp\ffmpeg-20121005-git-d9dfe9a-win64-static\ffmpeg-20121005-git-d9dfe9a-win6 4-static\bin>ffmpeg.exe -i a.flv -acodec mp4 aa.mp4 ffmpeg version N-45080-gd9dfe9a Copyright (c) 2000-2012 the FFmpeg developers built on Oct 5 2012 16:49:01 with gcc 4.7.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-l ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenj peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 51. 73.102 / 51. 73.102 libavcodec 54. 63.100 / 54. 63.100 libavformat 54. 29.105 / 54. 29.105 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 19.102 / 3. 19.102 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 Input #0, flv, from 'a.flv': Metadata: metadatacreator : iku hasKeyframes : true hasVideo : true hasAudio : true hasMetadata : true canSeekToEnd : false datasize : 16906383 videosize : 14558526 audiosize : 2270465 lasttimestamp : 530 lastkeyframetimestamp: 529 lastkeyframelocation: 16893721 Duration: 00:08:49.73, start: 0.000000, bitrate: 255 kb/s Stream #0:0: Video: h264 (Main), yuv420p, 448x336 [SAR 1:1 DAR 4:3], 218 kb/ s, 15 tbr, 1k tbn, 30 tbc Stream #0:1: Audio: aac, 44100 Hz, stereo, s16, 32 kb/s Unknown encoder 'mp4'

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• FFMPEG dropping frames while encoding JPEG sequence at color change

  - by Matt

  I'm trying to put together a slide show using imagemagick and FFMPEG. I use imagemagick to expand a single photo into 30fps video (imagemagick also handles things like putting some text captions on the frames along the way). When I go to let ffmpeg digest it into a video it clips along nicely on the color parts of the video, but when it gets to a black and white section it reports "frame= 2030 fps=102 q=32766.0 Lsize= 5203kB time=00:01:07.60 bitrate= 630.5kbits/s dup=0 drop=703" and drops every frame of video until it hits something with color. As you can imagine this results in entire photos being removed from the slideshow. Here is my latest dump... ffmpeg -y -r 30 -i "teststream/%06d.jpg" -c:v libx264 -r 30 newffmpeg.mp4 ffmpeg version git-2012-12-10-c3bb333 Copyright (c) 2000-2012 the FFmpeg developers built on Dec 10 2012 22:02:04 with gcc 4.6.1 (Ubuntu/Linaro 4.6.1-9ubuntu3) configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libx264 --enable-nonfree --enable-version3 libavutil 52. 12.100 / 52. 12.100 libavcodec 54. 79.101 / 54. 79.101 libavformat 54. 49.100 / 54. 49.100 libavdevice 54. 3.102 / 54. 3.102 libavfilter 3. 26.101 / 3. 26.101 libswscale 2. 1.103 / 2. 1.103 libswresample 0. 17.102 / 0. 17.102 libpostproc 52. 2.100 / 52. 2.100 Input #0, image2, from 'teststream/%06d.jpg': Duration: 00:12:02.80, start: 0.000000, bitrate: N/A Stream #0:0: Video: mjpeg, yuvj444p, 720x480 [SAR 72:72 DAR 3:2], 25 fps, 25 tbr, 25 tbn, 25 tbc [libx264 @ 0x3450140] using SAR=1/1 [libx264 @ 0x3450140] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x3450140] profile High, level 3.0 [libx264 @ 0x3450140] 264 - core 129 r2 1cffe9f - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'newffmpeg.mp4': Metadata: encoder : Lavf54.49.100 Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuvj420p, 720x480 [SAR 1:1 DAR 3:2], q=-1--1, 15360 tbn, 30 tbc Stream mapping: Stream #0:0 - #0:0 (mjpeg - libx264) Press [q] to stop, [?] for help Input stream #0:0 frame changed from size:720x480 fmt:yuvj444p to size:720x480 fmt:yuvj422p Input stream #0:0 frame changed from size:720x480 fmt:yuvj422p to size:720x480 fmt:yuvj444pp=584 frame= 2030 fps=102 q=32766.0 Lsize= 5203kB time=00:01:07.60 bitrate= 630.5kbits/s dup=0 drop=703 video:5179kB audio:0kB subtitle:0 global headers:0kB muxing overhead 0.472425% [libx264 @ 0x3450140] frame I:9 Avg QP:20.10 size: 33933 [libx264 @ 0x3450140] frame P:636 Avg QP:24.12 size: 6737 [libx264 @ 0x3450140] frame B:1385 Avg QP:27.04 size: 514 [libx264 @ 0x3450140] consecutive B-frames: 2.5% 15.2% 13.2% 69.2% [libx264 @ 0x3450140] mb I I16..4: 8.3% 80.3% 11.5% [libx264 @ 0x3450140] mb P I16..4: 1.5% 2.5% 0.2% P16..4: 41.7% 18.0% 10.3% 0.0% 0.0% skip:25.9% [libx264 @ 0x3450140] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 26.6% 0.6% 0.1% direct: 0.2% skip:72.3% L0:35.0% L1:60.3% BI: 4.7% [libx264 @ 0x3450140] 8x8 transform intra:64.1% inter:75.1% [libx264 @ 0x3450140] coded y,uvDC,uvAC intra: 51.6% 78.0% 43.7% inter: 10.6% 14.9% 2.1% [libx264 @ 0x3450140] i16 v,h,dc,p: 29% 19% 6% 46% [libx264 @ 0x3450140] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 23% 15% 17% 5% 9% 10% 7% 8% 6% [libx264 @ 0x3450140] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 31% 18% 11% 5% 9% 10% 6% 6% 4% [libx264 @ 0x3450140] i8c dc,h,v,p: 46% 18% 24% 12% [libx264 @ 0x3450140] Weighted P-Frames: Y:20.1% UV:18.7% [libx264 @ 0x3450140] ref P L0: 59.2% 23.2% 13.1% 4.3% 0.2% [libx264 @ 0x3450140] ref B L0: 88.7% 8.3% 3.0% [libx264 @ 0x3450140] ref B L1: 95.0% 5.0% [libx264 @ 0x3450140] kb/s:626.88 Received signal 2: terminating. One last note: If I remove the -r 30 from the input and output it works flawlessly. I have no idea why the -r 30 is causing it to freak out.

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• Ubuntu One Music Store: Tops or Flop?

  <b>Linux Pro Magazine:</b> "Thus Canonical is implementing the cloud in its context. The DRM-free songs in MP3 format with a bitrate of at least 256 kbits/second are not loaded on the local machine, but in the Ubuntu One cloud."

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• Digitize video?

  - by kire

  I got some movies on a VCR that i want to move to my computer somehow. (for personal use) I have a video capture card and all hardware required. I'm looking for the software - programs, codecs etc. I like the format that most torrents come in and got some of these i'd like to use as reference for comparison. How can i see what codec, bitrate, etc a movie is using so i can pick this and know that it will work and look good. For AVI-files the bitrate is visible in explorer but it doesn't mention the codec used and i also have a lot of MKV-files that explorer can't handle. All kinds of tips, tricks and other suggestions are welcome. This is completely new to me. How do I avoid that video/audio gets out of sync for example, many movies you download have audio out of sync so i guess this can happen quite easily. The encoding-program has to run on windows and for playback the movies should work on at least VLC for windows.

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• ffmpeg: converting an avi to a reduced, shareable flv/mp4...

  - by meder

  I recently followed a guide and recompiled my ffmpeg so x264 is enabled. I used some generic settings to convert my 700 MB avi file to a mp4 file, the result was a 407MB mp4 file. The original avi file's settings: Codec: DX50 Resolution: 704x304 Frame rate: 23.976023 Stream 1 Codec: mpga Type: Audio Channels: 2 Sample rate: 48000 Hz Bitrate 179 kb/s Command I used: ffmpeg -i input.avi -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -vpre hq -crf 22 -threads 0 output.mp4 The settings of the output file (output.mp4): Codec: avc1 Resolution: 704x304 Display resolution: 704x304 Frame rate: 11.988011 Stream 1 Codec: mp4a Type: Audio Channels: 2 Sample Rate: 48000 Hz Bits per sample: 16 Bitrate: 1536 kb/s The quality of the output mp4 is pretty nice, it seems as if it's pretty much the same as the original source. However, I'm trying to reduce the filesize and I'm not really sure whether I should go with an flv format or keep it mp4. The advantage the flv would have obviously is that it would be playable with a flash player ( I have come across some swf players which take a flash parameter to play an flv file ).. but maybe I could use the video element, as I'm only going to be displaying this video privately so I don't have to worry about supporting legacy browsers such as IE. Can someone recommend some settings to specify in order for the filesize to be around ~100-150MB or so? I don't mind a reduction in quality, nor do I mind resizing it - I was going to do it initially but I wasn't sure what the guidelines were ( if any ) for dealing with resolution.. since this video's is 704x304 would it still be ok if I forced it into one that isn't perfectly fit for the aspect ratio? I have no clue about that part. I realize that I could have probably specified 28 instead of 22 for the CRF, I'm not sure if I should do that as opposed to maybe specifying smaller resolution, which might make it smaller as well?

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• How do I convert a video to GIF using ffmpeg, with reasonable quality?

  - by Kamil Hismatullin

  I'm converting .flv movie to .gif file with ffmpeg. ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif It works great, but output gif file has a very law quality. Any ideas how can I improve quality of converted gif? Output of command: $ ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:52:53 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 Duration: 00:00:18.85, start: 0.000000, bitrate: 3098 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720, 2905 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 192 kb/s Metadata: creation_time : 2013-02-14 04:00:07 [buffer @ 0x92a8ea0] w:1280 h:720 pixfmt:yuv420p [scale @ 0x9215100] w:1280 h:720 fmt:yuv420p -> w:320 h:240 fmt:rgb24 flags:0x4 Output #0, gif, to 'output.gif': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 encoder : Lavf53.21.1 Stream #0.0(und): Video: rawvideo, rgb24, 320x240, q=2-31, 200 kb/s, 90k tbn, 10 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 101 fps= 32 q=0.0 Lsize= 8686kB time=10.10 bitrate=7045.0kbits/s dup=0 drop=149 video:22725kB audio:0kB global headers:0kB muxing overhead -61.778676% Thanks.

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• AVConv increases song duration when converting MP3

  - by chauffch

  I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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• ffmpeg cutting video duration

  - by Steve Spence

  When using ffmpeg on linux, my 4.3GB 2.21 second video is being chopped down to 1.56 duration. I'm trying to reduce file size, but not lose frames. steve@steve-OptiPlex-170L:~/Desktop$ ffmpeg -i microbe.avi microbe.mp4 ffmpeg version 0.8.3-4:0.8.3-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Jun 12 2012 16:37:58 with gcc 4.6.3 * THIS PROGRAM IS DEPRECATED * This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, avi, from 'microbe.avi': Duration: 00:02:21.80, start: 0.000000, bitrate: 242311 kb/s Stream #0.0: Video: rawvideo, bgr24, 1280x960, 10 tbr, 10 tbn, 10 tbc Incompatible pixel format 'bgr24' for codec 'mpeg4', auto-selecting format 'yuv420p' [buffer @ 0x9f861e0] w:1280 h:960 pixfmt:bgr24 [avsink @ 0x9f86440] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x9f7d800] w:1280 h:960 fmt:bgr24 - w:1280 h:960 fmt:yuv420p flags:0x4 Output #0, mp4, to 'microbe.mp4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: mpeg4, yuv420p, 1280x960, q=2-31, 200 kb/s, 10 tbn, 10 tbc Stream mapping: Stream #0.0 - #0.0 Press ctrl-c to stop encoding frame= 1164 fps= 6 q=31.0 Lsize= 3775kB time=116.40 bitrate= 265.7kbits/s video:3765kB audio:0kB global headers:0kB muxing overhead 0.272870% steve@steve-OptiPlex-170L:~/Desktop$

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• FFmpeg audio dont work in converted videos

  - by Juddy Swaft

  NOTICE: when i convert videos via terminal and download them from ftp into pc the audio works fine. I use: if($ext == "avi" && $convert_avi == true) { $convert_source = _VIDEOS_DIR_PATH.$new_name; $conv_name = substr(md5($file['name'].rand(1,888)), 2, 10).".mp4"; $converted_file = _VIDEOS_DIR_PATH.$conv_name; $ffmpeg_command = 'ffmpeg -i '.$convert_source.' -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 '.$converted_file; echo exec($ffmpeg_command); $sql = "UPDATE pm_temp SET url = '".$conv_name."' WHERE url = '".$new_name."' LIMIT 1"; $result = @mysql_query($sql); unlink($convert_source); } This code to convert avi to mp4 ffmpeg concole output: root@1tb:~# ffmpeg -i sample.avi -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 goodsample.mp4 ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers built on Feb 22 2013 07:18:58 with gcc 4.4.5 configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-encoder=libschroedinger - s libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 0x191d4100] Header missing [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'sample.avi': Metadata: encoder : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:01:01.81, start: 0.000000, bitrate: 1194 kb/s Stream #0.0: Video: mpeg4, yuv420p, 640x352 [PAR 1:1 DAR 20:11], 23.98 tbr, Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s [buffer @ 0x191d1c80] w:640 h:352 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x191d6880] w:640 h:352 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0 [libx264 @ 0x191ce5a0] Default settings detected, using medium profile [libx264 @ 0x191ce5a0] using SAR=45/44 [libx264 @ 0x191ce5a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle S [libx264 @ 0x191ce5a0] profile High, level 3.1 [libx264 @ 0x191ce5a0] 264 - core 118 - H.264/MPEG-4 AVC codec - Copyleft 2003-2 6 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_off 1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_l Output #0, mp4, to 'goodsample.mp4': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 45:44 DAR 20:11], q=2-31 Stream #0.1: Audio: libmp3lame, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help [mp3 @ 0x191d4100] Header missing Error while decoding stream #0.1 [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected [mp3 @ 0x191d4100] incomplete frame 9467kB time=00:01:00.32 bitrate=1285.5kbits/ Error while decoding stream #0.1 frame= 1852 fps= 20 q=29.0 Lsize= 9652kB time=00:01:01.72 bitrate=1280.9kbits video:9121kB audio:483kB global headers:0kB muxing overhead 0.499688% frame I:11 Avg QP:16.78 size: 51456 [libx264 @ 0x191ce5a0] frame P:784 Avg QP:20.81 size: 8954 [libx264 @ 0x191ce5a0] frame B:1057 Avg QP:26.06 size: 1659 [libx264 @ 0x191ce5a0] consecutive B-frames: 22.0% 3.1% 7.5% 67.4% [libx264 @ 0x191ce5a0] mb I I16..4: 31.1% 59.8% 9.1% [libx264 @ 0x191ce5a0] mb P I16..4: 1.8% 2.6% 0.2% P16..4: 24.3% 7.0% 4.0 [libx264 @ 0x191ce5a0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 22.7% 0.8% 0.2 [libx264 @ 0x191ce5a0] 8x8 transform intra:57.0% inter:72.6% [libx264 @ 0x191ce5a0] coded y,uvDC,uvAC intra: 44.4% 33.3% 10.3% inter: 7.6% 5. [libx264 @ 0x191ce5a0] i16 v,h,dc,p: 68% 14% 8% 10% [libx264 @ 0x191ce5a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 27% 5% 7% 7% 6 [libx264 @ 0x191ce5a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 14% 14% 6% 10% 9% 7 [libx264 @ 0x191ce5a0] i8c dc,h,v,p: 67% 13% 17% 3% [libx264 @ 0x191ce5a0] Weighted P-Frames: Y:1.9% UV:0.4% [libx264 @ 0x191ce5a0] ref P L0: 62.2% 12.8% 10.3% 14.5% 0.2% [libx264 @ 0x191ce5a0] ref B L0: 88.1% 5.5% 6.4% [libx264 @ 0x191ce5a0] ref B L1: 95.7% 4.3% [libx264 @ 0x191ce5a0] kb/s:1209.03 I know there is couple errors tough, but i dont know hot to fix it. Also i would be very thankfull if someone can help reduce video size but is not main problem video weights as original avi but sill.

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• Convert old AVI files to a modern format

  - by iWerner

  Hi, we have a collection of old home videos that were saved in AVI format a long time ago. I want to convert these files to a more modern format because the Totem Movie Player that comes with Ubuntu 10.4 seems to be the only program capable of playing them. The files seem to be encoded with a MJPEG codec, and playing them in VLC or Windows Media Player plays only the sound but there is no video. Avidemux was able to open the files, but the quality of the video is severely degraded: The video skips frames and is interlaced (it's not interlaced when playing it in Totem). Neither ffmpeg nor mencoder seems to be able to read the video stream. mencoder reports that it is using ffmpeg's codec. Here's a section from its output: ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family [mjpeg @ 0x92a7260]mjpeg: using external huffman table [mjpeg @ 0x92a7260]mjpeg: error using external huffman table, switching back to internal Unsupported PixelFormat -1 Selected video codec: [ffmjpeg] vfm: ffmpeg (FFmpeg MJPEG) while running ffmpeg produces the following: $ ffmpeg -i input.avi output.avi FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 4 2010 12:35:30, gcc: 4.4.3 [avi @ 0x87952c0]non-interleaved AVI Input #0, avi, from 'input.avi': Duration: 00:00:15.24, start: 0.000000, bitrate: 22447 kb/s Stream #0.0: Video: mjpeg, yuvj422p, 720x544, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 720x544, q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: mp2, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 0 fps= 0 q=0.0 Lsize= 143kB time=15.23 bitrate= 76.9kbits/s video:0kB audio:119kB global headers:0kB muxing overhead 20.101777% So the problem is that output does not contain any video, as evidenced by the video:0kB at the end. In all of the above cases the audio comes out fine. So my question is: What can I do to convert these files to a more modern format with more modern codecs?

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• Altruistic network connection bandwidth estimation

  - by datenwolf

  Assume two peers Alice and Bob connected over a IP network. Alice and Bob are exchanging packets of lossy compressed data which are generated and to be consumes in real time (think a VoIP or video chat application). The service is designed to cope with as little bandwidth available, but relies on low latencies. Alice and Bob would mark their connection with an apropriate QoS profile. Alice and Bob want use a variable bitrate compression and would like to consume all of the leftover bandwidth available for the connection between them, but would voluntarily reduce the consumed bitrate depending on the state of the network. However they'd like to retain a stable link, i.e. avoid interruptions in their decoded data stream caused by congestion and the delay until the bandwidth got adjusted. However it is perfectly possible for them to loose a few packets. TL;DR: Alice and Bob want to implement a VoIP protocol from scratch, and are curious about bandwidth and congestion control. What papers and resources do you suggest for Alice and Bob to read? Mainly in the area of bandwidth estimation and congestion control.

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• How to get the real, actual duration of an MP3 file (VBR or CBR) server-side

  - by Cummander Checkov

  I used to calculate the duration of MP3 files server-side using ffmpeg - which seemed to work fine. Today i discovered that some of the calculations were wrong. Somehow, for some reason, ffmpeg will miscalculate the duration and it seems to happen with variable bit rate mp3 files only. When testing this locally, i noticed that ffmpeg printed two extra lines in green. Command used: ffmpeg -i song_9747c077aef8.mp3 ffmpeg says: [mp3 @ 0x102052600] max_analyze_duration 5000000 reached at 5015510 [mp3 @ 0x102052600] Estimating duration from bitrate, this may be inaccurate After a nice, warm google session, i found some posts on this, but no solution was found. I then tried to increase the maximum duration: ffmpeg -analyzeduration 999999999 -i song_9747c077aef8.mp3 After this, ffmpeg returned only the second line: [mp3 @ 0x102052600] Estimating duration from bitrate, this may be inaccurate But in either case, the calculated duration was just plain wrong. Comparing it to VLC i noticed that here the duration is correct. After more research i stumbled over mp3info - which i installed and used. mp3info -p "%S" song_9747c077aef8.mp3 mp3info then returned the CORRECT duration, but only as an integer, which i cannot use as i need a more accurate number here. The reason for this was explained in a comment below, by user blahdiblah - mp3info is simply pulling ID3 info from the file and not actually performing any calculations. I also tried using mplayer to retrieve the duration, but just as ffmpeg, mplayer is returning the wrong value. Now i ran out of options. If somebody knows how to get around this, any hints, tips, guides or corrections are welcome! Thank You!

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• ffmpeg hangs when creating a video

  - by FearUs

  I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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• Is PLC speed affected by mixing different devices?

  - by CFP

  Hello everyone! At home, I have 4 PLC devices for my home network. Two of them are 85Mb/s powerlan PLC adapters, while the others are 10Mbps powerlan PLC adapters. I have not been able to determine reliably whether the presence of the 10Mb/s ones impact on the speed of the 85Mb/s ones. Is it possible that the bitrate is limited by the slowest devices on the network? Thanks!

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• Concatenation of a 2 second silence audio with a normal audio not working

  - by user1665130

  I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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• How do you splice out a part of an xvid encoded avi file, with ffmpeg? (no problems with other files

  - by yegor

  Im using the following command, which works for most files, except what seems to be xvid encoded ones /usr/bin/ffmpeg -sameq -i file.avi -ss 00:01:00 -t 00:00:30 -ac 2 -r 25 -copyts output.avi So this should basically splice out 30 seconds of video + audio, starting from 1 minute mark. It does START encoding at the 00:01:00 mark but it goes all the way to the end of the file for some reason, ignoring that I want just 30 seconds. The output looks like this. FFmpeg version git-ecc4bdd, Copyright (c) 2000-2010 the FFmpeg developers built on May 31 2010 04:52:24 with gcc 4.4.3 20100127 (Red Hat 4.4.3-4) configuration: --enable-libx264 --enable-libxvid --enable-libmp3lame --enable-libopenjpeg --enable-libfaac --enable-libvorbis --enable-gpl --enable-nonfree --enable-libxvid --enable-pthreads --enable-libfaad --extra-cflags=-fPIC --enable-postproc --enable-libtheora --enable-libvorbis --enable-shared libavutil 50.15. 2 / 50.15. 2 libavcodec 52.67. 0 / 52.67. 0 libavformat 52.62. 0 / 52.62. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.20. 0 / 1.20. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'file.avi': Metadata: ISFT : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:02:00.00, start: 0.000000, bitrate: 1587 kb/s Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s File 'lol6.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to 'lol6.avi': Metadata: ISFT : Lavf52.62.0 Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected [buffer @ 0x184b610]Buffering several frames is not supported. Please consume all available frames before adding a new one. frame= 1501 fps=104 q=0.0 Lsize= 15612kB time=30.02 bitrate=4259.7kbits/s ts/s video:15303kB audio:235kB global headers:0kB muxing overhead 0.482620% if I convert this file to mp4 for example, and then perform the same action, it works perfectly.

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• How do you splice out a part of an xvid encoded avi file, with ffmpeg? (no problems with other files)

  - by user11955

  Im using the following command, which works for most files, except what seems to be xvid encoded ones /usr/bin/ffmpeg -sameq -i file.avi -ss 00:01:00 -t 00:00:30 -ac 2 -r 25 -copyts output.avi So this should basically splice out 30 seconds of video + audio, starting from 1 minute mark. It does START encoding at the 00:01:00 mark but it goes all the way to the end of the file for some reason, ignoring that I want just 30 seconds. The output looks like this. FFmpeg version git-ecc4bdd, Copyright (c) 2000-2010 the FFmpeg developers built on May 31 2010 04:52:24 with gcc 4.4.3 20100127 (Red Hat 4.4.3-4) configuration: --enable-libx264 --enable-libxvid --enable-libmp3lame --enable-libopenjpeg --enable-libfaac --enable-libvorbis --enable-gpl --enable-nonfree --enable-libxvid --enable-pthreads --enable-libfaad --extra-cflags=-fPIC --enable-postproc --enable-libtheora --enable-libvorbis --enable-shared libavutil 50.15. 2 / 50.15. 2 libavcodec 52.67. 0 / 52.67. 0 libavformat 52.62. 0 / 52.62. 0 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.20. 0 / 1.20. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'file.avi': Metadata: ISFT : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:02:00.00, start: 0.000000, bitrate: 1587 kb/s Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s File 'lol6.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to 'lol6.avi': Metadata: ISFT : Lavf52.62.0 Stream #0.0: Video: mpeg4, yuv420p, 672x368 [PAR 1:1 DAR 42:23], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [mpeg4 @ 0x17cf770]Invalid and inefficient vfw-avi packed B frames detected [buffer @ 0x184b610]Buffering several frames is not supported. Please consume all available frames before adding a new one. frame= 1501 fps=104 q=0.0 Lsize= 15612kB time=30.02 bitrate=4259.7kbits/s ts/s video:15303kB audio:235kB global headers:0kB muxing overhead 0.482620% if I convert this file to mp4 for example, and then perform the same action, it works perfectly.

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• Is power line communication (PLC) speed affected by mixing different devices?

  - by CFP

  At home, I have 4 PLC devices for my home network. Two of them are 85 Mb/s PowerLAN PLC adapters, while the others are 10 Mbps PowerLAN PLC adapters. I have not been able to determine reliably whether the presence of the 10 Mb/s ones impact on the speed of the 85 Mb/s ones. Is it possible that the bitrate is limited by the slowest devices on the network?

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• reencode several videos with virtualdub?

  - by acidzombie24

  I have about 50 small videos (and a few large videos). I want to convert them all with the SAME settings. Its basically change audio to X with Y bitrate, change video to xvid. and do full processing on the video and audio. Then force the FPS to 15 since every program i tried (including virtualdub) thinks it 0.3 FPS. How do i apply all of these settings to all of my files?

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