Search Results

Search found 1211 results on 49 pages for 'stereo to mono'.

Page 24/49 | < Previous Page | 20 21 22 23 24 25 26 27 28 29 30 31  | Next Page >

• 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

  Read the article

• Issues with ACTION_HEADSET_PLUG broadcast in Android

  - by Denis M

  I've tried these phones: Moto Backflip 1.5, Nexus One 2.1 Basically I register BroadcastReceiver to get ACTION_HEADSET_PLUG broadcast and look on 3 extras that come in intent: state name microphone Here is the description from API: * state - 0 for unplugged, 1 for plugged. * name - Headset type, human readable string * microphone - 1 if headset has a microphone, 0 otherwise Issue #1: Broadcast comes when activity is started (not expected), when screen rotation happens (not expected) and when headset/headphones plugged/unplugged (expected). Issue #2: Backflip phone sends null for state + microphone, 'No Device' as name when headset/headphones unplugged, and sends null for state + microphone, 'Stereo HeadSet'/'Stereo HeadPhones' as name when headset/headphones plugged. Nexus even worse, it always sends null for state + microphone, 'Headset' as name when headset/headphones plugged or unplugged. Question: How it can be explained that API is broken so much on both 1.5 and 2.1 versions and different devices, manufactures?

  Read the article

• Banshee encountered a Fatal Error (sqlite error 11: database disk image is malformed)

  - by Nik

  I am running ubuntu 10.10 Maverick Meerkat, and recently I am helping in testing out indicator-weather using the unstable buids. However there was a bug which caused my system to freeze suddenly (due to indicator-weather not ubuntu) and the only way to recover is to do a hard reset of the system. This happened a couple of times. And when i tried to open banshee after a couple of such resets I get the following fatal error which forces me to quit banshee. The screenshot is not clear enough to read the error, so I am posting it below, An unhandled exception was thrown: Sqlite error 11: database disk image is malformed (SQL: BEGIN TRANSACTION; DELETE FROM CoreSmartPlaylistEntries WHERE SmartPlaylistID IN (SELECT SmartPlaylistID FROM CoreSmartPlaylists WHERE IsTemporary = 1); DELETE FROM CoreSmartPlaylists WHERE IsTemporary = 1; COMMIT TRANSACTION) at Hyena.Data.Sqlite.Connection.CheckError (Int32 errorCode, System.String sql) [0x00000] in <filename unknown>:0 at Hyena.Data.Sqlite.Connection.Execute (System.String sql) [0x00000] in <filename unknown>:0 at Hyena.Data.Sqlite.HyenaSqliteCommand.Execute (Hyena.Data.Sqlite.HyenaSqliteConnection hconnection, Hyena.Data.Sqlite.Connection connection) [0x00000] in <filename unknown>:0 Exception has been thrown by the target of an invocation. at System.Reflection.MonoCMethod.Invoke (System.Object obj, BindingFlags invokeAttr, System.Reflection.Binder binder, System.Object[] parameters, System.Globalization.CultureInfo culture) [0x00000] in <filename unknown>:0 at System.Reflection.MonoCMethod.Invoke (BindingFlags invokeAttr, System.Reflection.Binder binder, System.Object[] parameters, System.Globalization.CultureInfo culture) [0x00000] in <filename unknown>:0 at System.Reflection.ConstructorInfo.Invoke (System.Object[] parameters) [0x00000] in <filename unknown>:0 at System.Activator.CreateInstance (System.Type type, Boolean nonPublic) [0x00000] in <filename unknown>:0 at System.Activator.CreateInstance (System.Type type) [0x00000] in <filename unknown>:0 at Banshee.Gui.GtkBaseClient.Startup () [0x00000] in <filename unknown>:0 at Hyena.Gui.CleanRoomStartup.Startup (Hyena.Gui.StartupInvocationHandler startup) [0x00000] in <filename unknown>:0 .NET Version: 2.0.50727.1433 OS Version: Unix 2.6.35.27 Assembly Version Information: gkeyfile-sharp (1.0.0.0) Banshee.AudioCd (1.9.0.0) Banshee.MiniMode (1.9.0.0) Banshee.CoverArt (1.9.0.0) indicate-sharp (0.4.1.0) notify-sharp (0.4.0.0) Banshee.SoundMenu (1.9.0.0) Banshee.Mpris (1.9.0.0) Migo (1.9.0.0) Banshee.Podcasting (1.9.0.0) Banshee.Dap (1.9.0.0) Banshee.LibraryWatcher (1.9.0.0) Banshee.MultimediaKeys (1.9.0.0) Banshee.Bpm (1.9.0.0) Banshee.YouTube (1.9.0.0) Banshee.WebBrowser (1.9.0.0) Banshee.Wikipedia (1.9.0.0) pango-sharp (2.12.0.0) Banshee.Fixup (1.9.0.0) Banshee.Widgets (1.9.0.0) gio-sharp (2.14.0.0) gudev-sharp (1.0.0.0) Banshee.Gio (1.9.0.0) Banshee.GStreamer (1.9.0.0) System.Configuration (2.0.0.0) NDesk.DBus.GLib (1.0.0.0) gconf-sharp (2.24.0.0) Banshee.Gnome (1.9.0.0) Banshee.NowPlaying (1.9.0.0) Mono.Cairo (2.0.0.0) System.Xml (2.0.0.0) Banshee.Core (1.9.0.0) Hyena.Data.Sqlite (1.9.0.0) System.Core (3.5.0.0) gdk-sharp (2.12.0.0) Mono.Addins (0.4.0.0) atk-sharp (2.12.0.0) Hyena.Gui (1.9.0.0) gtk-sharp (2.12.0.0) Banshee.ThickClient (1.9.0.0) Nereid (1.9.0.0) NDesk.DBus.Proxies (0.0.0.0) Mono.Posix (2.0.0.0) NDesk.DBus (1.0.0.0) glib-sharp (2.12.0.0) Hyena (1.9.0.0) System (2.0.0.0) Banshee.Services (1.9.0.0) Banshee (1.9.0.0) mscorlib (2.0.0.0) Platform Information: Linux 2.6.35-27-generic i686 unknown GNU/Linux Disribution Information: [/etc/lsb-release] DISTRIB_ID=Ubuntu DISTRIB_RELEASE=10.10 DISTRIB_CODENAME=maverick DISTRIB_DESCRIPTION="Ubuntu 10.10" [/etc/debian_version] squeeze/sid Just to make it clear, this happened only after the hard resets and not before. I used to use banshee everyday and it worked perfectly. Can anyone help me fix this?

  Read the article

• How to delay static initialization within a property

  - by Mystagogue

  I've made a class that is a cross between a singleton (fifth version) and a (dependency injectable) factory. Call this a "Mono-Factory?" It works, and looks like this: public static class Context { public static BaseLogger LogObject = null; public static BaseLogger Log { get { return LogFactory.instance; } } class LogFactory { static LogFactory() { } internal static readonly BaseLogger instance = LogObject ?? new BaseLogger(null, null, null); } } //USAGE EXAMPLE: //Optional initialization, done once when the application launches... Context.LogObject = new ConLogger(); //Example invocation used throughout the rest of code... Context.Log.Write("hello", LogSeverity.Information); The idea is for the mono-factory could be expanded to handle more than one item (e.g. more than a logger). But I would have liked to have made the mono-factory look like this: public static class Context { private static BaseLogger LogObject = null; public static BaseLogger Log { get { return LogFactory.instance; } set { LogObject = value; } } class LogFactory { static LogFactory() { } internal static readonly BaseLogger instance = LogObject ?? new BaseLogger(null, null, null); } } The above does not work, because the moment the Log property is touched (by a setter invocation) it causes the code path related to the getter to be executed...which means the internal LogFactory "instance" data is always set to the BaseLogger (setting the "LogObject" is always too late!). So is there a decoration or other trick I can use that would cause the "get" path of the Log property to be lazy while the set path is being invoked?

  Read the article

• 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'

  Read the article

• per terminal bell in ubuntu classic

  - by Owen Maresh

  I'm using natty. I'm using classic. I use raw xterms (the latest build, 270, in fact). I've done xset b 100 pactl upload-sample /usr/share/sounds/ubuntu/stereo/message.ogg bell.ogg But I want something more fine grained than this: I want to say "if the bell originated in some particular pseudoterminal make a particular sound, but if it originated in some other particular pseudoterminal, generate some other sound"

  Read the article

• How can I set a per terminal bell with xterms?

  - by Owen Maresh

  I'm using natty. I'm using classic. I use raw xterms (the latest build, 270, in fact). I've done xset b 100 pactl upload-sample /usr/share/sounds/ubuntu/stereo/message.ogg bell.ogg But I want something more fine grained than this: I want to say "if the bell originated in some particular pseudoterminal make a particular sound, but if it originated in some other particular pseudoterminal, generate some other sound"

  Read the article

• Splitting an MP4 file

  - by Asaf Chertkoff

  what is the fastest and less resource consuming method for splitting an MP4 file? @Alex: it didn't work, i don't know why. see the out put here: asafche@asafche-laptop:~$ ffmpeg -vcodec copy -ss 0 -t 00:10:00 -i /home/asafche/Videos/myVideos/MAH00124.MP4 /home/asafche/Videos/myVideos/eh.mp4 FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1.1 --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 31 2011 18:53:20, gcc: 4.4.3 Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -> 59.94 (60000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/asafche/Videos/myVideos/MAH00124.MP4': Duration: 00:15:35.96, start: 0.000000, bitrate: 5664 kb/s Stream #0.0(und): Video: h264, yuv420p, 1280x720, 59.94 tbr, 59.94 tbn, 119.88 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16 Output #0, mp4, to '/home/asafche/Videos/myVideos/eh.mp4': Stream #0.0(und): Video: libx264, yuv420p, 1280x720, q=2-31, 90k tbn, 59.94 tbc Stream #0.1(und): Audio: 0x0000, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec for output stream #0.1 it says something about different frame rate...

  Read the article

• how to convert avi (xvid) to mkv or mp4 (h264)

  - by bcsteeve

  Very noob when it comes to video. I'm trying to make sense of what I"m finding via Google... but its mostly Greek to me. I have a bunch of Avi files that won't play in my WD TV Play box. Mediainfo tells me they are xvid. Specs for the box show that should be fine... but digging through forums says its hit-and-miss. So I'd like to try converting them to h264 encoded MKV or mp4 files. I gather avconv is the tool, but reading the manual just has me really really confused. I tried the very basic example of: avconv -i file.avi -c copy file.mp4 it took less than 4 seconds. And it worked... sort of. It "played" in that something came up on the screen... but there was horrible artifacting and scenes would just sort of melt into each other. I want to preserve quality if possible. I'm not concerned about file size. I'm not terribly concerned with the time it takes either, provided I can do them in a batch. Can someone familiar with the process please give me a command with the options? Thank you for your help. I'm posting the mediainfo in case it helps: General Complete name : \\SERVER\Video\Public\test.avi Format : AVI Format/Info : Audio Video Interleave File size : 189 MiB Duration : 11mn 18s Overall bit rate : 2 335 Kbps Writing application : Lavf52.32.0 Video ID : 0 Format : MPEG-4 Visual Format profile : Advanced Simple@L5 Format settings, BVOP : 2 Format settings, QPel : No Format settings, GMC : No warppoints Format settings, Matrix : Default (H.263) Muxing mode : Packed bitstream Codec ID : XVID Codec ID/Hint : XviD Duration : 11mn 18s Bit rate : 2 129 Kbps Width : 720 pixels Height : 480 pixels Display aspect ratio : 16:9 Frame rate : 29.970 fps Standard : NTSC Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.206 Stream size : 172 MiB (91%) Writing library : XviD 1.2.1 (UTC 2008-12-04) Audio ID : 1 Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Mode extension : MS Stereo Codec ID : 55 Codec ID/Hint : MP3 Duration : 11mn 18s Bit rate mode : Constant Bit rate : 192 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Compression mode : Lossy Stream size : 15.5 MiB (8%) Alignment : Aligned on interleaves Interleave, duration : 24 ms (0.72 video frame)

  Read the article

• Tutoriel Perceptual Computing (introduction à Unity), par Cédric Andréolli

  Salut,Je viens vous présenter un tutoriel qui va probablement vous plaire. Citation: Le Perceptual Computing peut être défini comme un ensemble de techniques mises à disposition du développeur pour offrir une expérience nouvelle à l'utilisateur. Ce concept, développé par Intel®, est composé de deux éléments :une caméra possédant de nombreux capteurs (HD, profondeur, micro stéréo) ; un SDK développé par Intel permettant de récupérer des événements captés par la caméra. Le SDK permet de réaliser les tâches suivantes :

  Read the article

• Volume control doesn't work for headphones

  - by Hendekagon

  Fresh install of 11.10 and the gnome panel's volume control doesn't work for headphones but does work for speakers on my laptop. In sound settings I have Internal Audio Analogue Stereo selected for output (Output tab) and Analogue Speakers selected for Connector. When I plug headphones in, the volume control doesn't work and the volume is maximum through the headphones. On a side note, only one of my two headphone sockets produces sound too (whereas both worked in 10.04)

  Read the article

• 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!

  Read the article

• Look strange on gvim after applying Source Sans Pro font

  - by abcdabcd987

  I downloaded the Source Sans Pro font and install on my Fedora17(Xfce). I did mkfontscale, mkfontdir, fc-cache -fv, and after fc-list, could see it on the list. Then I changed guifont in gvim to Source\ Sans\ Pro\ 10, but it looks quite strange. And then I changed it to DejaVu\ Sans\ Mono\ 10, it looks nothing strange. So, why would this happend? And how to solve it? Thanks! Source Sans Pro DejaVu Sans Mono

  Read the article

• Why are unicode characters not rendering correctly

  - by sw1nn

  Background: I have some unicode characters in my prompt (git status markers essentially) I'm running urxvt under xfce on arch linux. I'm using DejaVu Sans Mono for Powerline font, specified via .Xresources line: URxvt*font: xft:DejaVu Sans Mono for Powerline:pixelsize=14 When I start urxvt the unicode characters do not render correctly. For example ? renders as â However, if I then start a new urxvt from inside the first terminal everything renders correctly. There doesn't appear to be any difference in the environment between the two terminals. What could be the difference between the first invocation and the nested invocation? I suspect the font is not correct in the 'outer' instance, but I'm unsure how to check the font of a running X window screenshot demonstrates the problem: Note: I moved this question from serverfault.com - i hope this site is more appropriate

  Read the article

• Help with dynamic range compression function (audio)

  - by MusiGenesis

  I am writing a C# function for doing dynamic range compression (an audio effect that basically squashes transient peaks and amplifies everything else to produce an overall louder sound). I have written a function that does this (I think): public static void Compress(ref short[] input, double thresholdDb, double ratio) { double maxDb = thresholdDb - (thresholdDb / ratio); double maxGain = Math.Pow(10, -maxDb / 20.0); for (int i = 0; i < input.Length; i += 2) { // convert sample values to ABS gain and store original signs int signL = input[i] < 0 ? -1 : 1; double valL = (double)input[i] / 32768.0; if (valL < 0.0) { valL = -valL; } int signR = input[i + 1] < 0 ? -1 : 1; double valR = (double)input[i + 1] / 32768.0; if (valR < 0.0) { valR = -valR; } // calculate mono value and compress double val = (valL + valR) * 0.5; double posDb = -Math.Log10(val) * 20.0; if (posDb < thresholdDb) { posDb = thresholdDb - ((thresholdDb - posDb) / ratio); } // measure L and R sample values relative to mono value double multL = valL / val; double multR = valR / val; // convert compressed db value to gain and amplify val = Math.Pow(10, -posDb / 20.0); val = val / maxGain; // re-calculate L and R gain values relative to compressed/amplified // mono value valL = val * multL; valR = val * multR; double lim = 1.5; // determined by experimentation, with the goal // being that the lines below should never (or rarely) be hit if (valL > lim) { valL = lim; } if (valR > lim) { valR = lim; } double maxval = 32000.0 / lim; // convert gain values back to sample values input[i] = (short)(valL * maxval); input[i] *= (short)signL; input[i + 1] = (short)(valR * maxval); input[i + 1] *= (short)signR; } } and I am calling it with threshold values between 10.0 db and 30.0 db and ratios between 1.5 and 4.0. This function definitely produces a louder overall sound, but with an unacceptable level of distortion, even at low threshold values and low ratios. Can anybody see anything wrong with this function? Am I handling the stereo aspect correctly (the function assumes stereo input)? As I (dimly) understand things, I don't want to compress the two channels separately, so my code is attempting to compress a "virtual" mono sample value and then apply the same degree of compression to the L and R sample value separately. Not sure I'm doing it right, however. I think part of the problem may the "hard knee" of my function, which kicks in the compression abruptly when the threshold is crossed. I think I may need to use a "soft knee" like this: Can anybody suggest a modification to my function to produce the soft knee curve?

  Read the article

• Ubuntu 11.10 - How can i stop self-feedback-loop, coming directly from my microphone to speaker?

  - by YumYumYum

  I have microphone audio, which comes instantly to my speaker. I am using default pulseaudio and alsa from the package. I have tried to setup: 1) PA /etc/pulse/default.pa /etc/asound.conf $ ls analog-input-aux.conf analog-input-fm.conf analog-input-mic.conf analog-input-tvtuner.conf analog-output-desktop-speaker.conf analog-output-mono.conf analog-input.conf analog-input-front-mic.conf analog-input-mic.conf.common analog-input-video.conf analog-output-headphones-2.conf analog-output-speaker.conf analog-input.conf.common analog-input-internal-mic.conf analog-input-mic-line.conf analog-output.conf analog-output-headphones.conf iec958-stereo-output.conf analog-input-dock-mic.conf analog-input-linein.conf analog-input-rear-mic.conf analog-output.conf.common analog-output-lfe-on-mono.conf 2) ALSA in lsmod to make sure no loopback modules are loaded etc but none is resolving it. And there are many less information available on this. Has anyone similar problem solution in Ubuntu 11.10? (this problem i have resolved in Ubuntu 11.04 by replacing the default pulseaudio version to latest source from git, but while trying the same in Ubuntu 11.10 does not worked). Any tips please?

  Read the article

• unable to install anything ,getting error subprocess installed post-installation script returned error exit status 1

  - by soum

  dpkg: error processing mono-4.0-gac (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for mousetweaks ... No apport report written because MaxReports is reached already postinst called with unknown argument `triggered' dpkg: error processing mousetweaks (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for mozilla-plugin-vlc ... postinst called with unknown argument `triggered' dpkg: error processing mozilla-plugin-vlc (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for mtools ... No apport report written because MaxReports is reached already postinst called with unknown argument `triggered' dpkg: error processing mtools (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager-pptp-gnome ... No apport report written because MaxReports is reached already postinst called with unknown argument `triggered' dpkg: error processing network-manager-pptp-gnome (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-pptp ... postinst called with unknown argument `triggered' dpkg: error processing network-manager-pptp (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for network-manager-gnome ... /var/lib/dpkg/info/network-manager-gnome.postinst called with unknown argument `triggered' dpkg: error processing network-manager-gnome (--configure): subprocess installed post-installation script returned error exit status 1 Processing triggers for network-manager ... No apport report written because MaxReports is reached already /var/lib/dpkg/info/network-manager.postinst called with unknown argument `triggered' dpkg: error processing network-manager (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Processing triggers for mscompress ... postinst called with unknown argument `triggered' dpkg: error processing mscompress (--configure): subprocess installed post-installation script returned error exit status 1 No apport report written because MaxReports is reached already Errors were encountered while processing: netbase mtr-tiny module-init-tools mountmanager mono-4.0-gac mousetweaks mozilla-plugin-vlc mtools network-manager-pptp-gnome network-manager-pptp network-manager-gnome network-manager mscompress E: Sub-process /usr/bin/dpkg returned an error code (1)

  Read the article

• Ubuntu...I love you, I hate you

  - by gregarobinson

   I have been working on seeing if a .NET 3.5 application will port over to Linux, Ubuntu to be specific. I started with version 9.01, then 9.10 and now 10.04 as I find more and more that I need from Mono. I have a dual boot on a dev box, Windows 7 and Ubuntu. An upgrade from Ubuntu 9.01 to 9.10 caused my mouse and keyboard to lock up. I was able to boot from a 9.10 cd. Then, I upgraded to 10.04 as I needed Mono 2,2. Upgrade worked, lost my windows boot though. it seems grub somehow jumped in and messed up the windows boot. After Googlign liek crazy and trying this and that, these 2 links finally got me my windows boot back:http://sourceforge.net/apps/mediawiki/bootinfoscript/index.php?title=Boot_Problems:Boot_Sector http://support.microsoft.com/kb/927392So, I am now thinking about trying SuSe instead as I hear\read it's more stable. I think a lot of my pains have been related to learning and getting use to Linux.        

  Read the article

• Alternatives for the Snippet Compiler

  - by Marko Apfel

  It seems that the Snippet Compiler is not maintained anymore. So I need an alternative – also for getting syntax highlighting for code in publications. Preferable with the possibility to conserve this highlighting by copying selections to clipboard. Snippet Compiler does not allows this for selections – only by exporting the whole file content to clipboard with HTML- or RTF-formatting (File > Export > HTML to clipboard respectively RTF to clipboard). Today I switched to LINQPad. This application offers constructing LINQ-statements as well as compiling arbitrary code snippets. But there are some other alternatives too: CS-Script - The C# Script Engine CS-Script is a CLR (Common Language Runtime) based scripting system which uses ECMA-compliant C# as a programming language. CS-Script currently targets Microsoft implementation of CLR (.NET 2.0/3.0/3.5. sharpsnippetcompiler C# Snippet Compiler is a tiny IDE to create, debug and run your C# programs CsharpRepl C# interactive shell that is part of Mono's C# compiler. An interactive shell is usually referred to as a read eval print loop or repl. The C# interactive shell is built on top of the Mono.CSharp library, a library that provides a C# compiler service that can be used to evaluate expressions and statements on the flight. What I miss is an alternative with syntax highlighting like in my Visual Studio: Instead of:

  Read the article

• Why is Rhythmbox becoming the default (again)?

  - by Christoph

  So, it seems with 12.04, they're switching back to Rhythmbox, after switching from Rhythmbox a year ago. I don't get why. They say that it's because of a blocking bug in GTK3# (if I understand that correctly), but that's just one bug, and in the same breath they say RB is not well maintained. It seems Ubuntu guys were dissatisfied with Banshee in some way, but apparently the Banshee guys were never notified of any problems. Also, it can't be to save disc space by dropping mono, because at the same day it was announced that the install disc will be enlarged by 50MB. Also, isn't it a bit shortsighted to push Banshee for default inclusion, and then drop it again a year later? How is that a sustainable use of dev resources, or consistent? Apparently there was quite some heavy effort by banshee devs - David Nielsen used the term "bending over backwards for Ubuntu" iirc. In summary: Can anyone shed more light on this? Related question: Why is Banshee becoming the default? Sources: http://www.omgubuntu.co.uk/2011/11/banshee-tomboy-and-mono-dropped-from-ubuntu-12-04-cd/ http://www.omgubuntu.co.uk/2011/11/rhythmbox-to-return-as-ubuntu-12-04-default-music-app/ http://www.omgubuntu.co.uk/2011/11/ubuntu-12-04-disc-size-to-be-750mb/ http://summit.ubuntu.com/uds-p/meeting/19442/desktop-p-default-apps/ http://banshee-media-player.2283330.n4.nabble.com/banshee-being-dropped-from-ubuntu-because-of-GTK3-support-td3985298.html

  Read the article

• Launcher icons are invisible after upgrade from 11.10 to 12.04

  - by Clo Knibbe

  I am re-purposing an old laptop. I installed 11.10 on it and then immediately upgraded to 12.04. (I could not directly install 12.04 as my system does not support PAE.) When my system was (briefly) 11.10, the desktop appeared as expected. However, after the upgrade to 12.04, the icons in the launcher area are invisible. If I hover over the spot where the icon should be the little popup window showing the tool's name appears, and I can click to invoke the tool. I just cannot see the icons. ![invisible icons in launcher][1] The icons do appear as expected in other contexts, for example in the Home folder and in Dash Home. My theme is "Ambiance (default)" I do not have a ~/.icons folder. This is the top level contents of /usr/share/icons: default DMZ-Black DMZ-White gnome handhelds hicolor HighContrast HighContrastInverse Humanity Humanity-Dark locolor LoginIcons LowContrast redglass ubuntu-mono-dark ubuntu-mono-light unity-icon-theme whiteglass (Sorry for the poor formatting, can't get it to show in list.) I suspect that the launcher isn't looking for the icons in the right place, but I don't know how to confirm that, or how to correct. This is my first foray into Linux, although I used to use Unix a few decades ago. This doesn't look much like my old Sun workstation, though! Does anyone have any suggestions or insights for me? Thanks.

  Read the article

• Banshee doesn't like opening websites

  - by Allan

  I have come across two bugs (which will be added to launchpad if it's not resolved here) When I open any of the websites in Banshee Amazon or Miro Guide as soon as the site is finished loading it crashes Banshee. If I play any video local or remote it will show 1 frame maybe 0.5 sec of video then I get a black screen and audio continues in the backgound. Specs & Details I have a Fujitsu Amilo 1718 laptop with 2 gig of ram (original 1 gig) graphics is provided by ATI Radeon Xpress 200M (don't laugh it works with compiz....just) I have a link to the output of banshee --debug Here Don't have time to read? Here are the Highlights [2 Warn 11:52:34.814] Caught an exception - System.ArgumentNullException: Argument cannot be null. then abit later Debug info from gdb: Could not attach to process. If your uid matches the uid of the target process, check the setting of /proc/sys/kernel/yama/ptrace_scope, or try again as the root user. For more details, see /etc/sysctl.d/10-ptrace.conf ptrace: Operation not permitted. ================================================================= Got a SIGSEGV while executing native code. This usually indicates a fatal error in the mono runtime or one of the native libraries used by your application. ================================================================= Aborted Not music to my ears as you can expect. The version I am using is 1.9.4 from the daily ppa but these bugs happen in any version of banshee from 1.8.1 and up. So if any one has come across a fix for this problem please share!! additional info Both VLC and Miro work on my system so there isn't a system wide problem with video and I haven't mentioned mono so no trolling it will get voted down.

  Read the article

• Play audio over network with Windows 7?

  - by Josh

  I have a unique situation where I'd like to stream audio (ALL audio, not just mp3s, etc) from my laptop to another computer over the network. I live in a studio apartment and my laptop is my main computer but I'd like it's audio to play on my htpc with a nice stereo system. Since it's a studio, both computers are in the same room so I don't want 2 sets of speakers. I want my computer to directly play back through the stereo. I used to do this with pulseaudio but my job now requires that I run Windows full time. I'm aware of Shoutcast and other similar streaming solutions but I don't want any transcoding done. It's a waste of CPU and not to mention my laptop fans, and I don't mind the network bandwidth that uncompressed audio requires. Is there a way to run Shoutcast without encoding? Also, I know that Windows Remote Desktop can play audio over the network pretty easily. Is this part of .Net that I could just code a simple app that streams the audio without RD'ing in? I also don't want to run it over a physical wire. :)

  Read the article

• Splitting HDMI sound to 2 devices under Windows 7

  - by Jeramy

  Okay, this is a strange set-up and is frustrating me. I have an HDMI signal from my PC being split to my audio receiver and my HDTV. I need to split it to both so that I can choose to either play audio from the HDTV or from the surround sound speakers in the room. The problem that I am having is in Windows 7, the output is listed under "Playback Devices" and is auto-populated with the HDTV, which only has the option for stereo sound. If I unplug the HDTV from the splitter it will populate with my receiver information and let me set it to 5.1 surround, but as soon as I plug the HDTV back in it reverts. I tried reversing the order of the HDMI cables in the splitter and this seemed to work for a short while, then Windows must have polled the devices again or something because it reverted. It will work as long as Windows identifies the reciever, thereby unlocking the 5.1 surround option, otherwise I am stuck with stereo, which it assumes is all the HDTV is capable of. Is there a way to manually override this and set my own options? Or any other solutions?

  Read the article

• FFMPEG: how to add watermark to video?

  - by DocWiki

  My Platform: Ubuntu 10.10 + FFMPEG 0.5.3(I installed ffmpeg from source) I try to add Watermark to a .MOV video with FFMPEG 0.5.3 imlib2.so (Please note FFMPEG 0.6+ dont support imlib2.so, so I use ffmpeg 0.5.3) Here is my code: ffmpeg -sameq -i example.mov -vhook '/usr/local/lib/vhook/imlib2.so -x 0 -y 0 -i /var/www/files/watermark.png' newexample.mov Here is the output: FFmpeg version 0.5.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-avfilter --enable-filter=movie --enable-avfilter-lavf 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 built on Jul 3 2011 12:05:08, gcc: 4.4.5 Seems stream 1 codec frame rate differs from container frame rate: 59.94 (5994/100) - 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'example.mov': Duration: 00:03:14.06, start: 0.000000, bitrate: 3350 kb/s Stream #0.0(eng): Audio: aac, 48000 Hz, stereo, s16 Stream #0.1(eng): Video: h264, yuv420p, 1150x647, 29.97 tbr, 29.97 tbn, 59.94 tbc Output #0, mov, to 'newexample.mov': Stream #0.0(eng): Video: mpeg4, yuv420p, 1150x647, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Stream #0.1(eng): Audio: 0x0000, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 - #0.0 Stream #0.0 - #0.1 Unsupported codec for output stream #0.1 What could be the possible problem? Is that AAC or H264 that is not supported? I installed libavcodec-extra-52, linfaac, libfaad and etc. but the error is the same. Do I have to install following this instruction? HOWTO: Install and use the latest FFmpeg and x264 or there is a simpler solution?

  Read the article

< Previous Page | 20 21 22 23 24 25 26 27 28 29 30 31  | Next Page >