Search Results

Search found 174 results on 7 pages for 'bitrate'.

Page 6/7 | < Previous Page | 2 3 4 5 6 7  | Next Page >

• Should I partition a 1TB Hard Disk whose primary use is media storage?

  - by Senthil

  I am going to get a 1TB hard disk. I will be storing 1080p or 720p movies, high-bitrate music and pictures in it. I use my PC 90% of the time only to play/listen/see those. I am running out of space in my current HD so I am getting another one. My specs are 2.7GHz Dual Core, 512MB GeForce 9400GT, 2GB DDR2 RAM and all the proper matroska codecs/players. I guess that is enough to play 1080p movies withough a glitch, given an ideal hard disk. I've read about proper partitioning giving performance improvement etc.. I don't want my hard disk to be the bottleneck. Can someone tell me whether I should partition my 1TB hard disk into many drives? If I should, what is the ideal size of each partition? Smooth playing of movies is very important to me. Once I start filling up the disk, there is no turning back. So I want to get it right before I start. Thanks.

  Read the article

• convert decrypted .vobs to .avi with ffmpeg on ubuntu

  - by Arcath

  I have a .vob file that has bee ripped from a dvd, when I watch the .vob its very good quality video and 5.1 english audio but when I use ffmpeg it has rubbish video and mono french audio. That was using this command: ffmpeg -i /samba/ripping/vobs/12161840#2.vob -f avi /samba/ripping/avis/test.avi I've tried a few different variations on that but it never comes back with anything good just bigger files with bad video and incorrect sound. I know the videos good and the correct audio streams exist so how do I select a 5.1 track and get good video? ffmpeg gives the .vob details as: Input #0, mpeg, from '/samba/ripping/vobs/12161840#2.vob': Duration: 00:42:05.56, start: 0.287267, bitrate: 5738 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 8436 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, mono, s16, 192 kb/s Output #0, avi, to '/samba/ripping/avis/test.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.3 -> #0.1

  Read the article

• How to convert an MKV to AVI with minimal loss

  - by Linux Jedi

  To convert an MKV to AVI, I do two things. The first thing I do is this: ffmpeg -i filename.mkv -vcodec copy -acodec copy output.avi This converts the MKV to an AVI, but the problem is that the video does not play smoothly for some reason. That's fine, because if I do one more thing it gets fixed: ffmpeg -i output.avi -vcodec mpeg4 -b 4000k -acodec mp2 -ab 320k converted.avi After I do this then the file plays without problem. I had success doing it this way for one file, but then I tried it on another file, and there is a slight, but noticeable loss in video quality. This is the output I get when doing the second step: FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Dec 29 2010 18:02:10 with gcc 4.2.1 (Apple Inc. build 5664) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Seems stream 0 codec frame rate differs from container frame rate: 359.00 (359/1) -> 29.92 (359/12) Input #0, avi, from 'output.avi': Metadata: ISFT : Lavf52.64.2 Duration: 00:04:17.21, start: 0.000000, bitrate: 3074 kb/s Stream #0.0: Video: mpeg4, yuv420p, 704x480 [PAR 229:189 DAR 5038:2835], 29.92 fps, 29.92 tbr, 29.92 tbn, 359 tbc Stream #0.1: Audio: vorbis, 48000 Hz, stereo, s16 Output #0, avi, to 'nidome_no_kanojo.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 704x480 [PAR 229:189 DAR 5038:2835], q=2-31, 4000 kb/s, 29.92 tbn, 29.92 tbc Stream #0.1: Audio: mp2, 48000 Hz, stereo, s16, 320 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 I just used arbitrarily large settings on the second step and it worked nicely before but not in this case. What settings should I use?

  Read the article

• ffmpeg conversion problem

  - by user33126

  installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

  Read the article

• ffmpeg conversion problem

  - by Elamurugan

  installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

  Read the article

• Best video codec to store my own collection

  - by Jack

  Hello! I think this question has already been asked but with different flavours. My problem resised in the fact that my camera (Canon G9) creates video with almost raw codec (I think it's plain old MPEG) so a 10 minutes video is almost 900mb. I would like to convert them in a format that has a good trade-off between space and quality, but I would prefer having the quality as good as the original (of course this is not possible because of lossy compression) just saving as much space is possible with a minimal lose of quality. Which codec should I look for? H264? It seems to be the champion of the moment.. otherwise which other ones could I try? XviD? Which parameters should I use? I mean how many kbits/s is a fair good bitrate to keep high quality? And what about audio codec? video specs are 640x480 at 30fps or 1024x768 at 15fps.. thanks in advance!

  Read the article

• ffmpeg - h264 to xvid creates large file

  - by fatnic

  I'm trying to use ffmpeg to convert a h264/aac video file to an xvid/mp3 file so I can play it in my ultra-cheap media player. At the moment the converted video file is TWICE the size of the original mp4. Is there any way to get a smaller file size without loosing too much quality? Even a drop to -qmin 1 is pretty awful! The command i'm using is ffmpeg -i input.mp4 -vcodec libxvid -sameq -acodec libmp3lame -ab 128k -ac 2 output.avi And the ffmpeg output is Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4' Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 Duration: 01:34:27.69, start: 0.000000, bitrate: 1520 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x304 [PAR 1:1 DAR 45:19], 1387 kb/s, 25 fps, 25 tbr, 25k tbn, 50 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Output #0, avi, to 'output.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0(und): Video: mpeg4, yuv420p, 720x304 [PAR 1:1 DAR 45:19], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1(und): Audio: libmp3lame, 48000 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1

  Read the article

• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

  Read the article

• Would like to change audio codec, but keep video settings with ffmpeg

  - by Craig Tataryn

  I have a video for which I'd like to convert the audio codec to AAC 320 kbps / 44.100 kHz. What would I use for ffmpeg switches such that all the video settings and codec remain the same, but only the audio codec and settings change? Here's my video: $ ffmpeg -i Winnipeg.rb\ Scala-Talk.mov FFmpeg version SVN-r25375, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 6 2010 13:02:41 with gcc 4.2.1 (Apple Inc. build 5664) configuration: --enable-libmp3lame --enable-shared --disable-mmx --arch=x86_64 libavutil 50.32. 2 / 50.32. 2 libavcore 0. 9. 1 / 0. 9. 1 libavcodec 52.92. 0 / 52.92. 0 libavformat 52.80. 0 / 52.80. 0 libavdevice 52. 2. 2 / 52. 2. 2 libavfilter 1.48. 0 / 1.48. 0 libswscale 0.12. 0 / 0.12. 0 Seems stream 0 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 10.00 (10/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Winnipeg.rb Scala-Talk.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt Duration: 01:10:53.00, start: 0.000000, bitrate: 283 kb/s Stream #0.0(eng): Video: h264, yuv420p, 800x598, 94 kb/s, 10 fps, 10 tbr, 1k tbn, 2k tbc Stream #0.1(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 Stream #0.2(eng): Audio: adpcm_ima_qt, 22050 Hz, 1 channels, s16 At least one output file must be specified Many thanks in advance! One with with ffmpeg I've never been able to grok is how to just "tweak" files without having to regurgitate every little setting for things you don't want changes.

  Read the article

• converting to MXF using ffmpeg

  - by Prakash

  I have been trying to use FFmpeg utility to convert a avi file using DNxHD to mxf format. I am using "FFmpeg" with params as following: ffmpeg -i ccvt_box.avi -vcodec dnxhd -video_size 1920x1080 -r 24 -b:v 115m ex.mxf The error it is giving : ffmpeg version N-43737-g76c3fff Copyright (c) 2000-2012 the FFmpeg developers built on Aug 20 2012 18:50:42 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: libavutil 51. 70.100 / 51. 70.100 libavcodec 54. 53.100 / 54. 53.100 libavformat 54. 25.104 / 54. 25.104 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 11.101 / 3. 11.101 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 Input #0, avi, from 'ccvt_box.avi': Duration: 00:00:10.00, start: 0.000000, bitrate: 691 kb/s Stream #0:0: Video: indeo5 (IV50 / 0x30355649), yuv410p, 340x344, 10 tbr, 10 tbn, 10 tbc Metadata: title : bob.avi [dnxhd @ 0x7fcd60818e00] video parameters incompatible with DNxHD Output #0, mxf, to 'ex.mxf': Stream #0:0: Video: dnxhd, yuv422p, 340x344, q=2-1024, 90k tbn, 24 tbc Metadata: title : bob.avi Stream mapping: Stream #0:0 -> #0:0 (indeo5 -> dnxhd) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

  Read the article

• Android -- Can't play any videos (mp4/mov/3gp/etc.)?

  - by borg17of20

  Hello all, I'm having great difficulty getting my Android application to play videos from the SD card. It doesn't matter what size, bitrate, video format, or any other setting I can think of, neither the emulator nor my G1 will play anything I try to encode. I've also tried a number of videos from the web (various video formats, bitrates, with and without audio tracks, etc.), and none of those work either. All I keep getting is a dialog box that says: "Cannot play video" "Sorry, this video cannot be played." There are errors reported in LogCat, but I don't understand them and I've tried searching the Internet for further explanations without any luck. See below: 03-30 05:34:26.807: ERROR/QCOmxcore(51): OMXCORE API : Free Handle 390d4 03-30 05:34:26.817: ERROR/QCOmxcore(51): Unloading the dynamic library for OMX.qcom.video.decoder.avc 03-30 05:34:26.817: ERROR/PlayerDriver(51): Command PLAYER_PREPARE completed with an error or info PVMFErrNoResources 03-30 05:34:26.857: ERROR/MediaPlayer(14744): error (1, -15)03-30 05:34:26.867: ERROR/MediaPlayer(14744): Error (1,-15) Sometimes I also get this: 03-30 05:49:49.267: ERROR/PlayerDriver(51): Command PLAYER_INIT completed with an error or info PVMFErrResource 03-30 05:49:49.267: ERROR/MediaPlayer(19049): error (1, -17) 03-30 05:49:49.347: ERROR/MediaPlayer(19049): Error (1,-17) Here is the code I'm using (in my onCreate() method): this.setContentView(R.layout.main); //just a simple VideoView loading files from the SD card VideoView myIntroView = (VideoView) this.findViewById(R.id.VideoView01); MediaController mc = new MediaController(this); myIntroView.setMediaController(mc); myIntroView.setVideoPath("/sdcard/test.mp4"); myIntroView.requestFocus(); myIntroView.start(); Please help!

  Read the article

• what is the best way to stream a audio file to website users/listners

  - by Naveen Chamikara Gamage

  I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage.. As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one? I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality.. I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers? if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use? I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files? Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files? Any known best softwares for converting audio files while keeping quality in a good level? Note** - I know that I will not need complex requirements at the beginning of the site but I wanted to what are the best ways like they are using for soundcloud.com

  Read the article

• Encoding h.264 with libavcodec/x264

  - by Leviathan

  I am attempting to encode video using libavcodec/libavformat. I'm trying to change the standard output-example.c from ffmpeg source. The AVI file is created on the disk, but the only sound is encoded. I tried adding a lot of options for x264 from here. All the other codecs works fine, mpeg2, mpeg4, mjpeg, xvid. In addition to specifying the parameters x264, I also set the codec to AVOutputFormat structure. That's all I've done. AVOutputFormat *pOutFormat; // in header file av_register_all(); AVCodec *codec = avcodec_find_encoder_by_name("libx264"); pOutFormat = guess_format("avi", NULL, NULL); pOutFormat->video_codec = codec->id; The debug output of my application: Output #0, mp4, to 'D:\1.avi': Stream #0.0: Video: libx264, yuv420p, 320x240, q=10-51, 500 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: aac, 44100 Hz, 1 channels, s16, 128 kb/s [libx264 @ 0x694010]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x694010]bitrate tolerance too small, using .01 [libx264 @ 0x694010]profile Main, level 2.0 [libx264 @ 0x694010]frame I:150 Avg QP:14.76 size: 2534 [libx264 @ 0x694010]mb I I16..4: 75.9% 0.0% 24.1% [libx264 @ 0x694010]final ratefactor: 17.57 [libx264 @ 0x694010]coded y,uvDC,uvAC intra: 42.7% 92.4% 47.4% [libx264 @ 0x694010]i16 v,h,dc,p: 11% 14% 2% 73% [libx264 @ 0x694010]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 18% 29% 5% 8% 10% 3% 3% 2% [libx264 @ 0x694010]kb/s:506.79

  Read the article

• FFMpeg Error av_interleaved_write_frame():

  - by rajaneesh

  this my code . after running php code FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:05:03, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) - 25.00 (25/1) Input #0, flv, from 'demo.flv': Duration: 00:00:30.83, start: 0.000000, bitrate: 546 kb/s Stream #0.0: Video: h264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], 546 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: aac, 44100 Hz, stereo, s16 Output #0, image2, to 'demo.jpg': Stream #0.0: Video: mjpeg, yuvj420p, 640x360 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding av_interleaved_write_frame(): I/O error occurred Usually that means that input file is truncated and/or corrupted.

  Read the article

• c# regex split and extract multiple parts from a string

  - by nLL

  Hi, I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) - 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to diffrent string objects instead of single

  Read the article

• Segmentation fault while feeding in an mpeg file through ffmpeg

  - by angel6

  Hi, I've set up FFserver as the streaming server. I'm trying to feed in an mpeg file. But it comes up with a segmentation fault. Does anyone know how to fix this? The following is the command-line output I get $ ./ffmpeg -i test1.mpg http://localhost:8090/feed1.ffm FFmpeg version SVN-r22945, Copyright (c) 2000-2010 the FFmpeg developers built on Apr 22 2010 19:18:45 with gcc 4.4.1 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-pthreads --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libx264 --enable-libxvid --enable-x11grab libavutil 50.14. 0 / 50.14. 0 libavcodec 52.66. 0 / 52.66. 0 libavformat 52.61. 0 / 52.61. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0xab0c420]max_analyze_duration reached Input #0, mpeg, from 'test1.mpg': Duration: 00:00:20.96, start: 0.768300, bitrate: 269 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 160x120 [PAR 1:1 DAR 4:3], 104857 kb/s, 30 fps, 30 tbr, 90k tbn, 30 tbc Stream #0.1[0x1c0]: Audio: mp2, 32000 Hz, 2 channels, s16, 64 kb/s Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : Lavf52.61.0 Stream #0.0: Audio: mp2, 22050 Hz, 1 channels, s16, 48 kb/s Stream #0.1: Video: mpeg1video, yuv420p, 160x128, q=2-31, 40 kb/s, 1000k tbn, 50 tbc Stream #0.2: Audio: libmp3lame, 22050 Hz, 1 channels, s16, 64 kb/s Stream #0.3: Video: msmpeg4, yuv420p, 352x240, q=2-31, 256 kb/s, 1000k tbn, 15 tbc Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Stream #0.1 -> #0.2 Stream #0.0 -> #0.3 Press [q] to stop encoding Segmentation fault

  Read the article

• DataGridView not displaying a row after it is created

  - by joslinm

  Hi, I'm using Visual Studio 10 and I just created a Database using SQL Server CE. Within it, I made a table CSLDataTable and that automatically created a CSLDataSet & CSLDataTableTableAdapter. The three variables were automatically created in my MainWindow.cs class: cSLDataSet cSLDataTableTableAdapter cSLDataTableBindingSource I have added a DataGridView in my Form called dataGridView and datasource cSLDataTableBindingSource. In my MainWindow(), I tried adding a row as a test: public MainWindow() { InitializeComponent(); CSLDataSet.CSLDataTableRow row = cSLDataSet.CSLDataTable.NewCSLDataTableRow(); row.File_ = "file"; row.Artist = "artist11"; row.Album = "album"; row.Save_Structure = "save"; row.Sent = false; row.Error = true; row.Release_Format = "release"; row.Bit_Rate = "bitrate.."; row.Year = "year"; row.Physical_Format = "format"; row.Bit_Format = "bitformat"; row.File_Path = "File!!path"; row.Site_Origin = "what"; cSLDataSet.CSLDataTable.Rows.Add(row); cSLDataSet.AcceptChanges(); cSLDataTableTableAdapter.Fill(cSLDataSet.CSLDataTable); cSLDataTableTableAdapter.Update(cSLDataSet); dataGridView.Refresh(); dataGridView.Update(); } In regards to the DataSet methods I tried calling, I had been trying to find a "correct" way to interact with the adapter, dataset, and datatable to successfully show the row, but to no avail. I'm rather new to using SQL Server CE Database, and read a lot of the MSDN sites & thought I was on the right track, but I've had no luck. The DataGridView shows the headers correctly, but that new row does not show up.

  Read the article

• Data Modeling of Entity with Attributes

  - by StackOverflowNewbie

  I'm storing some very basic information "data sources" coming into my application. These data sources can be in the form of a document (e.g. PDF, etc.), audio (e.g. MP3, etc.) or video (e.g. AVI, etc.). Say, for example, I am only interested in the filename of the data source. Thus, I have the following table: DataSource Id (PK) Filename For each data source, I also need to store some of its attributes. Example for a PDF would be "numbe of pages." Example for audio would be "bit rate." Example for video would be "duration." Each DataSource will have different requirements for the attributes that need to be stored. So, I have modeled "data source attribute" this way: DataSourceAttribute Id (PK) DataSourceId (FK) Name Value Thus, I would have records like these: DataSource->Id = 1 DataSource->Filename = 'mydoc.pdf' DataSource->Id = 2 DataSource->Filename = 'mysong.mp3' DataSource->Id = 3 DataSource->Filename = 'myvideo.avi' DataSourceAttribute->Id = 1 DataSourceAttribute->DataSourceId = 1 DataSourceAttribute->Name = 'TotalPages' DataSourceAttribute->Value = '10' DataSourceAttribute->Id = 2 DataSourceAttribute->DataSourceId = 2 DataSourceAttribute->Name = 'BitRate' DataSourceAttribute->Value '16' DataSourceAttribute->Id = 3 DataSourceAttribute->DataSourceId = 3 DataSourceAttribute->Name = 'Duration' DataSourceAttribute->Value = '1:32' My problem is that this doesn't seem to scale. For example, say I need to query for all the PDF documents along with thier total number of pages: Filename, TotalPages 'mydoc.pdf', '10' 'myotherdoc.pdf', '23' ... The JOINs needed to produce the above result is just too costly. How should I address this problem?

  Read the article

• regex split and extract multiple parts from a string

  - by nLL

  I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) -> 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to different string objects instead of single

  Read the article

• Receive JSON payload with ZEND framework and / or PHP

  - by kent3800

  I'm receiving a JSON payload from a webservice at my site's internal webpage at /asset/setjob. The following is the JSON payload being posted to /asset/setjob: [{"job": {"source_filename": "beer-drinking-pig.mpg", "current_step": "waiting_for_file", "encoding_profile_id": "nil", "resolution": "nil", "status_url": "http://example.com/api/v1/jobs/1.json", "id": 1, "bitrate": "nil", "current_status": "waiting for file", "current_progress": "nil", "remote_id": "my-own-remote-id"}}] This payload posts one time to this page. The page is not meant for viewing but parsing the JSON object for the id and current_status so that I can insert it into a database. I'm using Zend framework. HOW DO I receive this payload in Zend? Do I $_GET['json']? $_POST['job']? None of these seem to work. I essentially need to assign this payload to a php variable so that I can then manipulate it. I've tried: $jsonStrGet = var_dump($_GET); $jsonStrPost = var_dump($_POST); And I've tried: $response = $this-getResponse(); $body = $response-getBody(); Blockquote Any help would be much appreciated! Thanks.

  Read the article

• How to convert an MKV to AVI with minimal loss

  - by OSX NINJA

  To convert an MKV to AVI, I do two things. The first thing I do is this: ffmpeg -i filename.mkv -vcodec copy -acodec copy output.avi or this: ffmpeg -i filename.mkv -sameq -acodec copy output.avi Either of these will convert the MKV to an AVI, but the problem is that the video does not play smoothly for some reason. That's fine though, because if I do one more thing it gets fixed: ffmpeg -i output.avi -vcodec mpeg4 -b 4000k -acodec mp2 -ab 320k converted.avi After I do this then the file plays without problem. I had success doing it this way for one file, but then I tried it on another file, and there is a slight, but noticeable loss in video quality. This is the output I get when doing the second step: FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Dec 29 2010 18:02:10 with gcc 4.2.1 (Apple Inc. build 5664) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Seems stream 0 codec frame rate differs from container frame rate: 359.00 (359/1) -> 29.92 (359/12) Input #0, avi, from 'output.avi': Metadata: ISFT : Lavf52.64.2 Duration: 00:04:17.21, start: 0.000000, bitrate: 3074 kb/s Stream #0.0: Video: mpeg4, yuv420p, 704x480 [PAR 229:189 DAR 5038:2835], 29.92 fps, 29.92 tbr, 29.92 tbn, 359 tbc Stream #0.1: Audio: vorbis, 48000 Hz, stereo, s16 Output #0, avi, to 'converted.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 704x480 [PAR 229:189 DAR 5038:2835], q=2-31, 4000 kb/s, 29.92 tbn, 29.92 tbc Stream #0.1: Audio: mp2, 48000 Hz, stereo, s16, 320 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 I just used arbitrarily large settings on the second step and it worked nicely before but not in this case. What settings should I use?

  Read the article

• ffmpeg - join / merge on top of each other

  - by AisIceEyes

  I'm trying to join together two videos on top of each other. I already did these two ffmpeg commands ffmpeg -i 2_Out_of_Control.VOB -aspect 16:9 \ -vf "yadif=0:-1:0,crop=w=714:h=476:x=6:y=0,scale=1280:720,boxblur=lp=13" \ -c:v libx264 -preset medium \ -c:a copy \ '2(blurred)Out_of_Control.mp4' ffmpeg -i 2_Out_of_Control.VOB \ -vf "yadif=0:-1:0,crop=w=714:h=476:x=6:y=0,scale=1080:720" \ -c:v libx264 -preset medium \ -c:a copy \ '2(clear)Out_of_Control.mp4' I'm currently stuck on making the "clear" version on top of the "blurred" version. I'm not sure how to do that. Can anybody help please? Been googling for around 2 days already. Only achieved it by using OpenShot but yeah, would prefer if there is an ffmpeg command to merge the two videos on top of each other. Edit: I want the "clear" video to be at the center at the top of the "blurred" video Edit2: console output would be the same as above: ffmpeg -i 2(blurred)Out_of_Control.mp4 \ -i 2(clear)Out_of_Control.mp4 \ -aspect 16:9 \ -vf <just something that will join the two together: the blurred at the bottom, clear at top that is centered> \ -c:v libx264 -preset medium \ -c:a copy \ '2_Out_of_Control_VOB.mp4' Edit3: here is the output when I used ffmpeg -i 2_Out_of_Control.VOB $ ffmpeg -i 2_Out_of_Control.VOB ffmpeg version git-2013-10-03-c7fe2a3 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 4 2013 05:22:06 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --prefix=/home/username/ffmpeg_build --extra-cflags=-I/home/username/ffmpeg_build/include --extra-ldflags=-L/home/username/ffmpeg_build/lib --bindir=/home/username/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpeg, from '2_Out_of_Control.VOB': Duration: 00:05:00.01, start: 0.500000, bitrate: 4574 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, smpte170m), 720x480 [SAR 8:9 DAR 4:3], max. 9334 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 384 kb/s At least one output file must be specified $

  Read the article

• Can't control connection bit rate using iwconfig with Atheros TL-WN821N (AR7010)

  - by Paul H

  I'm trying to reduce the connection bit rate on my Atheros TP-Link TL-WN821N v3 usb wifi adapter due to frequent instability issues (reported connection speed goes down to 1Mb/s and I have to physically reconnect the adapter to regain a connection). I know this is a common problem with this device, and I have tried everything I can think of to fix it, including using drivers from linux-backports; compiling and installing a custom firmware (following instructions on https://wiki.debian.org/ath9k_htc#fw-free) and (as a last resort) using ndiswrapper. When using ndiswrapper, the wifi adapter is stable and operates in g mode at 54Mb/s (whilst when using the default ath9k_htc module, the adapter connects in n mode and the bit rate fluctuates constantly). Unfortunately, with this setup I have to run my processor using only one core, since using SMP with ndiswrapper causes a kernel oops on my system. So I want to lock my bit rate to 54Mb/s (or less, if need be) for connection stability, using the ath9k_htc module. I've tried 'sudo iwconfig wlan0 rate 54M'; the command runs with no error but when I check the bit rate with 'sudo iwlist wlan0 bitrate' the command returns: wlan0 unknown bit-rate information. Current Bit Rate:78 Mb/s Any ideas? Here's some info (hopefully relevant) on my setup: Xubuntu (12.04.3) 64bit (kernel 3.2.0-55.85-generic) using Network Manager. My Router is from Virgin Media, the VMDG480. lshw -C network : *-network description: Wireless interface physical id: 1 bus info: usb@1:4 logical name: wlan0 serial: 74:ea:3a:8f:16:b6 capabilities: ethernet physical wireless configuration: broadcast=yes driver=ath9k_htc driverversion=3.2.0-55 firmware=1.3 ip=192.168.0.9 link=yes multicast=yes wireless=IEEE 802.11bgn lsusb -v: Bus 001 Device 003: ID 0cf3:7015 Atheros Communications, Inc. TP-Link TL-WN821N v3 802.11n [Atheros AR7010+AR9287] Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 255 Vendor Specific Subclass bDeviceProtocol 255 Vendor Specific Protocol bMaxPacketSize0 64 idVendor 0x0cf3 Atheros Communications, Inc. idProduct 0x7015 TP-Link TL-WN821N v3 802.11n [Atheros AR7010+AR9287] bcdDevice 2.02 iManufacturer 16 ATHEROS iProduct 32 UB95 iSerial 48 12345 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 60 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 6 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 255 Vendor Specific Subclass bDeviceProtocol 255 Vendor Specific Protocol bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0000 (Bus Powered) iwlist wlan0 scanning: wlan0 Scan completed : Cell 01 - Address: C4:3D:C7:3A:1F:5D Channel:1 Frequency:2.412 GHz (Channel 1) Quality=37/70 Signal level=-73 dBm Encryption key:on ESSID:"my essid" Bit Rates:1 Mb/s; 2 Mb/s; 5.5 Mb/s; 11 Mb/s; 18 Mb/s 24 Mb/s; 36 Mb/s; 54 Mb/s Bit Rates:6 Mb/s; 9 Mb/s; 12 Mb/s; 48 Mb/s Mode:Master Extra:tsf=00000070cca77186 Extra: Last beacon: 5588ms ago IE: Unknown: 0007756E69636F726E IE: Unknown: 010882848B962430486C IE: Unknown: 030101 IE: Unknown: 2A0100 IE: Unknown: 2F0100 IE: IEEE 802.11i/WPA2 Version 1 Group Cipher : TKIP Pairwise Ciphers (2) : CCMP TKIP Authentication Suites (1) : PSK IE: Unknown: 32040C121860 IE: Unknown: 2D1AFC181BFFFF000000000000000000000000000000000000000000 IE: Unknown: 3D1601080400000000000000000000000000000000000000 IE: Unknown: DD7E0050F204104A0001101044000102103B00010310470010F99C335D7BAC57FB00137DFA79600220102100074E657467656172102300074E6574676561721024000631323334353610420007303030303030311054000800060050F20400011011000743473331303144100800022008103C0001011049000600372A000120 IE: Unknown: DD090010180203F02C0000 IE: WPA Version 1 Group Cipher : TKIP Pairwise Ciphers (2) : CCMP TKIP Authentication Suites (1) : PSK IE: Unknown: DD180050F2020101800003A4000027A4000042435E0062322F00 iwconfig: lo no wireless extensions. wlan0 IEEE 802.11bgn ESSID:"my essid" Mode:Managed Frequency:2.412 GHz Access Point: C4:3D:C7:3A:1F:5D Bit Rate=78 Mb/s Tx-Power=20 dBm Retry long limit:7 RTS thr:off Fragment thr:off Power Management:off Link Quality=36/70 Signal level=-74 dBm Rx invalid nwid:0 Rx invalid crypt:0 Rx invalid frag:0 Tx excessive retries:0 Invalid misc:0 Missed beacon:0,

  Read the article

• [c#] SoundPlayer.PlaySync stopping prematurely

  - by JeffE

  I want to play a wav file synchronously on the gui thread, but my call to PlaySync is returning early (and prematurely stopping playback). The wav file is 2-3 minutes. Here's what my code looks like: //in gui code (event handler) //play first audio file JE_SP.playSound("example1.wav"); //do a few other statements doSomethingUnrelated(); //play another audio file JE_SP.playSound("example2.wav"); //library method written by me, called in gui code, but located in another assembly public static int playSound(string wavFile, bool synchronous = true, bool debug = true, string logFile = "", int loadTimeout = FIVE_MINUTES_IN_MS) { SoundPlayer sp = new SoundPlayer(); sp.LoadTimeout = loadTimeout; sp.SoundLocation = wavFile; sp.Load(); switch (synchronous) { case true: sp.PlaySync(); break; case false: sp.Play(); break; } if (debug) { string writeMe = "JE_SP: \r\n\tSoundLocation = " + sp.SoundLocation + "\r\n\t" + "Synchronous = " + synchronous.ToString(); JE_Log.logMessage(writeMe); } sp.Dispose(); sp = null; return 0; } Some things I've thought of are the load timeout, and playing the audio on another thread and then manually 'freeze' the gui by forcing the gui thread to wait for the duration of the sound file. I tried lengthening the load timeout, but that did nothing. I'm not quite sure what the best way to get the duration of a wav file is without using code written by somebody who isn't me/Microsoft. I suppose this can be calculated since I know the file size, and all of the encoding properties (bitrate, sample rate, sample size, etc) are consistent across all files I intend to play. Can somebody elaborate on how to calculate the duration of a wav file using this info? That is, if nobody has an idea about why PlaySync is returning early. Of Note: I encountered a similar problem in VB 6 a while ago, but that was caused by a timeout, which I don't suspect to be a problem here. Shorter (< 1min) files seem to play fine, so I might decide to manually edit the longer files down, then play them separately with multiple calls.

  Read the article

• CodePlex Daily Summary for Monday, July 01, 2013

  CodePlex Daily Summary for Monday, July 01, 2013Popular ReleasesQuickMon: Version 2.10.3: Mainly just a service release - no major changes. Toolbar buttons on main and config window can now be re-arrange (using ALT key) Added property to disable corrective scriptsDotNetNuke® IFrame: IFrame 04.05.00: New DNN6/7 Manifest file and Azure Compatibility.VidCoder: 1.5.2 Beta: Fixed crash on presets with an invalid bitrate.Roadkill - .NET Wiki engine: Roadkill v1.7: New features in 1.7: New file manager: Multiple file uploads Drag and drop uploads Delete folders (admins only) Delete files (admins only) (Experimental) Syntaxhighlighting custom variable (using https://github.com/alexgorbatchev/SyntaxHighlighter) - use [[[code lang=c#|your code here]]] (Experimental) MathJax custom variable - use [[[Mathjax]]] and $$your tex$$ on the page. Improved black bar theme Site speed improvements for Javascript/CSS files - now just two files files ea...Download Sharepoint Solution package: Release 4: version updated for SP2013WinRT XAML Toolkit: WinRT XAML Toolkit - 1.5: WinRT XAML Toolkit based on the Windows 8.0 and 8.1 Preview SDKs. Do not download the source code from here if you are looking for latest updates! You can download the latest source from the SOURCE CODE page. For compiled version use NuGet. You can add it to your project in Visual Studio by going to View/Other Windows/Package Manager Console and entering: PM> Install-Package winrtxamltoolkit Features Attachable Behaviors AwaitableUI extensions Composition library for visual tree rende...Gardens Point LEX: Gardens Point LEX version 1.2.1: The main distribution is a zip file. This contains the binary executable, documentation, source code and the examples. ChangesVersion 1.2.1 has new facilities for defining and manipulating character classes. These changes make the construction of large Unicode character classes more convenient. The runtime code for performing automaton backup has been re-implemented, and is now faster for scanners that need backup. Source CodeThe distribution contains a complete VS2010 project for the appli...ZXMAK2: Version 2.7.5.7: - fix TZX emulation (Bruce Lee, Zynaps) - fix ATM 16 colors for border - add memory module PROFI 512K; add PROFI V03 rom image; fix PROFI 3.XX configTwitter image Downloader: Twitter Image Downloader 2 with Installer: Application file with Install shield and Dot Net 4.0 redistributableUltimate Music Tagger: Ultimate Music Tagger 1.0.0.0: First release of Ultimate Music TaggerBlackJumboDog: Ver5.9.2: 2013.06.28 Ver5.9.2 (1) ??????????(????SMTP?????)?????????? (2) HTTPS???????????Outlook 2013 Add-In: Configuration Form: This new version includes the following changes: - Refactored code a bit. - Removing configuration from main form to gain more space to display items. - Moved configuration to separate form. You can click the little "gear" icon to access the configuration form (still very simple). - Added option to show past day appointments from the selected day (previous in time, that is). - Added some tooltips. You will have to uninstall the previous version (add/remove programs) if you had installed it ...Terminals: Version 3.0 - Release: Changes since version 2.0:Choose 100% portable or installed version Removed connection warning when running RDP 8 (Windows 8) client Fixed Active directory search Extended Active directory search by LDAP filters Fixed single instance mode when running on Windows Terminal server Merged usage of Tags and Groups Added columns sorting option in tables No UAC prompts on Windows 7 Completely new file persistence data layer New MS SQL persistence layer (Store data in SQL database)...NuGet: NuGet 2.6: Released June 26, 2013. Release notes: http://docs.nuget.org/docs/release-notes/nuget-2.6Python Tools for Visual Studio: 2.0 Beta: We’re pleased to announce the release of Python Tools for Visual Studio 2.0 Beta. Python Tools for Visual Studio (PTVS) is an open-source plug-in for Visual Studio which supports programming with the Python language. PTVS supports a broad range of features including CPython/IronPython, Edit/Intellisense/Debug/Profile, Cloud, HPC, IPython, and cross platform debugging support. For a quick overview of the general IDE experience, please watch this video: http://www.youtube.com/watch?v=TuewiStN...Player Framework by Microsoft: Player Framework for Windows 8 and WP8 (v1.3 beta): Preview: New MPEG DASH adaptive streaming plugin for Windows Azure Media Services Preview: New Ultraviolet CFF plugin. Preview: New WP7 version with WP8 compatibility. (source code only) Source code is now available via CodePlex Git Misc bug fixes and improvements: WP8 only: Added optional fullscreen and mute buttons to default xaml JS only: protecting currentTime from returning infinity. Some videos would cause currentTime to be infinity which could cause errors in plugins expectin...AssaultCube Reloaded: 2.5.8: SERVER OWNERS: note that the default maprot has changed once again. Linux has Ubuntu 11.10 32-bit precompiled binaries and Ubuntu 10.10 64-bit precompiled binaries, but you can compile your own as it also contains the source. If you are using Mac or other operating systems, please wait while we continue to try to package for those OSes. Or better yet, try to compile it. If it fails, download a virtual machine. The server pack is ready for both Windows and Linux, but you might need to compi...Microsoft Ajax Minifier: Microsoft Ajax Minifier 4.95: update parser to allow for CSS3 calc( function to nest. add recognition of -pponly (Preprocess-Only) switch in AjaxMinManifestTask build task. Fix crashing bug in EXE when processing a manifest file using the -xml switch and an error message needs to be displayed (like a missing input file). Create separate Clean and Bundle build tasks for working with manifest files (AjaxMinManifestCleanTask and AjaxMinBundleTask). Removed the IsCleanOperation from AjaxMinManifestTask -- use AjaxMinMan...VG-Ripper & PG-Ripper: VG-Ripper 2.9.44: changes NEW: Added Support for "ImgChili.net" links FIXED: Auto UpdaterDocument.Editor: 2013.25: What's new for Document.Editor 2013.25: Improved Spell Check support Improved User Interface Minor Bug Fix's, improvements and speed upsNew ProjectsAerCloud.net Client - Java, Linux & Windows: This project source code provides a step by step guide for using AerCloud.net Framework as a Service API. For more information please visit http://www.aercloudAmiClient – Asterisk Manager Interface (AMI) client based on the Rx Framework: Asterisk Manager Interface (AMI) client based on the Rx Frameworkbaidupan: cdcddddC#??????: C#??????ImageHelper: imagehelperIP switcher: IP switcher is a simple tool for switching settings, and store presets, on networkadapters.MastersProject: A MS project with a goal of creating a fully Code Contracts verified physics engine and a relatively simple game that uses it.Multiplatform card game: Example multipatform project.PhoneTools: A collection of tools designed to help developers create beautiful Windows Phone 8 apps.rodidexter: lllSharePoint 2013 List Item Encryption: This coding exercise project enables you to encrypt/decrypt list item text field in the browser using industry standard algorithms.tvaSoft: simulation, rotor dynamics, Finite Element Analisys, FEM, ODE, torsional vibration, flexural vibrationX3DML Project: X3DML is an xml-based markup language that defines rules for modeling 3D scenes from a tag-based document. It may be usefull in 3D web design and VR.zhuang-tfs: zhuang tfs

  Read the article

< Previous Page | 2 3 4 5 6 7  | Next Page >