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• tod to avi mpg wmv, convert tod (.mpeg-2) to avi mpg wmv for Movie Maker.

  - by yearofhao

  Need to convert .tod (mpeg-2) to avi mpg wmv download from JVC Everio to PC with tod to avi mpg wmv converter convert tod to uncompressed/raw avi, mpg wmv Have a JVC Everio camcorder? Then you may encounter problems when saving the .tod files to your computer windows movie maker says it can't recognize and edit them to make videos. You may play them using media player but the problem is how to edit them? The bundled software Power cinema could be annoying, since you can only edit when the camera is plugged in to the PC - Power cinema can’t seem to edit from the saved clips alone. So, how do you save them to PC so that you can edit them without the camera and also using windows movie maker? JVC Everio Tod to avi mpg wmv converter costs you a penny to but help you perfectly convert tod file to AVI, MPG, WMV, YouTube FLV, MP4, DV, QuickTime.MOV or other common video formats with fast speed and while keeping the original HD quality. High definition TOD recordings from JVC Camcorders can playback fluidly, convert smoothly and edit professionally on with iOrgsoft TOD file converter iOrgsoft tod to avi mpg wmv Converter has been mostly used by Windows users who use Windows 7 or Vista, after the .tod (mpeg-2 the same codec) downloaded from JVC Everio to PC, it’s best to convert tod to avi, convert to xvid divx, convert tod to uncompressed avi or convert tod to raw avi, tod to mpg, tod to wmv, which are three Windows movie maker best formats to import. TOD to avi mpg wmv converter is a competent video-editing program that allows you to clip/cut TOD video clip, crop the video to encode, and help transfers video to devices like iPhone, iPod, to HDTV connected with Apple TV.

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• PHP-FPM issue on LEMP Stack and WordPress

  - by jw60660

  I'm very much a NGINX and Server Admin beginner. I used this tutorial to install NGINX / PHP / mySQL / WordPress: C3M Digital Tutorial In this tutorial the backend php-cgi setup is configured using fastcgi. php5-fpm was installed during this tutorial: apt-get install nginx-full php5-fpm php5 php5-mysql php5-apc php5-mysql php5-xsl php5-xmlrpc php5-sqlite php5-snmp php5-curl After reading that the NGINX configuration on the WordPress codec was more secure than most tutorials, I decided to use the codex configuration: WordPress NGINX configuration in Codex The Codex configuration uses php-fpm for backend php-cgi. When opening the browser I got a 502 Bad Gateway error. The error log was: "2012/06/10 21:18:27 [crit] 14009#0: *4 connect() to unix:/tmp/php-fpm.sock failed (2: No such file or directory) while connecting to upstream, client: 12.3.456.789, server: mywebsite.com, request: "GET / HTTP/1.1", upstream: "fastcgi://unix:/tmp/php-fpm.sock:", hos t: "mywebsite.com"" In the main NGINX configuration file supplied by the codex I noticed the line starting "server unix:" in the upstream php block which point to the empty directory: # Upstream to abstract backend connection(s) for PHP. upstream php { server unix:/tmp/php-fpm.sock; # server 127.0.0.1:9000; } I checked the folder at /tmp and it was empty. Seems I missed configuring php-fpm to play with NGINX. Can someone point me in the right direction? Much appreciated!

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• 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?

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• DV-AVI in DivX Plus Player and VirtualDub - playback issue

  - by user714965

  I have an AVI-video which I had (digitally) transferred from a DV camera to my PC. The video contains errors at the beginning most likely because the DV tape was pretty old. When playing this video in the DivX Plus Player I get some picture artefacts and some high noise peaks of the sound. These stops after two seconds. When I'm playing this video in VirtualDub (where I want to cut it) I get the same picture artefacts. But the sound errors (those loud high peaks) lasts ten seconds. These sound errors are also contained in the cutted video. Why does the video have more errors when played in VirtualDub? I think because of different codecs which are used for decoding the video? How can I change the codec which VirtualDub uses for decoding? I have installed ffdshow for this but it seems that it is not used because I don't get the ffdshow icon in the taskbar when playing the video in VirtualDub. When playing in DivX Player Plus I get this icon.

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• 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

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• 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

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• 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.

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• Is there a good alternative to Videora iPod Converter?

  - by Richard

  I use Videora to convert my videos (in DivX/XviD format) to something I can play on my iPod Classic. I really dislike it. It's clunky, riddled with adverts, sometimes doesn't convert properly (the infamous "invalid public atom" error - see Google for more) and has a UI that truly stinks. On the upside, it's free, accepts a list of video files (via the oddly hard to find "1-click convert" button) and just gets on with the converting as it already knows the correct settings for my iPod. One final nice touch is that once they are converted, it'll automatically upload them into iTunes. Are there any alternatives which have all the upsides but none of the downsides? Bonus points if they can set the metadata in iTunes correctly for TV shows (season, show, episode) and delete the converted file afterwards (as my iTunes settings means that a copy is made elsewhere). I've looked at a bunch of applications (handbrake, virtualdub, mediacoder, format factory, any video converter, convertxtodvd) but many of them fail the "just select a list of files and get on with converting" test - let alone all the other features I want. I have no desire to individually set the video size of each file or the codec or the post-processing options. I'm currently using the command line version of HandBrake (handbrakecli) and a hand-written DOS batch file to go through every file in a folder and convert it. It does most of what I want, just not in a very slick way. Can anyone recommend anything better? It needs to work on Windows 7 and be free.

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• Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

  - by davr

  First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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• ffmpeg cutting video duration

  - by Steve Spence

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

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• How to install/configure ffmpeg to compress mp4 videos for flash player delivery?

  - by Andrew Fulton

  We have a flash web-app that created interactive video, and are using ffmpeg to do some compression/resizing when a user "publishes" their project. The user can upload flv files and mp4 files, both of which play fine in the Flash UI before publishing. After publishing the flv files work fine, but the mp4 files will not play in the flash player: Audio will play but video won't. The mp4 files will play fine if I download them and play them in the Quicktime player but if I attempt to open them in the Adobe Media Player it reports "The media file does not contain a supported video track". If I open the Movie inspector in quicktime it tells me that the original file is an "h264" video and the ffmpeg-processed ones are "mpeg-4". I have tried forcing it to h264 by adding flags like -f h264 and -vcodec h264 but I get a screenfull of errors (no frame, illegal POC type, sps_id out of range) ending with Could not find codec parameters (Video: h264) h264 will show up if I run ffmpeg -formats and ffmpeg -codecs, and as I said it will play fine in Quicktime. Is there anything else I need to do to convince the flash player to play them? Is there anything else I need to tell you about the server that will help?

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• ffmpeg - creating DNxHD MFX files with alphas

  - by Hugh

  I'm struggling with something in FFMpeg at the moment... I'm trying to make DNxHD 1080p/24, 36Mb/s MXF files from a sequence of PNG files. My current command-line is: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mxf -pix_fmt rgb32 -b 36Mb /tmp/temp.mxf To which ffmpeg gives me the output: Input #0, image2, from '/tmp/temp.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mxf, to '/tmp/temp.mxf': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 [mxf @ 0x1005800]unsupported video frame rate Could not write header for output file #0 (incorrect codec parameters ?) There are a few things in here that concern me: The output stream is insisting on being yuv422p, which doesn't support alpha. 24fps is an unsupported video frame rate? I've tried 23.976 too, and get the same thing. I then tried the same thing, but writing to a quicktime (still DNxHD, though) with: ffmpeg -y -f image2 -i /tmp/temp.%04d.png -s 1920x1080 -r 24 -vcodec dnxhd -f mov -pix_fmt rgb32 -b 36Mb /tmp/temp.mov This gives me the output: Input #0, image2, from '/tmp/1274263259.28098.%04d.png': Duration: 00:00:01.60, start: 0.000000, bitrate: N/A Stream #0.0: Video: png, rgb32, 1920x1080, 25 tbr, 25 tbn, 25 tbc Output #0, mov, to '/tmp/1274263259.28098.mov': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 90k tbn, 24 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 39 fps= 9 q=1.0 Lsize= 7177kB time=1.62 bitrate=36180.8kbits/s video:7176kB audio:0kB global headers:0kB muxing overhead 0.013636% Which obviously works, to a certain extent, but still has the issue of being yuv422p, and therefore losing the alpha. If I'm going to QuickTime, then I can get what I need using Shake, but my main aim here is to be able to generate .mxf files. Any thoughts? Thanks

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• How can I get DVDs playing after a Vista to XP change?

  - by Liath

  I replaced my vista install on a Dell Inspiron 1525 with XP and have managed to get most things up and running again however I'm having trouble with playing DVDs. When I try and play a DVD I get the following message: Windows Media Player cannot play this DVD because there is a problem with digital copy protection between your DVD drive, decoder, and video card. Try installing an updated driver for your video card. I have ensured that my drive is configured to play Region 2 discs (I'm in the UK), I've installed the most up to date XP codec pack which makes me think it's a driver issue. In device manager I have got my DVD drivers up to date however under "Other Devices" I'm missing several which sound key: Audio Device on High Definition Audio Bus Modem Device on High Definition Audio Bus Video Controller Video Controller (VGA Compatible) However I've installed all the relevant drivers I can find on the Dell website. The drive itself is working - I've run software from the drive. I'm afraid I am far from a sys-admin so I'm struggling on this one. How can I get my DVDs playing again?

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• How can I get DVDs playing after a Vista to XP change? [closed]

  - by Liath

  I replaced my vista install on a Dell Inspiron 1525 with XP and have managed to get most things up and running again however I'm having trouble with playing DVDs. When I try and play a DVD I get the following message: Windows Media Player cannot play this DVD because there is a problem with digital copy protection between your DVD drive, decoder, and video card. Try installing an updated driver for your video card. I have ensured that my drive is configured to play Region 2 discs (I'm in the UK), I've installed the most up to date XP codec pack which makes me think it's a driver issue. In device manager I have got my DVD drivers up to date however under "Other Devices" I'm missing several which sound key: Audio Device on High Definition Audio Bus Modem Device on High Definition Audio Bus Video Controller Video Controller (VGA Compatible) However I've installed all the relevant drivers I can find on the Dell website. The drive itself is working - I've run software from the drive. I'm afraid I am far from a sys-admin so I'm struggling on this one. How can I get my DVDs playing again?

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• Migrate existing Maven Project into an OSGI Bundle

  - by user1706291

  i am new to the whole OSGi stuff and my task is to create an OSGi Bundle out from an exisitng maven project. To get started i decided to pick the smallest part and starting with it: Here is the pom.xml project xmlns="http://maven.apache.org/POM/4.0.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/maven-v4_0_0.xsd"> <modelVersion>4.0.0</modelVersion> <parent> <artifactId>cross</artifactId> <groupId>net.sf.maltcms</groupId> <version>1.2.12-SNAPSHOT</version> </parent> <artifactId>cross-main</artifactId> <packaging>jar</packaging> <name>cross-main</name> <dependencies> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-annotations</artifactId> <version>${project.version}</version> </dependency> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-event</artifactId> <version>${project.version}</version> </dependency> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-tools</artifactId> <version>${project.version}</version> </dependency> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-exception</artifactId> <version>${project.version}</version> </dependency> <dependency> <groupId>commons-codec</groupId> <artifactId>commons-codec</artifactId> <version>1.4</version> </dependency> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-main-api</artifactId> <version>${project.version}</version> <exclusions> <exclusion> <artifactId>commons-logging</artifactId> <groupId>commons-logging</groupId> </exclusion> </exclusions> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-aop</artifactId> <version>3.0.6.RELEASE</version> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-asm</artifactId> <version>3.0.6.RELEASE</version> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-beans</artifactId> <version>3.0.6.RELEASE</version> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-context</artifactId> <version>3.0.6.RELEASE</version> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-core</artifactId> <version>3.0.6.RELEASE</version> <exclusions> <exclusion> <artifactId>commons-logging</artifactId> <groupId>commons-logging</groupId> </exclusion> </exclusions> </dependency> <dependency> <groupId>org.springframework</groupId> <artifactId>spring-expression</artifactId> <version>3.0.6.RELEASE</version> </dependency> <dependency> <groupId>commons-io</groupId> <artifactId>commons-io</artifactId> <version>2.1</version> </dependency> <dependency> <groupId>net.sf.ehcache</groupId> <artifactId>ehcache-core</artifactId> <version>2.4.6</version> </dependency> <dependency> <groupId>${project.groupId}</groupId> <artifactId>cross-math</artifactId> <version>${project.version}</version> </dependency> <dependency> <groupId>com.db4o</groupId> <artifactId>db4o-all</artifactId> <version>8.0.249</version> </dependency> <dependency> <groupId>net.sf.mpaxs</groupId> <artifactId>mpaxs-spi</artifactId> <version>1.6.10</version> </dependency> <dependency> <groupId>net.sf.mpaxs</groupId> <artifactId>mpaxs-server</artifactId> <version>1.6.10</version> </dependency> </dependencies> I did some research and found the Apache Bundle Plugin for maven and changed the pom to this <packaging>bundle</packaging> and added <build> <plugins> <plugin> <groupId>org.apache.felix</groupId> <artifactId>maven-bundle-plugin</artifactId> <extensions>true</extensions> <configuration> <instructions> <Bundle-SymbolicName>${pom.artifactId}</Bundle-SymbolicName> </instructions> </configuration> </plugin> </plugins> </build> mvn clean install went fine and i got a jar file containing the manifest, but of course the bundle could not be resolved BundleException: The bundle "cross-main_1.2.12.SNAPSHOT [30]" could not be resolved. Reason: Missing Constraint: Import-Package: com.db4o; version="[8.0.0,9.0.0) To make a long story short: What are the possibiliteis to migrate a maven application into an OSGi Bundle? Espacially how to manage the dependencys

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• Development process for an embedded project with significant hardware changes

  - by pierr

  I have a good idea about Agile development process but it seems it does not fit well with a embedded project with significant hardware changes. I will describe below what we are currently doing (Ad-hoc way, no defined process yet). The changes are divided into three categories and different processes are used for each of them: complete hardware change example : use a different video codec IP a) Study the new IP b) RTL/FPGA simulation c) Implement the legacy interface - go to b) d) Wait until hardware (tape out) is ready f) Test on the real hardware hardware improvement example : enhance the image display quality by improving the underlying algorithm a) RTL/FPGA simulation b) Wait until hardware and test on the hardware Minor change example : only change hardware register mapping a) Wait until hardware and test on the hardware The worry is it seems we don't have too much control and confidence about software maturity for the hardware changes as the bring-up schedule is always very tight and the customer desired a seamless change when updating to a new version of hardware. How did you manage this kind of hardware change? Did you solve that by a Hardware Abstraction Layer (HAL)? Did you have a automatic test for the HAL layer? How did you test when the hardware platform is not even ready? Do you have well-documented processes for this kind of change?

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• Django: Unicode Filenames with ASCII headers?

  - by TheLizardKing

  I have a list of strangely encoded files: 02 - Charlie, Woody and You/Study #22.mp3 which I suppose isn't so bad but there are a few particular characters which Django OR nginx seem to be snagging on. >>> test = u'02 - Charlie, Woody and You/Study #22.mp3' >>> test u'02 - Charlie, Woody and You\uff0fStudy #22.mp3' I am using nginx as a reverse proxy to connect to django's built in webserver (still in development stages) and postgresql for my database. My database and tables are all en_US.UTF-8 and I am using pgadmin3 to view my tables outside of django. My issue goes a little beyond my title, firstly how should I be saving possibly whacky filenames in my database? My current method is 'path': smart_unicode(path.lstrip(MUSIC_PATH)), 'filename': smart_unicode(file) and when I pprint out the values they do show u'whateverthecrap' I am not sure if that is how I should be doing it but assuming it is now I have issues trying to spit out the download. My download view looks something like this: def song_download(request, song_id): song = get_object_or_404(Song, pk=song_id) url = u'/static_music/%s/%s' % (song.path, song.filename) print url response = HttpResponse() response['X-Accel-Redirect'] = url response['Content-Type'] = 'audio/mpeg' response['Content-Disposition'] = "attachment; filename=test.mp3" return response and most files will download but when I get to 02 - Charlie, Woody and You/Study #22.mp3 I receive this from django: 'ascii' codec can't encode character u'\uff0f' in position 118: ordinal not in range(128), HTTP response headers must be in US-ASCII format. How can I use an ASCII acceptable string if my filename is out of bounds? 02 - Charlie, Woody and You\uff0fStudy #22.mp3 doesn't seem to work... EDIT 1 I am using Ubuntu for my OS.

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• Easiest way to use speex preprocessor in C#?

  - by eselk

  I need to use the speex preprocessor, and only the preprocessor in my VOIP app (don't need to use the codec). My app is written in C#. I think I know the easiest steps, but not sure where to find these items. Easiest in my opinion, if I can find these: Windows DLL that contains only the preprocessor functions, or if small enough in size I guess the entire speex library would be OK. So far I've only found binarys in EXE format, so if I can't find the binaries I'd have to install the compiler they use to build their source and probably several other libraries (as is my experience with most open source builds). C# versions of the header files, for pinvoking the DLL functions. So my question is, does anyone know where I can find these? I'm sure people have created these before, based on the huge number of speex users, just not able to find any on-line. Hope people don't think I'm lazy, I just hate doing this kind of "busy work" if I know many others have probably already done the exact same thing :) Update: I did find http://www.rarewares.org/files/others/libspeex-dll-1.2rc1.zip which includes libspeex.dll, but the DLL has no exports so not sure how they expect that to work. The other binaries they have are also just EXEs.

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• Windows-Mobile Directshow: Specifying bitrate/quality of a WMV video capture

  - by Landstander

  Hi- I'm stumped on this, and I'm really hoping someone could point me in the right direction. I'm currently capturing video in Windows Mobile and encoding it using the WMV 9 DMO (CLSID_CWMV9EncMediaObject). That all works well enough, but the output video's bitrate is too high, resulting in a video file that's much too large for my needs. Ultimately, my goal is to mimic the video settings that Microsoft's Camera Capture Dialog outputs in the "messaging" quality mode (64kbps) from my C++ code. Currently, my code's outputting a WMV file with a bitrate of 352kbps. The only example I could find of specifying the capture bitrate with a WMV9 DMO was this. The idea in that code was basically to use a propertybag to write a bitrate to a property of the DMO. Update: In windows mobile, the closest codec property I can find that seems to equate to the bitrate is "g_wszWMVCVBRQuality". Microsoft's documentation of this property is extremely confusing to me: It basically seems to say that a higher number equates to a higher quality, but it gives absolutely no explanation of the specifics for each number. When I attempt to set this property to value like "1" via a propertybag for the WMV9 DMO, I run into a -2147467259 (unknown) error. To summarize: What is the basic strategy to specify the bitrate/quality of a video being captured via directshow (wmv9) on a windows mobile platform? I've heard (or wondered about) the following methods: Use the propertybag to change the encoder DMO's property that corresponds to bitrate/quality (currently failing) Create your own custom transcoder/encoder to specify it. This seems unnecessary since the WMV encoder works well enough- it's just at too high a bitrate. The VIDEOINFOHEADER has a bitrate property, but I suspect that specifying new settings here will do nothing to alter the actual encoding process since I wouldn't think file attributes would come into play until after the encoding. Any suggestions? PS: I would post specific source code, but at this point it may confuse more than it helps since I'm floundering so much on how to do this. At this point, I'm just trying to validate the general strategy. THANKS!

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• android compile error: could not reserve enough space for object heap

  - by moonlightcheese

  I'm getting this error during compilation: Error occurred during initialization of VM Could not create the Java virtual machine. Could not reserve enough space for object heap What's worse, the error occurs intermittently. Sometimes it happens, sometimes it doesn't. It seems to be dependent on the amount of code in the application. If I get rid of some variables or drop some imported libraries, it will compile. Then when I add more to it, I get the error again. I've included the following sources into the application in the [project_root]/src/ directory: org.apache.httpclient (I've stripped all references to log4j from the sources, so don't need it) org.apache.codec (as a dependency) org.apache.httpcore (dependency of httpclient) and my own activity code consisting of nothing more than an instance of HttpClient. I know this has something to do with the amount of memory necessary during compile time or some compiler options, and I'm not really stressing my system while i'm coding. I've got 2GB of memory on this Core Duo laptop and windows reports only 860MB page file usage (haven't used any other memory tools. I should have plenty of memory and processing power for this... and I'm only compiling some common http libs... total of 406 source files. What gives? Android API Level: 5 Android SDK rel 5 JDK version: 1.6.0_12

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• android compile error: could not reserve enough space for object heap

  - by moonlightcheese

  I'm getting this error during compilation: Error occurred during initialization of VM Could not create the Java virtual machine. Could not reserve enough space for object heap What's worse, the error occurs intermittently. Sometimes it happens, sometimes it doesn't. It seems to be dependent on the amount of code in the application. If I get rid of some variables or drop some imported libraries, it will compile. Then when I add more to it, I get the error again. I've included the following sources into the application in the [project_root]/src/ directory: org.apache.httpclient (I've stripped all references to log4j from the sources, so don't need it) org.apache.codec (as a dependency) org.apache.httpcore (dependency of httpclient) and my own activity code consisting of nothing more than an instance of HttpClient. I know this has something to do with the amount of memory necessary during compile time or some compiler options, and I'm not really stressing my system while i'm coding. I've got 2GB of memory on this Core Duo laptop and windows reports only 860MB page file usage (haven't used any other memory tools. I should have plenty of memory and processing power for this... and I'm only compiling some common http libs... total of 406 source files. What gives? edit (4/30/2010-18:24): Just compiled some code where I got the above stated error. I closed some web browser windows and recompiled the same exact code with no edits and it compiled with no issue. this is definitely a compiler issue related to memory usage. Any help would be great.... because I have no idea where to go from here. Android API Level: 5 Android SDK rel 5 JDK version: 1.6.0_12 Sorry I had to repost this question because regardless of whether I use the native HttpClient class in the Android SDK or my custom version downloaded from apache, the error still occurs.

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• H.264 / FLV best practices for HTML

  - by Steve Murch

  I run a website with about 700 videos (And no, it's not porn -- get your mind out of the gutter :-) ). The videos are currently in FLV format. We use the JWPlayer to render those videos. IIS6 hosted. Everything works just fine. As I understand it, H.264 (not FLV and likely not OGG) is the emerging preferred HTML5 video standard. Today, the iPad really only respects H.264 or YouTube. Presumably, soon many more important browsers will follow Apple's lead and respect only the HTML5 tag. OK, so I think I can figure out how to convert my existing videos into the proper H.264 format. There are various tools available, including ffmpeg.exe. I haven't tried it yet, but I don't think that's going to be a problem after fiddling with the codec settings. My question is more about the container itself -- that is, planning graceful transition for all users. What's the best-practice recommendation for rendering these videos? If I just use the HTML5 tag, then presumably any browser that doesn't yet support HTML5 won't see the videos. And if I render them in Flash format via the JWPlayer or some other player, then they won't be playable on the iPad. Do I have to do ugly UserAgent detection here to figure out what to render? I know the JWPlayer supports H.264 media, but isn't the player itself a Flash component and therefore not playable on the iPad? Sorry if I'm not being clear, but I'm scratching my head on a graceful transition plan that will work for current browsers, the iPad and the upcoming HTML5 wave. I'm not a video expert, so any advice would be most welcome, thanks.

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• ffmpeg(libavcodec). memory leaks in avcodec_encode_video

  - by gavlig

  I'm trying to transcode a video with help of libavcodec. On transcoding big video files(hour or more) i get huge memory leaks in avcodec_encode_video. I have tried to debug it, but with different video files different functions produce leaks, i have got a little bit confused about that :). [Here] (http://stackoverflow.com/questions/4201481/ffmpeg-with-qt-memory-leak) is the same issue that i have, but i have no idea how did that person solve it. QtFFmpegwrapper seems to do the same i do(or i missed something). my method is lower. I took care about aFrame and aPacket outside with av_free and av_free_packet. int Videocut::encode( AVStream *anOutputStream, AVFrame *aFrame, AVPacket *aPacket ) { AVCodecContext *outputCodec = anOutputStream->codec; if (!anOutputStream || !aFrame || !aPacket) { return 1; /* NOTREACHED */ } uint8_t * buffer = (uint8_t *)malloc( sizeof(uint8_t) * _DefaultEncodeBufferSize ); if (NULL == buffer) { return 2; /* NOTREACHED */ } int packetSize = avcodec_encode_video( outputCodec, buffer, _DefaultEncodeBufferSize, aFrame ); if (packetSize < 0) { free(buffer); return 1; /* NOTREACHED */ } aPacket->data = buffer; aPacket->size = packetSize; return 0; }

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• Python interface to PayPal - urllib.urlencode non-ASCII characters failing

  - by krys

  I am trying to implement PayPal IPN functionality. The basic protocol is as such: The client is redirected from my site to PayPal's site to complete payment. He logs into his account, authorizes payment. PayPal calls a page on my server passing in details as POST. Details include a person's name, address, and payment info etc. I need to call a URL on PayPal's site internally from my processing page passing back all the params that were passed in abovem and an additional one called 'cmd' with a value of '_notify-validate'. When I try to urllib.urlencode the params which PayPal has sent to me, I get a: While calling send_response_to_paypal. Traceback (most recent call last): File "<snip>/account/paypal/views.py", line 108, in process_paypal_ipn verify_result = send_response_to_paypal(params) File "<snip>/account/paypal/views.py", line 41, in send_response_to_paypal params = urllib.urlencode(params) File "/usr/local/lib/python2.6/urllib.py", line 1261, in urlencode v = quote_plus(str(v)) UnicodeEncodeError: 'ascii' codec can't encode character u'\ufffd' in position 9: ordinal not in range(128) I understand that urlencode does ASCII encoding, and in certain cases, a user's contact info can contain non-ASCII characters. This is understandable. My question is, how do I encode non-ASCII characters for POSTing to a URL using urllib2.urlopen(req) (or other method) Details: I read the params in PayPal's original request as follows (the GET is for testing): def read_ipn_params(request): if request.POST: params= request.POST.copy() if "ipn_auth" in request.GET: params["ipn_auth"]=request.GET["ipn_auth"] return params else: return request.GET.copy() The code I use for sending back the request to PayPal from the processing page is: def send_response_to_paypal(params): params['cmd']='_notify-validate' params = urllib.urlencode(params) req = urllib2.Request(PAYPAL_API_WEBSITE, params) req.add_header("Content-type", "application/x-www-form-urlencoded") response = urllib2.urlopen(req) status = response.read() if not status == "VERIFIED": logging.warn("PayPal cannot verify IPN responses: " + status) return False return True Obviously, the problem only arises if someone's name or address or other field used for the PayPal payment does not fall into the ASCII range.

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