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• Best way to convert episodic DVDs for Windows Media Center?

  - by Roger Lipscombe

  I'm archiving my DVD collection. My goal is to be able to play them back in Windows Media Center. For feature-length DVDs, I'm using AnyDVD and CloneDVD, which is working well. For playing back TV shows (and other episodic content), I'm using Media Browser, which doesn't support a VIDEO_TS folder per episode. It expects the shows to be broken up into one file per episode (e.g. "Willo the Wisp - S01E12.avi"). For this, I'm attempting to use Handbrake, which, for extracting the episodes from DVD (or already-ripped VIDEO_TS folder), is working pretty well. The problem that I have is that the default x264 encoder over-compresses the resulting video stream, which results in hideous artifacts in animated shows. The aforementioned Willo the Wisp is a particularly bad example, because the original DVD is particularly "noisy". If I switch to using the ffmpeg encoder, the artifacts are gone in Windows Media Player, but I can't get the resulting files to play back in Windows Media Center. I see the first frame, and then there's an error message. I've installed the CCCP codec collection, but it doesn't seem to have made any difference. So: what's the best way to convert VIDEO_TS to individual episode files for playback in Windows Media Center?

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• What is the fastest and best way to convert an rmvb video to mp4/mkv without losing any quality?

  - by Eric Leung

  the file will be played in a popbox3d. my old method was to convert the video using vidcoder (an offshoot of handbrake) using normal settings, but i've recently confirmed that this significantly reduces video and audio quality. i bumped up the conversion quality to 'high profile' and this produced a higher quality video but raised the conversion time to about twice the video length (95 minutes to convert a 45 minute video) on a core2duo laptop. this is less than ideal when a large number of videos need to be converted. i have tried a direct remuxing using mkv toolnix but this produced a video that refused to display video on the popbox3d, which is consistent with the reported: [quote=other old thread] it is possible to put RealMedia A/V in MKV container (used MKVtoolnix) - however, it is awkward to play later. RV40 is only suspected to be based on H.264 - simplify, is not consistent with MPEG-4 AVC specification. [/quote] i have read that ... [quote=from old thread] Under normal circumstances, [ffmpeg] should convert the video to .video.mp4 and the audio to (.wav then to) .audio.mp4, then mux the video and audio into a new .mp4 file and delete the temporary video-only and audio-only files.[/quote] and i am currently attempting to discover how this is done. help? PS: i download a lot of series from asia and for some strange reason, rmvb is a really popular format over there. sometimes, it's the only format that's available. unfortunately, it's a format that is incompatible with the popbox3d, so i have to convert the files before i can watch them on my tv.

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• Starting my own server - basic recommendations and questions [closed]

  - by Ilia Rostovtsev

  Possible Duplicate: Can you help me with my capacity planning? I'm planning to start my own high-performance server and then use collocation services for keeping it up and running. I'm planning to USE it for processing videos and keeping big video site up! (using FFMpeg, MENcoder and etc.) I just need recommendations on whether listed hardware is good enough and will work together well and fast enough. Do I need anything else (missed something). I remember about CPU coolers though! ;) I'm planning to use SSD drives so please tell me if it's going to work just as regular HDDs (but much faster)? Are they going to be used as RAID (is this possible for SSDs)? Here is what I would like to get: Intel ® Server System SR1600URHSR (Urbanna) or Intel® Server System SR1695WBAC 2 x Intel Xeon X5650 4 x 16Gb DDR-III 1333MHz Kingston ECC Reg (KVR13R9D4/16) 3 x (or maybe 4x) 480Gb SSD Intel 520 Series (SSDSC2CW480A3K5) Which server system would be better? Is listed hardware new/good enough and worth buying it at the moment? Should I probably take a look at something slightly more expensive but more up to date and powerful, may be? After all as software I would like to use CentOS 6 64 bit + WHM/CPanel? Any other suggestions on maybe cheaper and same/more powerful server management system but WHM? What most important points to keep in mind when starting/maintaining your own server?

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• Split MPEG video from command line?

  - by Tim

  I have a homemade DVD that I'm effectively trying to insert chapters into and rearrange - the original author burned it as one long chapter, and I'd like to rip it into smaller pieces and re-encode it into a new DVD. I ripped the DVD with the following command: mplayer dvd:// -dvd-device /dev/sr2 -dumpstream -dumpfile raw.vob I'm running Gentoo Linux with mplayer version 1.0-rc2_p20090731 (the latest available in Portage). I have a list of times that the chapters are supposed to span (for example 30:11-33:25), so my first thought was to rip the entire DVD and use mpgtx to cut out certain pieces of the file. My issue is that running mpgtx -i on the file reports quite a few timestamp jumps: Time stamps jumped from 59.753789 to 0.001622 at position 1d29800 Time stamps jumped from 204963823030450.343750 to 31.165900 at position 2d4f800 Time stamps jumped from 60.077878 to 0.001622 at position 43cc000 Time stamps jumped from 60.024233 to 0.001622 at position 65c5000 Time stamps jumped from 204963823068631.718750 to 52.549244 at position 7fd1000 I've tried to fix the indexes using: mencoder raw.vob -oac copy -ovc copy -forceidx -o fixed.vob -of mpeg But mpgtx will still report timestamp issues. My immediate question: is there a way to take the ripped movie I have and correct its timestamps so I can cut it with mpgtx? If I can get that one issue out of the way, building the rest of the DVD will be smooth sailing. If it's not possible to fix the timestamps on this file: is there a better way to rip small chunks of the DVD into separate files for recompilation later? I'd very much like this to be done on Linux, and it'd be even better if I could script it somehow (feed in a list of start and end positions, or start times and durations, and get out a series of ripped files). If need be, I also have a Mac OS X machine available, but no Windows. Edit: I wound up finding another solution involving HandBrake and ffmpeg (with help from this question), but the question stands. Edit again: Turns out my other solution didn't quite work - the audio desynchronized by about five seconds, in about half of my cut mpgs - so I'm back to square one. Anyone?

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• Errors with Using Webcam

  - by C.G.

  I have been having some issues accessing a webcam from my machine. Sometimes (not always) when I run a program that accesses the device (cheese, guvcview, and code using openCV), I get either of two messages, which lead to the program crashing. The first occurs after running the webcam for some time. libv4l2: error dequeuing buf: No such device VIDIOC_DQBUF: No such device The other will occur without even letting me have a chance to run the webcam. libv4l2: error turning on stream: No space left on device VIDIOC_STREAMON - Unable to start capture: No space left on device Occasionally after getting these errors I will also receive a message saying that no such device can be found for subsequent runs. Other than the times that the "No device found" message appears the webcam appears when I use lsusb. My machine runs Linux Fedora 16, and the webcam is a Logitech C920. I do have ffmpeg installed, and I have been able to run the web camera many times in the past without errors. What is particularly puzzling about these errors is that they just sprung up this past weekend. No new software or hardware has been installed on this machine recently; I haven't changed any settings recently either. It could possibly be a driver issue, but I don't know what could have changed which could lead to this issue. Any attempts at researching this problem has been fruitless as this seems to most commonly occur with multiple webcams. I am only working with one device. I'd appreciate any advice for this problem, as this has become a bit frustrating.

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• Cut (smart edit) .mts (AVCHD Progressive) files un Ubuntu Lucid

  - by pts

  I have a bunch of .mts files containing AVCHD Progressive video recorded by a Panasonic camera, and I need software on Ubuntu Lucid with which I can remove the boring parts, and concatenate the interesting parts, all this without reencoding the video stream. It's OK for me to cut at keyframe boundary. If Avidemux was able to open the files, it would take about 60 hours of work for me to cut the files. (At least that was it last time I tried with similar videos, but of a file format supported by Avidemux.) So I need a fast, powerful and stable video editor, because I don't want that 60 hours of work go up to 240 or even 480 hours just because the tool is too slow or unstable or has a terrible UI. I've tried Avidemux 2.5.5 and 2.5.6, but they crash trying to open such a file, even if I convert the file to .avi first using mencoder -oac copy -ovc copy. mplayer can play the files. I've tried Avidemux 2.6.0, which can open the file, but it cannot jump to the previous or next keyframe etc. (if I make it jump to the next keyframe, and then to the previous keyframe, it doesn't end up at the original keyframe, sometimes displays an error etc.). Also I'm not sure if Avidemux 2.6.x would let me save the result without reencoding. I've tried Kdenlive 0.7.7.1, but playback is very choppy, and it cannot play audio at all (complaining that SDL cannot find the device; but many other programs on the system can play audio). It would be a pain to work with. I've tried converting the .mts file to .mkv using ffmpeg -i input.mts -vcodec copy -sameq -acodec copy -f matroska output.mkv, but that caused too much visible distortions in the video in both mplayer and Avidemux. I've tried converting the .mts file with TsRemux.exe, but Avidemux 2.5.x still can't open that file. Is there another program to cut and concatenate the files? Is there a preprocessor which would create a file (without reencoding the video) on which Avidemux wouldn't crash?

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• MediaElement.js setSrc() Loading The File But Not Changing pluginType

  - by doubleJ

  I'm working on a page that uses mediaelement.js to play mp3/mp4/wmv (yes, we have a lot of wmv). I have a list of links and those links should change the player. My effort is to make the changes to the player through javascript so that the page doesn't refresh. This code is working, but it refreshes every time. See a live demo of the non-ajax version. <?php $file = null; $file = $_GET["file"]; $format = null; if (preg_match("/mp4/i", $file)) $format = "mp4"; if (preg_match("/webm/i", $file)) $format = "webm"; if (preg_match("/wmv/i", $file)) $format = "wmv"; if (preg_match("/mp3/i", $file)) $format = "mp3"; if (preg_match("/ogg/i", $file)) $format = "ogg"; $mime = null; if ($format == "mp4") $mime = "video/mp4"; if ($format == "webm") $mime = "video/webm"; if ($format == "wmv") $mime = "video/wmv"; if ($format == "mp3") $mime = "audio/mp3"; if ($format == "ogg") $mime = "audio/ogg"; $element = "video"; if ($format == "mp3" || $format == "ogg") $element = "audio"; // you have to escape (\) the escape (\) character (hehehe...) $poster = "media\\120701Video.jpg"; $height = "360"; if ($format == "mp3") $height = "30"; ?> <!doctype html> <html> <head> <meta charset="utf-8"> <title>Embed</title> <link rel="stylesheet" href="include/johndyer-mediaelement-b090320/build/mediaelementplayer.min.css"> <style> audio {width:640px; height:30px;} video {width:640px; height:360px;} </style> <script src="include/johndyer-mediaelement-b090320/build/jquery.js"></script> <script src="include/johndyer-mediaelement-b090320/build/mediaelement-and-player.js"></script> </head> <body> <ul> <li><a href="embed.php">Reset</a></li> <li><a href="?file=media/120701Video-AnyVideoConverter.mp4">Alternative (mp4)</a></li> <li><a href="?file=media/120701Video-Ffmpeg-Defaults.webm">Alternative (webm)</a></li> <li><a href="?file=media/AreYouHurting-Death.wmv">Alternative (wmv)</a><li> <li><a href="?file=media/AreYouHurting-Death.mp3">Alternative (mp3)</a></li> </ul> <?php if ($file) { ?> <video src="<?php echo $file; ?>" controls poster="<?php echo $poster; ?>" width="640" height="360"></video> <div id="type"></div> <script> var video = document.getElementsByTagName("video")[0]; var player = new MediaElementPlayer(video, { success: function(player) { $('#type').html(player.pluginType); } }); <?php } ?> </script> </body> </html> This code requires <video> to be loaded, initially and with a file, so that the player mode (pluginType) is set. It will, then, only play formats that the pre-established mode supports (firefox in native mode won't play mp4). See a live demo of the ajax version. <!doctype html> <html> <head> <meta charset="utf-8"> <title>Embed</title> <link rel="stylesheet" href="http://www.mediaelementjs.com/js/mejs-2.9.2/mediaelementplayer.min.css"> <script src="//ajax.googleapis.com/ajax/libs/jquery/1.7.2/jquery.min.js"></script> <script src="http://www.mediaelementjs.com/js/mejs-2.9.2/mediaelement-and-player.js"></script> </head> <body> <ul> <li><a href="javascript:player.pause(); player.setSrc('media/120701Video-AnyVideoConverter.mp4'); player.load(); player.play();">Alternative (mp4)</a></li> <li><a href="javascript:player.pause(); player.setSrc('media/120701Video-Ffmpeg-Defaults.webm'); player.load(); player.play();">Alternative (webm)</a></li> <li><a href="javascript:player.pause(); player.setSrc('media/AreYouHurting-Death.wmv'); player.load(); player.play();">Alternative (wmv)</a></li> <li><a href="javascript:player.pause(); player.setSrc('media/AreYouHurting-Death.mp3'); player.load(); player.play();">Alternative (mp3)</a></li> </ul> <video controls src="media/WordProductionCenter.mp4"></video> <div id="type"></div> <script> var video = document.getElementsByTagName("video")[0]; var player = new MediaElementPlayer(video, { success: function(player) { $('#type').html(player.pluginType); } }); </script> </body> </html> It seems like I need something like setType(), but I see no such option. I've read a couple pages that referenced refreshing the DOM after the javascript runs, but I haven't been able to successfully do it (I know enough about javascript to hack things around and get stuff working, but not enough to create whole new things). It is worth noting that Silverlight doesn't work with Internet Explorer 8 or Safari (not sure if it's my code, mejs, or the browsers). Also, neither Silverlight nor Flash play mp3 or webm (again, not sure where the problem lies). Is there a way to dynamically load different types of files into mediaelement?

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• CDN on Hosted Service in Windows Azure

  - by Shaun

  Yesterday I told Wang Tao, an annoying colleague sitting beside me, about how to make the static content enable the CDN in his website which had just been published on Windows Azure. The approach would be Move the static content, the images, CSS files, etc. into the blob storage. Enable the CDN on his storage account. Change the URL of those static files to the CDN URL. I think these are the very common steps when using CDN. But this morning I found that the new Windows Azure SDK 1.4 and new Windows Azure Developer Portal had just been published announced at the Windows Azure Blog. One of the new features in this release is about the CDN, which means we can enabled the CDN not only for a storage account, but a hosted service as well. Within this new feature the steps I mentioned above would be turned simpler a lot.   Enable CDN for Hosted Service To enable the CDN for a hosted service we just need to log on the Windows Azure Developer Portal. Under the “Hosted Services, Storage Accounts & CDN” item we will find a new menu on the left hand side said “CDN”, where we can manage the CDN for storage account and hosted service. As we can see the hosted services and storage accounts are all listed in my subscriptions. To enable a CDN for a hosted service is veru simple, just select a hosted service and click the New Endpoint button on top. In this dialog we can select the subscription and the storage account, or the hosted service we want the CDN to be enabled. If we selected the hosted service, like I did in the image above, the “Source URL for the CDN endpoint” will be shown automatically. This means the windows azure platform will make all contents under the “/cdn” folder as CDN enabled. But we cannot change the value at the moment. The following 3 checkboxes next to the URL are: Enable CDN: Enable or disable the CDN. HTTPS: If we need to use HTTPS connections check it. Query String: If we are caching content from a hosted service and we are using query strings to specify the content to be retrieved, check it. Just click the “Create” button to let the windows azure create the CDN for our hosted service. The CDN would be available within 60 minutes as Microsoft mentioned. My experience is that about 15 minutes the CDN could be used and we can find the CDN URL in the portal as well.   Put the Content in CDN in Hosted Service Let’s create a simple windows azure project in Visual Studio with a MVC 2 Web Role. When we created the CDN mentioned above the source URL of CDN endpoint would be under the “/cdn” folder. So in the Visual Studio we create a folder under the website named “cdn” and put some static files there. Then all these files would be cached by CDN if we use the CDN endpoint. The CDN of the hosted service can cache some kind of “dynamic” result with the Query String feature enabled. We create a controller named CdnController and a GetNumber action in it. The routed URL of this controller would be /Cdn/GetNumber which can be CDN-ed as well since the URL said it’s under the “/cdn” folder. In the GetNumber action we just put a number value which specified by parameter into the view model, then the URL could be like /Cdn/GetNumber?number=2. 1: using System; 2: using System.Collections.Generic; 3: using System.Linq; 4: using System.Web; 5: using System.Web.Mvc; 6:  7: namespace MvcWebRole1.Controllers 8: { 9: public class CdnController : Controller 10: { 11: // 12: // GET: /Cdn/ 13:  14: public ActionResult GetNumber(int number) 15: { 16: return View(number); 17: } 18:  19: } 20: } And we add a view to display the number which is super simple. 1: <%@ Page Title="" Language="C#" MasterPageFile="~/Views/Shared/Site.Master" Inherits="System.Web.Mvc.ViewPage<int>" %> 2:  3: <asp:Content ID="Content1" ContentPlaceHolderID="TitleContent" runat="server"> 4: GetNumber 5: </asp:Content> 6:  7: <asp:Content ID="Content2" ContentPlaceHolderID="MainContent" runat="server"> 8:  9: <h2>The number is: <% 1: : Model.ToString() %></h2> 10:  11: </asp:Content> Since this action is under the CdnController the URL would be under the “/cdn” folder which means it can be CDN-ed. And since we checked the “Query String” the content of this dynamic page will be cached by its query string. So if I use the CDN URL, http://az25311.vo.msecnd.net/GetNumber?number=2, the CDN will firstly check if there’s any content cached with the key “GetNumber?number=2”. If yes then the CDN will return the content directly; otherwise it will connect to the hosted service, http://aurora-sys.cloudapp.net/Cdn/GetNumber?number=2, and then send the result back to the browser and cached in CDN. But to be notice that the query string are treated as string when used by the key of CDN element. This means the URLs below would be cached in 2 elements in CDN: http://az25311.vo.msecnd.net/GetNumber?number=2&page=1 http://az25311.vo.msecnd.net/GetNumber?page=1&number=2 The final step is to upload the project onto azure. Test the Hosted Service CDN After published the project on azure, we can use the CDN in the website. The CDN endpoint we had created is az25311.vo.msecnd.net so all files under the “/cdn” folder can be requested with it. Let’s have a try on the sample.htm and c_great_wall.jpg static files. Also we can request the dynamic page GetNumber with the query string with the CDN endpoint. And if we refresh this page it will be shown very quickly since the content comes from the CDN without MCV server side process. We style of this page was missing. This is because the CSS file was not includes in the “/cdn” folder so the page cannot retrieve the CSS file from the CDN URL.   Summary In this post I introduced the new feature in Windows Azure CDN with the release of Windows Azure SDK 1.4 and new Developer Portal. With the CDN of the Hosted Service we can just put the static resources under a “/cdn” folder so that the CDN can cache them automatically and no need to put then into the blob storage. Also it support caching the dynamic content with the Query String feature. So that we can cache some parts of the web page by using the UserController and CDN. For example we can cache the log on user control in the master page so that the log on part will be loaded super-fast. There are some other new features within this release you can find here. And for more detailed information about the Windows Azure CDN please have a look here as well.   Hope this helps, Shaun All documents and related graphics, codes are provided "AS IS" without warranty of any kind. Copyright © Shaun Ziyan Xu. This work is licensed under the Creative Commons License.

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• How does one convert 16-bit RGB565 to 24-bit RGB888?

  - by jleedev

  I’ve got my hands on a 16-bit rgb565 image (specifically, an Android framebuffer dump), and I would like to convert it to 24-bit rgb888 for viewing on a normal monitor. The question is, how does one convert a 5- or 6-bit channel to 8 bits? The obvious answer is to shift it. I started out by writing this: uint16_t buf; while (read(0, &buf, sizeof buf)) { unsigned char red = (buf & 0xf800) >> 11; unsigned char green = (buf & 0x07c0) >> 5; unsigned char blue = buf & 0x003f; putchar(red << 3); putchar(green << 2); putchar(blue << 3); } However, this doesn’t have one property I would like, which is for 0xffff to map to 0xffffff, instead of 0xf8fcf8. I need to expand the value in some way, but I’m not sure how that should work. The Android SDK comes with a tool called ddms (Dalvik Debug Monitor) that takes screen captures. As far as I can tell from reading the code, it implements the same logic; yet its screenshots are coming out different, and white is mapping to white. Here’s the raw framebuffer, the smart conversion by ddms, and the dumb conversion by the above algorithm. (By the way, this conversion is implemented in ffmpeg, but it’s just performing the dumb conversion listed above, leaving the LSBs at all zero.) I guess I have two questions: What’s the most sensible way to convert rgb565 to rgb888? How is DDMS converting its screenshots?

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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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• VLC desktop streaming

  - by StackedCrooked

  Edit I stopped using VLC and switched to GMax FLV Encoder. It does a much better job IMO. Original post I am sending my desktop (screen) as an H264 video stream to another machine that saves it to a file using the follwoing command lines: Sender of the stream: vlc -I dummy --sout='#transcode{vcodec=h264,vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' Receiver of the stream: vlc -I rc rtp://@:4444 --sout='#std{access=file,mux=ps,dst=/home/user/output.mp4}' --ipv4 This works, but there are a few issues: The file is not playable with most players. VLC is able to playback the file but with some weirdness: = it takes about 10 seconds before the playback actually begins. = seeking doesn't work. Can someone point me in the right direction on how to fix these issues? EDIT: I made a little progress. The initial delay in playback is because the player is waiting for a keyframe. By forcing the sender of the stream to create a new key-frame every 4 seconds I could decrease the delay: :screen-fps=10 --sout='#transcode{vcodec=h264,venc=x264{keyint=40},vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' The seeking problem is not solved however, but I understand it a little better. The RTP stream is saved as a file in its original streaming format, which is normally not playable as a regular video file. VLC manages to play this file, but most other players don't. So I need to convert it to a regular video file. I am currently investigating whether I can do this with ffmpeg if I provide it with an SDP file for the recorded stream. All help is welcome!

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• 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

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• 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

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• How to add clickable logo on the flash web player?

  - by kishore

  Hi All, I have a code like this <script type="text/javascript" src="http://www.clipul.com/play/swfobject.js"></script> <div id="player">This text will be replaced</div> <script type="text/javascript"> var so = new SWFObject('http://montsmile.com/jwplayer/player.swf','mpl','480','380','8'); so.addParam('allowscriptaccess','always'); so.addParam('allowfullscreen','true'); so.addVariable('height','310'); so.addVariable('width','470'); so.addVariable('file','http://localhost:81/newtip/<?=$videopath1?>'); so.addVariable('logo','http://localhost:81/newtip/ffmpeg/logo2.jpg'); so.addVariable("ClickURL", "http://www.google.com"); so.addVariable('captions','/upload/corrie.xml'); so.addVariable('link','<?=$full_url.'tip.php?vid='.$row_video['vid']?>'); so.addVariable('linkfromdisplay','true'); so.addVariable('linktarget','_blank'); so.addVariable('searchbar','false'); so.addVariable('skin','http://montsmile.com/jwplayer/player.swf'); so.write('player'); </script> It is displaying a player and playing the video. If I Click on the Logo It has to go to www.google.com. But when I click on the logo it becomes pause. Help me how to add clickable Logo on the player

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• 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

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• Java CORBA Client Disconnects Immediately

  - by Benny

  I have built a Java CORBA application that subscribes to an event server. The application narrows and logs on just fine, but as soon as an event is sent to the client, it breaks with the error below. Please advise. 2010/04/25!13.00.00!E00555!enserver!EventServiceIF_i.cpp!655!PID(7390)!enserver - e._info=system exception, ID 'IDL:omg.org/CORBA/TRANSIENT:1.0' TAO exception, minor code = 54410093 (invocation connect failed; ECONNRESET), completed = NO EDIT: Please note, this only happens when running on some machines. It works on some, but not others. Even on the same platform (I've tried Windows XP/7 and CentOS linux) Some work, some don't... Here is the WireShark output...looks like the working PC is much more interactive with the network compared to the non-working PC. Working PC No. Time Source Destination Protocol Info 62 28.837255 10.10.10.209 10.10.10.250 TCP 50169 > 23120 [SYN] Seq=0 Win=8192 Len=0 MSS=1260 WS=8 63 28.907068 fe80::5de0:8d21:937e:c649 ff02::1:3 LLMNR Standard query A isatap 64 28.907166 10.10.10.209 224.0.0.252 LLMNR Standard query A isatap 65 29.107259 10.10.10.209 10.255.255.255 NBNS Name query NB ISATAP<00> 66 29.227000 10.10.10.250 10.10.10.209 TCP 23120 > 50169 [SYN, ACK] Seq=0 Ack=1 Win=32768 Len=0 MSS=1260 WS=0 67 29.227032 10.10.10.209 10.10.10.250 TCP 50169 > 23120 [ACK] Seq=1 Ack=1 Win=66560 Len=0 68 29.238063 10.10.10.209 10.10.10.250 GIOP GIOP 1.1 Request s=326 id=5 (two-way): op=logon 69 29.291765 10.10.10.250 10.10.10.209 GIOP GIOP 1.1 Reply s=420 id=5: No Exception 70 29.301395 10.10.10.209 10.10.10.250 GIOP GIOP 1.1 Request s=369 id=6 (two-way): op=registerEventStat 71 29.348275 10.10.10.250 10.10.10.209 GIOP GIOP 1.1 Reply s=60 id=6: No Exception 72 29.405250 10.10.10.209 10.10.10.250 TCP 50170 > telnet [SYN] Seq=0 Win=8192 Len=0 MSS=1260 WS=8 73 29.446055 10.10.10.250 10.10.10.209 TCP telnet > 50170 [SYN, ACK] Seq=0 Ack=1 Win=32768 Len=0 MSS=1260 WS=0 74 29.446128 10.10.10.209 10.10.10.250 TCP 50170 > telnet [ACK] Seq=1 Ack=1 Win=66560 Len=0 75 29.452021 10.10.10.209 10.10.10.250 TELNET Telnet Data ... 76 29.483537 10.10.10.250 10.10.10.209 TELNET Telnet Data ... 77 29.483651 10.10.10.209 10.10.10.250 TELNET Telnet Data ... 78 29.523463 10.10.10.250 10.10.10.209 TCP telnet > 50170 [ACK] Seq=4 Ack=5 Win=32768 Len=0 79 29.554954 10.10.10.209 10.10.10.250 TCP 50169 > 23120 [ACK] Seq=720 Ack=505 Win=66048 Len=0 Non-working PC No. Time Source Destination Protocol Info 1 0.000000 10.10.10.209 10.10.10.250 TCP 64161 > 23120 [SYN] Seq=0 Win=8192 Len=0 MSS=1260 WS=8 2 2.999847 10.10.10.209 10.10.10.250 TCP 64161 > 23120 [SYN] Seq=0 Win=8192 Len=0 MSS=1260 WS=8 3 4.540773 Cisco_3c:78:00 Cisco-Li_55:87:72 ARP Who has 10.0.0.1? Tell 10.10.10.209 4 4.540843 Cisco-Li_55:87:72 Cisco_3c:78:00 ARP 10.0.0.1 is at 00:1a:70:55:87:72 5 8.992284 10.10.10.209 10.10.10.250 TCP 64161 > 23120 [SYN] Seq=0 Win=8192 Len=0 MSS=1260

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• function not working in production mode

  - by maps

  I am using the rvideo gem to transcode files to a .flv format. class Video < ActiveRecord::Base include AASM aasm_column :status aasm_initial_state :initial aasm_state :initial aasm_state :converting, :exit => :transcode aasm_state :transfering , :exit => :send_s3 aasm_state :completed aasm_state :failed aasm_event :convert do transitions :from => [:initial], :to => :converting end aasm_event :transfer do transitions :from => [:converting], :to => :transfering end aasm_event :complete do transitions :from => [:transfering], :to => :completed end aasm_event :error do transitions :from => [:initial, :converting, :transfering, :completed] end has_attached_file :asset, :path => "uploads/:attachment/:id.:basename.:extension" def flash_path return self.asset.path + '.flv' end def flash_name return File::basename(self.asset.path)# + '.flv' end def flash_url return "#{AWS_HOST}/#{AWS_BUCKET}/#{self.flash_name}" end # transcode file def transcode begin RVideo::Transcoder.logger = logger file = RVideo::Inspector.new(:file => self.asset.path) command = "ffmpeg -i $input_file$ -y -s $resolution$ -ar 44100 -b 64k -r 15 -sameq $output_file$" options = { :input_file => "#{RAILS_ROOT}/#{self.asset.path}", :output_file => "#{RAILS_ROOT}/#{self.flash_path}", :resolution => "320x200" } transcoder = RVideo::Transcoder.new transcoder.execute(command, options) rescue RVideo::TranscoderError => e logger.error "Encountered error transcoding #{self.asset.path}" logger.error e.message end end The input file is added to the asset directory, but I never get an outputted file. On the view page aasm hangs on "converting".

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• response.write only working IE for ASP.NET

  - by slowlycooked

  I'm using uploadify (http://www.uploadify.com/) to upload video to my site then convert them into *.flv using ffmpeg and play preview. But it dosen't fully working with firefox, chrome or safari. uploadify provides a onComplete interface, so when the script (.ashx, .php) used on your site for saving uploaded files. you can use response.write("blabla") or (echo "blabla") to invoke the javascript function that registed as OnComplete. i have test with few video files like avi, mpg, mp4, they are less then 50mb,and they all worked with all 4 browsers. However, when i was trying to upload a 75mb mp4 file, it worked in IE, but didn't working in other three. I can see the .flv file has been create in the upload folder, i can see debug messsage output after response.write("blabla"), but the javascript function was not invoked. i.e. the preview didn't play. anyone knows why? is there a timeout or something on response.write so after a period of time it wont work? e.g. 75mb file took longer time to convert than other smaller size file i tried. thansk

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• PHP : How to add a clickable logo on below right corner on a flash video player?

  - by kishore

  Hi all, I have a code as shown below var so = new SWFObject('http://montsmile.com/jwplayer/player.swf','mpl','480','380','8'); so.addParam('allowscriptaccess','always'); so.addParam('allowfullscreen','true'); so.addVariable('height','310'); so.addVariable('width','470'); so.addVariable('file','http://localhost:81/newtip/'); so.addVariable('logo','http://localhost:81/newtip/ffmpeg/logo2.jpg'); //so.addVariable("logoPosition", "bottom_left"); so.addVariable("ClickURL", "http://www.google.com"); so.addVariable('captions','/upload/corrie.xml'); so.addVariable('link',''); so.addVariable('linkfromdisplay','true'); so.addVariable('linktarget','_blank'); so.addVariable('searchbar','false'); so.addVariable('skin','http://montsmile.com/jwplayer/player.swf'); so.write('player'); It is displaying the logo on the player. but when i click the logo it does not going to the target page. I want to make the logo as clickable. Please help me on this Thanks in advance...

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• Problem with unlink() in php!

  - by Holicreature

  I'm creating a temp image always named 1.png under specific folder and once i read the image_contents and process, i use unlink() to delete that specific image from that folder. But sometimes the image file is not deleted and the same image is file is read and processed. That script is working otherwise fine... There is no permission related issues , as the files are deleted sometimes... Will there be any issue when the script is repeatedly called and the image with the name is already present and not deleted etc.. ??? Please suggest me what would be the problem extension_loaded('ffmpeg'); $max_width = 120; $max_height = 72; $path ="/home/fff99/public_html/temp/"; ..... ..... $nname = "/home/friend99/public_html/temp/".$imgname; $fileo = fopen($nname,"rb"); if($fileo) { $imgData = addslashes(file_get_contents($nname)); .... ... .. } unlink('$nname');

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• Node.js Adventure - Node.js on Windows

  - by Shaun

  Two weeks ago I had had a talk with Wang Tao, a C# MVP in China who is currently running his startup company and product named worktile. He asked me to figure out a synchronization solution which helps his product in the future. And he preferred me implementing the service in Node.js, since his worktile is written in Node.js. Even though I have some experience in ASP.NET MVC, HTML, CSS and JavaScript, I don’t think I’m an expert of JavaScript. In fact I’m very new to it. So it scared me a bit when he asked me to use Node.js. But after about one week investigate I have to say Node.js is very easy to learn, use and deploy, even if you have very limited JavaScript skill. And I think I became love Node.js. Hence I decided to have a series named “Node.js Adventure”, where I will demonstrate my story of learning and using Node.js in Windows and Windows Azure. And this is the first one.   (Brief) Introduction of Node.js I don’t want to have a fully detailed introduction of Node.js. There are many resource on the internet we can find. But the best one is its homepage. Node.js was created by Ryan Dahl, sponsored by Joyent. It’s consist of about 80% C/C++ for core and 20% JavaScript for API. It utilizes CommonJS as the module system which we will explain later. The official definition of Node.js is Node.js is a platform built on Chrome's JavaScript runtime for easily building fast, scalable network applications. Node.js uses an event-driven, non-blocking I/O model that makes it lightweight and efficient, perfect for data-intensive real-time applications that run across distributed devices. First of all, Node.js utilizes JavaScript as its development language and runs on top of V8 engine, which is being used by Chrome. It brings JavaScript, a client-side language into the backend service world. So many people said, even though not that actually, “Node.js is a server side JavaScript”. Additionally, Node.js uses an event-driven, non-blocking IO model. This means in Node.js there’s no way to block currently working thread. Every operation in Node.js executed asynchronously. This is a huge benefit especially if our code needs IO operations such as reading disks, connect to database, consuming web service, etc.. Unlike IIS or Apache, Node.js doesn’t utilize the multi-thread model. In Node.js there’s only one working thread serves all users requests and resources response, as the ST star in the figure below. And there is a POSIX async threads pool in Node.js which contains many async threads (AT stars) for IO operations. When a user have an IO request, the ST serves it but it will not do the IO operation. Instead the ST will go to the POSIX async threads pool to pick up an AT, pass this operation to it, and then back to serve any other requests. The AT will actually do the IO operation asynchronously. Assuming before the AT complete the IO operation there is another user comes. The ST will serve this new user request, pick up another AT from the POSIX and then back. If the previous AT finished the IO operation it will take the result back and wait for the ST to serve. ST will take the response and return the AT to POSIX, and then response to the user. And if the second AT finished its job, the ST will response back to the second user in the same way. As you can see, in Node.js there’s only one thread serve clients’ requests and POSIX results. This thread looping between the users and POSIX and pass the data back and forth. The async jobs will be handled by POSIX. This is the event-driven non-blocking IO model. The performance of is model is much better than the multi-threaded blocking model. For example, Apache is built in multi-threaded blocking model while Nginx is in event-driven non-blocking mode. Below is the performance comparison between them. And below is the memory usage comparison between them. These charts are captured from the video NodeJS Basics: An Introductory Training, which presented at Cloud Foundry Developer Advocate.   Node.js on Windows To execute Node.js application on windows is very simple. First of you we need to download the latest Node.js platform from its website. After installed, it will register its folder into system path variant so that we can execute Node.js at anywhere. To confirm the Node.js installation, just open up a command windows and type “node”, then it will show the Node.js console. As you can see this is a JavaScript interactive console. We can type some simple JavaScript code and command here. To run a Node.js JavaScript application, just specify the source code file name as the argument of the “node” command. For example, let’s create a Node.js source code file named “helloworld.js”. Then copy a sample code from Node.js website. 1: var http = require("http"); 2:  3: http.createServer(function (req, res) { 4: res.writeHead(200, {"Content-Type": "text/plain"}); 5: res.end("Hello World\n"); 6: }).listen(1337, "127.0.0.1"); 7:  8: console.log("Server running at http://127.0.0.1:1337/"); This code will create a web server, listening on 1337 port and return “Hello World” when any requests come. Run it in the command windows. Then open a browser and navigate to http://localhost:1337/. As you can see, when using Node.js we are not creating a web application. In fact we are likely creating a web server. We need to deal with request, response and the related headers, status code, etc.. And this is one of the benefit of using Node.js, lightweight and straightforward. But creating a website from scratch again and again is not acceptable. The good news is that, Node.js utilizes CommonJS as its module system, so that we can leverage some modules to simplify our job. And furthermore, there are about ten thousand of modules available n the internet, which covers almost all areas in server side application development.   NPM and Node.js Modules Node.js utilizes CommonJS as its module system. A module is a set of JavaScript files. In Node.js if we have an entry file named “index.js”, then all modules it needs will be located at the “node_modules” folder. And in the “index.js” we can import modules by specifying the module name. For example, in the code we’ve just created, we imported a module named “http”, which is a build-in module installed alone with Node.js. So that we can use the code in this “http” module. Besides the build-in modules there are many modules available at the NPM website. Thousands of developers are contributing and downloading modules at this website. Hence this is another benefit of using Node.js. There are many modules we can use, and the numbers of modules increased very fast, and also we can publish our modules to the community. When I wrote this post, there are totally 14,608 modules at NPN and about 10 thousand downloads per day. Install a module is very simple. Let’s back to our command windows and input the command “npm install express”. This command will install a module named “express”, which is a MVC framework on top of Node.js. And let’s create another JavaScript file named “helloweb.js” and copy the code below in it. I imported the “express” module. And then when the user browse the home page it will response a text. If the incoming URL matches “/Echo/:value” which the “value” is what the user specified, it will pass it back with the current date time in JSON format. And finally my website was listening at 12345 port. 1: var express = require("express"); 2: var app = express(); 3:  4: app.get("/", function(req, res) { 5: res.send("Hello Node.js and Express."); 6: }); 7:  8: app.get("/Echo/:value", function(req, res) { 9: var value = req.params.value; 10: res.json({ 11: "Value" : value, 12: "Time" : new Date() 13: }); 14: }); 15:  16: console.log("Web application opened."); 17: app.listen(12345); For more information and API about the “express”, please have a look here. Start our application from the command window by command “node helloweb.js”, and then navigate to the home page we can see the response in the browser. And if we go to, for example http://localhost:12345/Echo/Hello Shaun, we can see the JSON result. The “express” module is very populate in NPM. It makes the job simple when we need to build a MVC website. There are many modules very useful in NPM. - underscore: A utility module covers many common functionalities such as for each, map, reduce, select, etc.. - request: A very simple HTT request client. - async: Library for coordinate async operations. - wind: Library which enable us to control flow with plain JavaScript for asynchronous programming (and more) without additional pre-compiling steps.   Node.js and IIS I demonstrated how to run the Node.js application from console. Since we are in Windows another common requirement would be, “can I host Node.js in IIS?” The answer is “Yes”. Tomasz Janczuk created a project IISNode at his GitHub space we can find here. And Scott Hanselman had published a blog post introduced about it.   Summary In this post I provided a very brief introduction of Node.js, includes it official definition, architecture and how it implement the event-driven non-blocking model. And then I described how to install and run a Node.js application on windows console. I also described the Node.js module system and NPM command. At the end I referred some links about IISNode, an IIS extension that allows Node.js application runs on IIS. Node.js became a very popular server side application platform especially in this year. By leveraging its non-blocking IO model and async feature it’s very useful for us to build a highly scalable, asynchronously service. I think Node.js will be used widely in the cloud application development in the near future.   In the next post I will explain how to use SQL Server from Node.js.   Hope this helps, Shaun All documents and related graphics, codes are provided "AS IS" without warranty of any kind. Copyright © Shaun Ziyan Xu. This work is licensed under the Creative Commons License.

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• Use an audio/video file from a Linux laptop via USB to be played by Magic Sing ET-23H

  - by AisIceEyes

  I am one of the technical directors of a regular karaoke contest event. For the karaoke contest itself, due to tight budget, we are using what one of the sponsors are providing - Magic Sing ET-23H . The video output of the Magic Sing ET-23H are broadcasted at two big screens that are being shown to the audience and event attendees. When a karaoke contestant provides his / her karaoke video, the video itself is in a readable USB flashdrive and is attached to the USB input of Magic Sing ET-23H. What really bugs me is that the interface of Magic Sing ET-23H are also being broadcasted at the big screen video feeds. The interface of choosing the video file is being seen in the Magic Sing ET-23H - also to the big video screens that are seen by the audience and event goers. I will post in the comments ( if my less than 10 reputation would allow me) the picture of Magic Sing ET-23KH USB input of the device. I always bring my laptop, Acer AS5742-7653, during the regular karaoke event. I'm using my laptop also for tallying of scores from the judges, and also playing audio files from contestants that did not provide a karaoke video. I personally am using different Linux distros, but I next to all the time use my Ubuntu Studio 12.04.3 64bit partition during the regular karaoke contest event. My question is this: Is there a way I can share a temporary video/audio file directly from the laptop I'm using, going to the Magic Sing ET-23H that can broadcast both the video/audio file? Just like how in Window's Avisynth AVS files, or VirtualDub's temporary avi file, or like using ffplay (of ffmpeg), etc. I have researched somewhat the matter and found links in SuperUser.com. Though I can only provide the links at the comments section of this post if my reputation of less than 10 would allow me. I have a hunch it is possible, but I have not fully understood the device being used at the event, Magic Sing ET-23H, if there are other ways for it to broadcast video and audio files besides its USB input. Any help to my current predicament is highly appreciated. Thank you. PS: Since I need at least 10 reputation to post more than 2 links and also post images, I will try to post the image & links at the comments (if my below 10 reputation would allow me).

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• Updates broke my themes/shell [Ubuntu 12.04 running Gnome 3 ]

  - by APNW

  I am running gnome-session 3.4.2.1. After the latest updates (listed below) my theme regressed to what looks like tango - not sure. Am unable to change it using Gnome-tweak tool or the display settings. I am also unable to change the wallpaper. Here's what it looks like: Synaptic: Chromium and this is the wallpaper page even though I have selected the wallpaper, it actually does not change. This same problem occurred on my personal computer, and one other computer I have, all running the same software/config. The interesting thing is that while Gnome 3 and Unity are affected, Cinnamon is not. What I've done so far: purged and re-installed both gnome 3 and Unity- no change noted. So, how do I fix this? Thanks Here's the installation log: Start-Date: 2013-11-07 12:01:28 Upgrade: chromium-browser-l10n:i386 (28.0.1500.71-0ubuntu1.12.04.1, 30.0.1599.114-0ubuntu0.12.04.3), libswscale2:i386 (0.8.6-0ubuntu0.12.04.1, 0.8.8-0ubuntu0.12.04.1), chromium-codecs-ffmpeg:i386 (28.0.1500.71-0ubuntu1.12.04.1, 30.0.1599.114-0ubuntu0.12.04.3), chromium-browser:i386 (28.0.1500.71-0ubuntu1.12.04.1, 30.0.1599.114-0ubuntu0.12.04.3), libpostproc52:i386 (0.8.6-0ubuntu0.12.04.1, 0.8.8-0ubuntu0.12.04.1), libavcodec-extra-53:i386 (0.8.6ubuntu0.12.04.1, 0.8.8ubuntu0.12.04.1), libavformat53:i386 (0.8.6-0ubuntu0.12.04.1, 0.8.8-0ubuntu0.12.04.1), libavutil-extra-51:i386 (0.8.6ubuntu0.12.04.1, 0.8.8ubuntu0.12.04.1) End-Date: 2013-11-07 12:02:00 Start-Date: 2013-11-07 17:32:55 Commandline: aptdaemon role='role-commit-packages' sender=':1.136' Install: libmusicbrainz5-0:i386 (5.0.1-2~precise2), udisks2:i386 (1.98.0-1~precise1), libclutter-gst-1.0-0:i386 (1.5.4-0ubuntu2), libudisks2-0:i386 (1.98.0-1~precise1), cinnamon-session-common:i386 (2.0.4-20131105043005-precise), librhythmbox-core6:i386 (2.97-1ubuntu1~precise1), gcr:i386 (3.4.1-3~precise1), libcluttergesture-0.0.2-0:i386 (0.0.2.1-2ubuntu3), libmx-1.0-2:i386 (1.4.3-0ubuntu1), guile-2.0-libs:i386 (2.0.5+1-1), libclutter-imcontext-0.1-0:i386 (0.1.4-2build1), libnatpmp1:i386 (20110808-3ubuntu1) Upgrade: gnome-keyring:i386 (3.2.2-2ubuntu4.1, 3.4.1-4ubuntu1~precise1), cinnamon:i386 (2.0.6-20131026040307-precise, 2.0.10-20131105040309-precise), gir1.2-muffin-3.0:i386 (2.0.3-20131023003029-precise, 2.0.3-20131105003012-precise), gir1.2-totem-1.0:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), nemo:i386 (2.0.2-20131023010018-precise, 2.0.5-20131105010007-precise), aisleriot:i386 (3.2.3.2-0ubuntu1, 3.4.1-1~precise1), procps:i386 (3.2.8-11ubuntu6.2, 3.2.8-11ubuntu6.3), libcinnamon-desktop0:i386 (2.0.2-20131025011504-precise, 2.0.3-20131105011505-precise), libgck-1-0:i386 (3.2.2-2ubuntu4.1, 3.4.1-3~precise1), totem-plugins:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), cinnamon-desktop-data:i386 (2.0.2-20131025011504-precise, 2.0.3-20131105011505-precise), rhythmbox:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), libgcr-3-1:i386 (3.2.2-2ubuntu4.1, 3.4.1-3~precise1), seahorse:i386 (3.2.2-0ubuntu2.1, 3.4.1-2~precise1), muffin-common:i386 (2.0.3-20131023003029-precise, 2.0.3-20131105003012-precise), totem-common:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), libtotem0:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), rhythmbox-data:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), gir1.2-cinnamondesktop-3.0:i386 (2.0.2-20131025011504-precise, 2.0.3-20131105011505-precise), cinnamon-session:i386 (2.0.1-20131021043004-precise, 2.0.4-20131105043005-precise), rhythmbox-mozilla:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), rhythmbox-plugin-zeitgeist:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), libmuffin0:i386 (2.0.3-20131023003029-precise, 2.0.3-20131105003012-precise), cjs:i386 (2.0.0-20131021020602-precise, 2.0.0-20131105020703-precise), rhythmbox-plugin-cdrecorder:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), cinnamon-common:i386 (2.0.6-20131026040307-precise, 2.0.10-20131105040309-precise), gnome-disk-utility:i386 (3.0.2-2ubuntu7, 3.4.1-0ubuntu1~precise1), nemo-fileroller:i386 (2.0.0-20131021020004-precise, 2.0.0-20131105020003-precise), libnemo-extension1:i386 (2.0.2-20131023010018-precise, 2.0.5-20131105010007-precise), rhythmbox-plugins:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), gimp:i386 (2.8.6-0precise1~ppa, 2.8.8-0precise0~ppa), cinnamon-settings-daemon:i386 (2.0.5-20131026004504-precise, 2.0.6-20131105004505-precise), libgimp2.0:i386 (2.8.6-0precise1~ppa, 2.8.8-0precise0~ppa), gir1.2-rb-3.0:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), wpasupplicant:i386 (0.7.3-6ubuntu2.1, 0.7.3-6ubuntu2.2), libcjs0c:i386 (2.0.0-20131021020602-precise, 2.0.0-20131105020703-precise), nemo-data:i386 (2.0.2-20131023010018-precise, 2.0.5-20131105010007-precise), totem:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), gimp-data:i386 (2.8.6-0precise1~ppa, 2.8.8-0precise0~ppa), transmission-common:i386 (2.51-0ubuntu1.3, 2.73-0ubuntu1~precise1), cinnamon-translations:i386 (2.0.1-20131021040407-precise, 2.0.1-20131105040807-precise), totem-mozilla:i386 (3.0.1-0ubuntu21.1, 3.4.3-0ubuntu1~precise1), rhythmbox-plugin-magnatune:i386 (2.96-0ubuntu4.3, 2.97-1ubuntu1~precise1), transmission-gtk:i386 (2.51-0ubuntu1.3, 2.73-0ubuntu1~precise1) End-Date: 2013-11-07 17:34:40

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• H.264 over RTP - Identify SPS and PPS Frames

  - by Toby

  I have a raw H.264 Stream from an IP Camera packed in RTP frames. I want to get raw H.264 data into a file so I can convert it with ffmpeg. So when I want to write the data into my raw H.264 file I found out it has to look like this: 00 00 01 [SPS] 00 00 01 [PPS] 00 00 01 [NALByte] [PAYLOAD RTP Frame 1] // Payload always without the first 2 Bytes -> NAL [PAYLOAD RTP Frame 2] [... until PAYLOAD Frame with Mark Bit received] // From here its a new Video Frame 00 00 01 [NAL BYTE] [PAYLOAD RTP Frame 1] .... So I get the SPS and the PPS from the Session Description Protocol out of my preceding RTSP communication. Additionally the camera sends the SPS and the PPSin two single messages before starting with the video stream itself. So I capture the messages in this order: 1. Preceding RTSP Communication here ( including SDP with SPS and PPS ) 2. RTP Frame with Payload: 67 42 80 28 DA 01 40 16 C4 // This is the SPS 3. RTP Frame with Payload: 68 CE 3C 80 // This is the PPS 4. RTP Frame with Payload: ... // Video Data Then there come some Frames with Payload and at some point a RTP Frame with the Marker Bit = 1. This means ( if I got it right) that I have a complete video frame. Afer this I write the Prefix Sequence ( 00 00 01 ) and the NALfrom the payload again and go on with the same procedure. Now my camera sends me after every 8 complete Video Frames the SPS and the PPS again. ( Again in two RTP Frames, as seen in the example above ). I know that especially the PPS can change in between streaming but that's not the problem. My questions are now: 1. Do I need to write the SPS/PPS every 8th Video Frame? If my SPS and my PPS don't change it should be enough to have them written at the very beginning of my file and nothing more? 2. How to distinguish between SPS/PPS and normal RTP Frames? In my C++ Code which parses the transmitted data I need make a difference between the RTP Frames with normal Payload an the ones carrying the SPS/PPS. How can I distinguish them? Okay the SPS/PPS frames are usually way smaller, but that's not a save call to rely on. Because if I ignore them I need to know which data I can throw away, or if I need to write them I need to put the 00 00 01 Prefix in front of them. ? Or is it a fixed rule that they occur every 8th Video Frame?

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• How to configure the framesize using AudioUnit.framework on iOS

  - by Piperoman

  I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

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