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• How do I play back a WAV in ActionScript?

  - by Jeremy White

  Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

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• What is the best way to send structs containing enum values via sockets in C.

  - by Axel

  I've lots of different structs containing enum members that I have to transmit via TCP/IP. While the communication endpoints are on different operating systems (Windows XP and Linux) meaning different compilers (gcc 4.x.x and MSVC 2008) both program parts share the same header files with type declarations. For performance reasons, the structures should be transmitted directly (see code sample below) without expensively serializing or streaming the members inside. So the question is how to ensure that both compilers use the same internal memory representation for the enumeration members (i.e. both use 32-bit unsigned integers). Or if there is a better way to solve this problem... //type and enum declaration typedef enum { A = 1, B = 2, C = 3 } eParameter; typedef enum { READY = 400, RUNNING = 401, BLOCKED = 402 FINISHED = 403 } eState; #pragma pack(push,1) typedef struct { eParameter mParameter; eState mState; int32_t miSomeValue; uint8_t miAnotherValue; ... } tStateMessage; #pragma pack(pop) //... send via socket tStateMessage msg; send(iSocketFD,(void*)(&msg),sizeof(tStateMessage)); //... receive message on the other side tStateMessage msg_received; recv(iSocketFD,(void*)(&msg_received),sizeof(tStateMessage)); Additionally... Since both endpoints are little endian maschines, endianess is not a problem here. And the pack #pragma solves alignment issues satisfactorily. Thx for your answers, Axel

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• How does GMail implement Comet?

  - by Morgan Cheng

  With the help of HttpWatch, I tried to figure out how GMail implement Comet. I Login in GMail with two account, one in IE and the other in Firefox. Chatting in GTalk in GMail with some magic words like "WASSUP". Then, I logoff both GMail accounts, filter any http content without "WASSUP" string. The result shows which HTTP request is the streaming channel. (Note: I have to logoff. Otherwise, never-ending HTTP would not show content in HttpWatch.) The result is interesting. The URL for stream channel is like: https://mail/channel/bind?VER=8&at=xn3j33vcvk39lkfq..... There is no surprise that GMail do Comet in IE with IFRAME. The Http content starts with " Originally, I guessed that GMail do Comet in Firefox with multipart XmlHttpRequest. To my surprise, the response header doesn't have "multipart/x-mixed-replace" header. The response headers are as below: HTTP/1.1 200 OK Content-Type: text/plain; charset=utf-8 Cache-Control: no-cache, no-store, max-age=0, must-revalidate Pragma: no-cache Expires: Fri, 01 Jan 1990 00:00:00 GMT Date: Sat, 20 Mar 2010 01:52:39 GMT X-Frame-Options: ALLOWALL Transfer-Encoding: chunked X-Content-Type-Options: nosniff Server: GSE X-XSS-Protection: 0 Unfortunately, the HttpWatch doesn't tell whether a HTTP request is from XmlHttpRequest or not. The content is not HTML but JSON. It looks like a response for XHR, but that would not work for Comet without multipart/x-mixed-replace, right? Is there any way else to figure out how GMail implement Comet? Thanks.

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• Problems with MediaPlayer, raw resources, stop and start

  - by arakn0

  Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaPlayer class to reproduce some sounds/music. I am playing raw resources (res/raw) and it looks kind of easy. To play a raw resource, the MediaPlayer has to be initialized like this: MediaPlayer mp = MediaPlayer.create(appContext, R.raw.song); mp.start(); Until here there is no problem. The sound is played, and everything works fine. My problem appears when I want to add more options to my application. Specifically when I add the "Stop" button/option. Basically, what I want to do is...when I press "Stop", the music stops. And when I press "Start", the song/sound starts over. (pretty basic!) To stop the media player, you only have to call stop(). But to play the sound again, the media player has to be reseted and prepared. mp.reset(); mp.setDataSource(params); mp.prepare(); The problem is that the method setDataSource() only accepts as params a file path, Content Provider URI, streaming media URL path, or File Descriptor. So, since this method doesn't accept a resource identifier, I don't know how to set the data source in order to call prepare(). In addition, I don't understand why you can't use a Resouce identifier to set the data source, but you can use a resource identifier when initializing the MediaPlayer. I guess that I'm missing something. I wonder if I am mixing concepts, and the method stop() doesn't have to be called in the "Stop" button. Any help? Thanks in advanced!!!

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• Keymap issues with NX from Mac OS X Lion

  - by Andy

  I tried to answer the question from Mark: Keymap issues with NX from Mac OS X Lion to Ubuntu However, it is locked so I figured I would post a new question / answer. I have been trying to answer this for a few days now because I have no issues when connecting through NX Client (technically OpenNX) to FreeNX server from an iMac (with Lion), but if I try to connect with a Macbook Pro I get horrible keyboard binding issues. The fix that is working for me is to go into: ~/.nx/config/HOST.nxs and change: <option key="Current keyboard" value="false"/> <option key="Custom keyboard layout" value="empty"/> <option key="Grab keyboard" value="false"/> I have tried this on three NX Servers and all are fixed. Hope it helps or gets you closer. Always check in the ~/.nx/temp/ for the sshlog and see if --keyboard="empty/empty" instead of "pc105/en" because the Mac is really pc104. 9:05:35: startsession --session="HOST" --type="unix-gnome" --cache="8M" --images="32M" --link="adsl" --geometry="2556\ x1396" --screeninfo="2560x1440x32+render" --keyboard="empty/empty" --backingstore="1" --encryption="1" --composite="1" --\ shmem="1" --shpix="1" --streaming="1" --samba="0" --cups="0" --nodelay="1" --defer="0" --client="macosx" --media="0" --st\ rict="0" --aux="1"

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• Frame Accurate Browser Launchable Video Player ... ?

  - by cliftonc

  I have a requirement where I need to enable playback (full screen) of a h.264 MPEG4 (thanks for the correction!) video from a local network, launchable from a browser link on a Windows workstation, and be frame accurate. By frame accurate I mean that I need to be able to scrub through the video in the same way you would with a vtr, stop at a frame, and then move backwards and forwards frame by frame (it is for a very specific compliance requirement where have to be able to check every frame if there is something that is potentially against broadcasting guidelines). The application itself is used to capture notes while viewing the material, so the end model is for a dual monitor workstation, with a web form in one, the video playing full screen in the second (no issue launching the video and manually having to move it to the second screen), and then the user controls the video via keyboard shortcuts or a jog shuttle. I have looked at QT, but the java bindings seem to be dead or nearly so, flash isn't frame accurate, VLC given its streaming heritage seems to be only able to move forward by a frame and not backwards, and all I have left are commercial offerings that in my experience are difficult and expensive to change. Any ideas of where I should look or alternative options? Any advice appreciated!

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• Executing bat file and returning the prompt

  - by Lieven Cardoen

  I have a problem with cruisecontrol where an ant scripts executes a bat file that doesn't give me the prompt back. As a result, the project in cruisecontrol keeps on bulding forever until I restart cruisecontrol. How can I resolve this? It's a startup.bat from wowza (Streaming Server) that I'm executing: @echo off call setenv.bat if not %WMSENVOK% == "true" goto end set _WINDOWNAME="Wowza Media Server 2" set _EXESERVER= if "%1"=="newwindow" ( set _EXESERVER=start %_WINDOWNAME% shift ) set CLASSPATH="%WMSAPP_HOME%\bin\wms-bootstrap.jar" rem cacls jmxremote.password /P username:R rem cacls jmxremote.access /P username:R rem NOTE: Here you can configure the JVM's built in JMX interface. rem See the "Server Management Console and Monitoring" chapter rem of the "User's Guide" for more information on how to configure the rem remote JMX interface in the [install-dir]/conf/Server.xml file. set JMXOPTIONS=-Dcom.sun.management.jmxremote=true rem set JMXOPTIONS=%JMXOPTIONS% -Djava.rmi.server.hostname=192.168.1.7 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.port=1099 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.authenticate=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.ssl=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.password.file= "%WMSCONFIG_HOME%/conf/jmxremote.password" rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.access.file= "%WMSCONFIG_HOME%/conf/jmxremote.access" rem log interceptor com.wowza.wms.logging.LogNotify - see Javadocs for ILogNotify %_EXESERVER% "%_EXECJAVA%" %JAVA_OPTS% %JMXOPTIONS% -Dcom.wowza.wms.AppHome="%WMSAPP_HOME%" -Dcom.wowza.wms.ConfigURL="%WMSCONFIG_URL%" -Dcom.wowza.wms.ConfigHome="%WMSCONFIG_HOME%" -cp %CLASSPATH% com.wowza.wms.bootstrap.Bootstrap start :end

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• Webrick transparent proxy

  - by zzeroo

  Hi there, I've a absolute simple proxy running. require 'webrick' require 'webrick/httpproxy' s = WEBrick::HTTPProxyServer.new(:Port => 8080, :RequestCallback => Proc.new{|req,res| puts req.request_line, req.raw_header}) # Shutdown functionality trap("INT"){s.shutdown} # run the beast s.start This should in my mind not influence the communication in any way. But some sites doesn't work any more. Specially http://lastfm.de 's embedded flash players doesn't work. The header looks link: - -> http://ext.last.fm/2.0/?api%5Fsig=aa3e9ac9edf46ceb9a673cb76e61fef4&flashresponse=true&y=1269686332&streaming=true&playlistURL=lastfm%3A%2F%2Fplaylist%2Ftrack%2F42620245&fod=true&sk=ee93ae4f438767bf0183d26478610732&lang=de&api%5Fkey=da6ae1e99462ee22e81ac91ed39b43a4&method=playlist%2Efetch GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1 Host: play.last.fm User-Agent: Mozilla/5.0 (X11; U; Linux i686; de; rv:1.9.2) Gecko/20100308 Ubuntu/10.04 (lucid) Firefox/3.6 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: de,en-us;q=0.7,en;q=0.3 Accept-Encoding: gzip,deflate Accept-Charset: ISO-8859-1,utf-8;q=0.7,*;q=0.7 Keep-Alive: 115 Proxy-Connection: keep-alive Cookie: AnonWSSession=ee93ae4f438767bf0183d26478610732; AnonSession=cb8096e3b0d8ec9f4ffd6497a6d052d9-12bb36d49132e492bb309324d8a4100fc422b3be9c3add15ee90eae3190db5fc localhost - - [27/Mar/2010:11:38:52 CET] "GET http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout HTTP/1.1" 404 7593 - -> http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout localhost - - [27/Mar/2010:11:38:52 CET] "GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1" 302 0 I nead some hints why or what the communication disturb.

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• Internet Explorer Warning when embedding Youtube on HTTPS site?

  - by pellepim

  Our application is run over HTTPS which rarely presents any problems for us. When it comes to youtube however, the fact that they do not present any content over SSL connections is giving us some head ache when trying to embed clips. Mostly because of Internet Explorers famous little warning message: "Do you want to view only the webpage content that was delivered securely? This page contains content that will not be delivered using a secure HTTPS ... etc" I've tried to solve this in several ways. The most promising one was to use the ProxyPass functionality in Apache to map to YouTube. Like this: ProxyPass: /youtube/ http://www.youtube.com ProxyPassReverse: /youtube/ http://www.youtube.com This gets rid of the annoying warning. However, the youtube SWF fails to start streaming The SWF i manage to load into the browser simply states : "An error occurred, please try again later". Potential solutions are perhaps: Download youtube FLV:s and serve them out of own domain (gah) Use custom FLV-player and stream only FLV:s from youtube over a https proxy? Update 10 March: I've tried to use Googles Youtube API for ActionScript to load a player. It looked promising at first and I was able to load a player through my https:// proxy. However, the SWF that is loaded contains loads of explicit calls to different non-ssl urls to create authentication links for the FLV-stream and for loading different crossdomain policies. It really seems like we're not supposed to access flv-streams directly. This makes it very hard to bypass the Internet Explorer warning, short of ripping out the FLV:s from youtube and serving them out of your own domain. There are solutions out there for downloading youtubes FLV:s. But that is not compliant with the Youtube terms of use and is really not an option for us.

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• Should I move big data blobs in JSON or in separate binary connection?

  - by Amagrammer

  QUESTION: Is it better to send large data blobs in JSON for simplicity, or send them as binary data over a separate connection? If the former, can you offer tips on how to optimize the JSON to minimize size? If the latter, is it worth it to logically connect the JSON data to the binary data using an identifier that appears in both, e.g., as "data" : "< unique identifier " in the JSON and with the first bytes of the data blob being < unique identifier ? CONTEXT: My iPhone application needs to receive JSON data over the 3G network. This means that I need to think seriously about efficiency of data transfer, as well as the load on the CPU. Most of the data transfers will be relatively small packets of text data for which JSON is a natural format and for which there is no point in worrying much about efficiency. However, some of the most critical transfers will be big blobs of binary data -- definitely at least 100 kilobytes of data, and possibly closer to 1 megabyte as customers accumulate a longer history with the product. (Note: I will be caching what I can on the iPhone itself, but the data still has to be transferred at least once.) It is NOT streaming data. I will probably use a third-party JSON SDK -- the one I am using during development is here. Thanks

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• Playing a .TS file on iOS

  - by Jonathan Grynspan

  We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

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• Internet Radio Station for University

  - by ryan

  I am trying to help my University Student Radio station rethink the setup of the way they stream music, but I have some questions regarding the use of Ubuntu to stream music. Currently, the radio station uses two windows machines: one of which is used to stream the radio station and serve the website, and the other is used by rotating djs to select songs and create playlists. The computer used by djs feeds mono into the sound card of the server and the server streams the feed online. -Ideally I would like to maintain a two-computer setup: One computer as server, and another that is used to select and play music by rotating djs. -I would like to use Ubuntu for the server. -I would like to use Windows for the other machine. -The server should be able to stream song information. First, is there a way to somehow get the song information from an analog feed? Second, what is the best streaming server for radio? I have encountered shoutcast, icecast, and darwin, but I don't know where to begin in attempting to gauge them. Finally, if anyone has any tips or pointers about small internet radio station management/ setup they would be appreciated as this is my first radio station, and I am eager to hear of past experiences.

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• Rate limiting a ruby file stream

  - by Matthew Savage

  I am working on a project which involves uploading flash video files to a S3 bucket from a number of geographically distributed nodes. The video files are about 2-3mb each, and we are only sending one file (per node) every ten minutes, however the bandwidth we consume needs to be rate limited to ~20k/s, as these nodes are delivering streaming media to a CDN, and due to the locations we are only able to get 512k max upload. I have been looking into the ASW-S3 gem and while it doesn't offer any kind of rate limiting I am aware that you can pass in a IO Stream. Given this I am wondering if it might be possible to create a rate-limited stream which overrides the read method, adds in the rate limiting logic (e.g. in its simplest form a call to sleep between reads) and then call out to the super of the overridden method. Another option I considered is hacking the code for Net::HTTP and putting the rate limiting into the send_request_with_body_stream method which is using a while loop, but I'm not entirely sure which would be the best option. I have attempted at extending the IO class, however that didn't work at all, simply inheriting from the class with class ThrottledIO < IO didn't do anything. Any suggestions will be greatly appreciated.

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• Elegant Algorithm for Parsing Data Stream Into Record

  - by Matt Long

  I am interfacing with a hardware device that streams data to my app over Wifi. The data is streaming in just fine. The data contains a character header (DATA:) that indicates a new record has begun. The issues is that the data I receive doesn't necessarily fall on the header boundary, so I have to capture the data until what I've captured contains the header. Then, everything that precedes the header goes into the previous record and everything that comes after it goes into a new record. I have this working, but wondered if anyone has done this before and has a good computer-sciencey way to solve the problem. Here's what I do: Convert the NSData of the current read to an NSString Append the NSString to a placeholder string Check placeholder string for the header (DATA:). If the header is not there, just wait for the next read. If the header exists, append whatever precedes it to a previous record placeholder and hand that placeholder off to an array as a complete record that I can further parse into fields. Take whatever shows up after the header and place it in the record placeholder so that it can be appended to in the next read. Repeat steps 3 - 5. Let me know if you see any flaws with this or have a suggestion for a better way. Seems there should be some design pattern for this, but I can't think of one. Thanks.

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• 4.0/WCF: Best approach for bi-idirectional message bus?

  - by TomTom

  Just a technology update, now that .NET 4.0 is out. I write an application that communicates to the server through what is basically a message bus (instead of method calls). This is based on the internal architecture of the application (which is multi threaded, passing the messages around). There are a limited number of messages to go from the client to the server, quite a lot more from the server to the client. Most of those can be handled via a separate specialized mechanism, but at the end we talk of possibly 10-100 small messages per second going from the server to the client. The client is supposed to operate under "internet conditions". THis means possibly home end users behind standard NAT devices (i.e. typical DSL routers) - a firewalled secure and thus "open" network can not be assumed. I want to have as little latency and as little overhad for the communication as possible. What is the technologally best way to handle the message bus callback? I Have no problem regularly calling to the server for message delivery if something needs to be sent... ...but what are my options to handle the messagtes from the server to the client? WsDualHttp does work how? Especially under a NAT scenario? Just as a note: polling is most likely out - the main problem here is that I would have a significant overhead OR a significant delay, both aren ot really wanted. Technically I would love some sort of streaming appraoch, where the server can write messags to a stream while he generates them and they get sent to the client as they come. Not esure this is doable with WCF, though (if not, I may acutally decide to handle the whole message part outside of WCF and just do control / login / setup / destruction via WCF).

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• Database design for a media server containing movies, music, tv and everything in between?

  - by user364114

  In the near future I am attempting to design a media server as a personal project. MY first consideration to get the project underway is architecture, it will certainly be web based but more specifically than that I am looking for suggestions on the database design. So far I am considering something like the following, where I am using [] to represent a table, the first text is the table name to give an idea of purpose and the items within {} would be fields of the table. Also not, fid is functional id referencing some other table. [Item {id, value/name, description, link, type}] - this could be any entity, single song or whole music album, game, movie - almost see this as a recursive relation, ie. a song is an item but an album that song is part of is also an item or for example a tv season is an item, with multiple items being tv episodes [Type {id, fid, mime type, etc}] - file type specific information - could identify how code handles streaming/sending this item to a user [Location {id, fid, path to file?}] [Users {id, username, email, password, ...? }] - user account information [UAC {id, fid, acess level}] - i almost feel its more flexible to seperate access control permissions form the user accounts themselves [ItemLog {id, fid, fid2, timestamp}] - fid for user id, and fid2 for item id - this way we know what user access what when [UserLog {id, fid, timestamp}] -both are logs for access, whether login or last item access [Quota {id, fid, cap}] - some sort of way to throttle users from queing up the entire site and letting it download ... Suggestions or comments are welcome as the hope is that this project will be a open source project once some code is laid out.

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• Optimizing quality for available bandwidth in Flash/RTMFP

  - by Artem M.

  I'm developing a simple one-on-one P2P video chat using ActionScript, and I'd like to ensure the best video quality for the peers given their bandwidth. This means: Setting the best quality given the available bandwidth when the chat starts Responding to network congestions during chat by decreasing the quality. The task is similar to dynamic stream switching, but P2P has its specifics that make dynamic streaming approaches not work. For example, the maxBytesPerSecond metric monitored in dynamic stream switching is pretty useless in P2P where the receiving NetStream's buffer size is set to 0 to minimize latency. So far, it looks like the most reliable QoS metric for P2P is SRTT. In my simulated tests on a local network, a bandwidth congestion makes it shot up to 500 ms and more when there's a bandwidth limit introduced. However, it gives no hint as to how best adjust the value for bandwidth in Camera.setQuality(0, bandwidth) to respond to the congestion. I've done lots of experiments, and I still don't see a clear and simple solution to the problem. I'm also wondering how this issue is addressed (if at all) in other RTMFP chat solutions.

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• New TabItem Content ActualHeight crashes Xaml Window

  - by Jack Navarro

  I am able to create new TabItems with Content dynamically to a new window by streaming the Xaml with XamlReader: NewWindow newWindow = new NewWindow(); newWindow.Show(); TabControl myTabCntrol = newWindow.FindName("GBtabControl") as TabControl; StringReader stringReader = new StringReader(XamlGrid); XmlReader xmlReader = XmlReader.Create(stringReader); TabItem myTabItem = new TabItem(); myTabItem.Header = qDealName; myTabItem.Content = (UIElement)XamlReader.Load(xmlReader); myTabCntrol.Items.Add(myTabItem); This works fine. It displays a new grid wrapped in a scrollviewer. The problem is access the TabItem content from the newWindow. TabItem ti = GBtabControl.SelectedItem as TabItem; string scrollvwnm = "scrollViewer" + ti.Header.ToString(); MessageBox.Show(ti.ActualHeight.ToString()); // returns 21.5 ScrollViewer scrlvwr = this.FindName(scrollvwnm) as ScrollViewer; MessageBox.Show(scrollvwnm); // Displays name double checked for accuracy MessageBox.Show(scrlvwr.ActualHeight.ToString()); //Crashes ScrollViewer scrlvwr = ti.FindName(scrollvwnm) as ScrollViewer; MessageBox.Show(scrollvwnm); // Displays name double checked for accuracy MessageBox.Show(scrlvwr.ActualHeight.ToString()); //Also Crashes Is there a method to refresh UI in XAML so the new window is able to access the newly loaded tab item content? Thanks

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• How to get a fully transparent backbuffer in directx 9 without vista Desktop Window Manager

  - by flawlesslyfaulted

  I currently have an activex control that initiates a media (video/audio) framework another development group in my company developed and I am providing a window handle to that code. That handle is being used by their rendering plugin in the pipeline that uses Direct3d for rendering the video using that handle. I have seperate LPDIRECT3D9EX and LPDIRECT3DDEVICE9EX pointers that I initialize in my activex control. I am trying to clear a backbuffer to transparent and then use directx drawing primatives to draw on that backbuffer producing a transparent window with my drawing primatives over the streaming video on the directx surface below. It appears that clearing a device backbuffer with full alpha transparency is ignored by directx. d3ddev->Clear(0, NULL, D3DCLEAR_TARGET, D3DCOLOR_RGBA(0, 0, 1, 0 /*full alpha*/), 1.0f, 0); I can see the object I draw but they are drawn on top of a backbuffer that has the RGB color specified without the alpha value. The project linked (http://www.codeproject.com/KB/directx/umvistad3d.aspx) to in the stackoverflow question below does what I want but requires vista's Desktop Window Manager and won't work for XP. http://stackoverflow.com/questions/148275/how-do-i-draw-transparent-directx-content-in-a-transparent-window I have tried with D3DRS_ALPHABLENDENABLE true with configured blend with no avail. I have also tried to have pixels with full alpha values not rendered using D3DRS_ALPHATESTENABLE, D3DRS_ALPHAREF, and D3DRS_ALPHAFUNC setup but this doesn't work either. I have tried using ColorFill with alpha after retrieving the backbuffer with GetBackBuffer but this doesn't work either. (again only RGB is used) Finally I have tried creating a texture, selecting a surface, colorfilling that surface with a fully transparent alpha value, then loading that surface onto the backbuffer but only the RGB values appear to be used. I have checked the capabilities using the DXCapsViewer.exe and the D3DFMT_A8R8G8B8 backbuffer format that I am using for the backbuffer is valid so it can't be that. Has anyone gotten a transparent backbuffer in directx to work in XP?

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• Adding multiple rss feeds to a script in SCALA InfoChannel Designer 5

  - by godleuf

  Okay, since it is impossible to talk to anyone on the phone or get support through Scala's "forum", I am going to take a shot and see if anyone out there is feeling my pain. I have a client that uses Scala's InfoChannel Designer and Content Manager. I have had to learn this software from scratch and I have to say it hasn't been easy. I think I am at a point where the overall design is set, but I need to implement a couple of things before I can make this happen. RSS feeds are my issue at this point. Multiple RSS feeds to be specific. I need a feed coming in for 3 areas of content: Wiki News (or equivalent), local weather and a stock ticker. I have learned how to setup a "crawl" using a script example available from Scala's file center and copying and pasting into my design. But from what I have learned first hand and through reading through other forums, you can not have a feed from 3 different sources or urls happening simultaneously. Doesn't seem like it would be an issue, but apparently it is. This small step has held up this project for far too long and I need to get it figured out. This doesn't even touch on my issue of feeding in streaming video as a background but I have gone over this in another question but with no luck thus far. If there is ANYONE out there who is in anything similar using this software, your feedback and/or suggestions would be greatly appreciated. Thanks you for allowing me to vent!

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• Logging class using delegates (NullReferenceException)

  - by Leroy Jenkins

  I have created a small application but I would now like to incorporate some type of logging that can be viewed via listbox. The source of the data can be sent from any number of places. I have created a new logging class that will pass in a delegate. I think Im close to a solution but Im receiving a NullReferenceException and I don’t know the proper solution. Here is an example of what Im trying to do: Class1 where the inbound streaming data is received. class myClass { OtherClass otherClass = new OtherClass(); otherClass.SendSomeText(myString); } Logging Class class OtherClass { public delegate void TextToBox(string s); TextToBox textToBox; Public OtherClass() { } public OtherClass(TextToBox ttb) { textToBox = ttb; } public void SendSomeText(string foo) { textToBox(foo); } } The Form public partial class MainForm : Form { OtherClass otherClass; public MainForm() { InitializeComponent(); otherClass = new OtherClass(this.TextToBox); } public void TextToBox(string pString) { listBox1.Items.Add(pString); } } Whenever I receive data in myClass, its throwing an error. Any help you could give would be appreciated.

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• Microsoft products such as Visual Studio 2010 does not require to enter serial number

  - by MainMa

  Hi, I am member of WebsiteSpark and was member of DreamSpark. Both programs enable to download software and provide serial keys to use. Some software like Windows Server has an ISO file to download and a serial number displayed on the website which I must enter during installation. Some other software does not have any serial key. For example, when I downloaded Visual Studio 2010, there was just a link to an ISO file. During installation, there was no such a field as serial number (whereas Visual Studio 2008 had this field at the beginning of installation process). There is the same thing with SQL Server 2008 and Microsoft Expression Studio 3. Even when I've downloaded the public trial RTM version of Windows Seven Enterprise, there were no serial number to enter. I don't think that such expensive products as SQL Server 2008 Enterprise are delivered without serials and online validation, so I suppose that the serial is embedded into the product itself, either in installation binaries or in a separate config file, so is already in the ISO I download so I do not have to enter it. So my question is, how it is done technically? Is each 2 GBs ISO generated on-demand on the server to embed a serial each time this ISO is requested? I suppose that if it is done, it has a huge impact on servers performance (no caching, no streaming...), so what may be the techniques used behind? I want to implement the same feature in a product I intend to ship (to simplify installation by avoiding to ask to enter serial number), but I really don't see how to do it with low impact on server performance.

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• Concurrent WCF calls via shared channel

  - by Kent Boogaart

  I have a web tier that forwards calls onto an application tier. The web tier uses a shared, cached channel to do so. The application tier services in question are stateless and have concurrency enabled. But they are not being called concurrently. If I alter the web tier to create a new channel on every call, then I do get concurrent calls onto the application tier. But I wanted to avoid that cost since it is functionally unnecessary for my scenario. I have no session state, and nor do I need to re-authenticate the caller each time. I understand that the creation of the channel factory is far more expensive than than the creation of the channels, but it is still a cost I'd like to avoid if possible. I found this article on MSDN that states: While channels and clients created by the channels are thread-safe, they might not support writing more than one message to the wire concurrently. If you are sending large messages, particularly if streaming, the send operation might block waiting for another send to complete. Firstly, I'm not sending large messages (just a lot of small ones since I'm doing load testing) but am still seeing the blocking behavior. Secondly, this is rather open-ended and unhelpful documentation. It says they "might not" support writing more than one message but doesn't explain the scenarios under which they would support concurrent messages. Can anyone shed some light on this?

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• Is Stream.Write thread-safe?

  - by Mike Spross

  I'm working on a client/server library for a legacy RPC implementation and was running into issues where the client would sometimes hang when waiting to a receive a response message to an RPC request message. It turns out the real problem was in my message framing code (I wasn't handling message boundaries correctly when reading data off the underlying NetworkStream), but it also made me suspicious of the code I was using to send data across the network, specifically in the case where the RPC server sends a large amount of data to a client as the result of a client RPC request. My send code uses a BinaryWriter to write a complete "message" to the underlying NetworkStream. The RPC protocol also implements a heartbeat algorithm, where the RPC server sends out PING messages every 15 seconds. The pings are sent out by a separate thread, so, at least in theory, a ping can be sent while the server is in the middle of streaming a large response back to a client. Suppose I have a Send method as follows, where stream is a NetworkStream: public void Send(Message message) { //Write the message to a temporary stream so we can send it all-at-once MemoryStream tempStream = new MemoryStream(); message.WriteToStream(tempStream); //Write the serialized message to the stream. //The BinaryWriter is a little redundant in this //simplified example, but here because //the production code uses it. byte[] data = tempStream.ToArray(); BinaryWriter bw = new BinaryWriter(stream); bw.Write(data, 0, data.Length); bw.Flush(); } So the question I have is, is the call to bw.Write (and by implication the call to the underlying Stream's Write method) atomic? That is, if a lengthy Write is still in progress on the sending thread, and the heartbeat thread kicks in and sends a PING message, will that thread block until the original Write call finishes, or do I have to add explicit synchronization to the Send method to prevent the two Send calls from clobbering the stream?

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• Stream PDF to another local App

  - by Nathan

  Hi, I'm currently trying to optimize a small firefox extension that will grab a pdf off the current document and send it to a port that another local application is listening on. Right now it uses a terrifying hackjob of cache viewer. The way I'm getting it is loading the cache, searching through it using the current URL and grabbing the file and saving it to a temp directory. Then I stream the file in, delete the temp, and send it through the socket. Now, my new design, ideally I'd want to build it from scratch and cut out saving it to the local machine at all, and just stream it through the socket. I've been looking at doing something like, //check page to ensure its a pdf //init in/out streams //stream through sock //flush Now, this would be vastly superior to the 400 line hacked up mess I have now, but I'm new to building FF extensions, and after reading a lot about URIs and the file streaming and such I'm probably more confused than when I started trying to fix this three hours ago. I'm okay with sending things through the sockets and whatnot, I understand that, I'm mainly confused about what multitude of interfaces I want to use. Gah! Thanks! Also, long time reader, first time poster!

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