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  • How to check at runtime if a class implements certain interface?

    - by mare
    Let's say I have some content classes like Page, TabGroup, Tab, etc. Certain of those will be implementing my IWidgetContainer interface - it means they will geet an additional field named ContainedItems from the interface and some methods for manipulating this field. Now I need to reflect the fact that some class implements this interface by rendering out some special custom controls in my ASP.NET MVC Views (like jQuery Add/Remove/Move/Reorder buttons). For instance, TabGroup will implement IWidgetContainer because it will contain tabs but a tab will not implement it because it won't have the ability to contain anything. So I have to somehow check in my view, when I render my content objects (The problem is, I use my base class as strong type in my view not concrete classes), whether it implements IWidgetContainer. How is that possible or have I completely missed something? To rephrase the question, how do you reflect some special properties of a class (like interface implementation) in the UI in general (not necessarily ASP.NET MVC)? Here's my code so far: [DataContract] public class ContentClass { [DataMember] public string Slug; [DataMember] public string Title; [DataMember] protected ContentType Type; } [DataContract] public class Group : ContentClass, IWidgetContainer { public Group() { Type = ContentType.TabGroup; } public ContentList ContainedItems { get; set; } public void AddContent(ContentListItem toAdd) { throw new NotImplementedException(); } public void RemoveContent(ContentListItem toRemove) { throw new NotImplementedException(); } } [DataContract] public class GroupElement : ContentClass { public GroupElement() { Type = ContentType.Tab; } } Interface: interface IWidgetContainer { [DataMember] ContentList ContainedItems { get; set; } void AddContent(ContentListItem toAdd); void RemoveContent(ContentListItem toRemove); }

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  • Check packet vlan tag using Tap virtual interface

    - by ankit
    Hi all, I am trying to learn how to implement virtual interfaces using the Tap driver. So far my understanding is that using the tap driver I can create a virtual interface and then have a userspace program attach to this interface to analyse the data coming into this device. Now what if I attach a cisco switch to my LAN interface using a TRUNK link, forward all the packets coming into the LAN interface to the virtual tap interface and then in my program attached to this interface do some coding to analyze the vlan tag in the packet and only allow certain vlans to be forwarded to the WAN interface ? Does this sound plausible or is there is flaw in my basic understanding ? Thanks for the help! ankit

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  • Audio doesn't work on Windows XP guest (WS 7.0)

    - by Mads
    Hi, I can't get audio to work with on a Windows XP guest running on VMware Workstation 7.0 and Ubuntu 9.10 host. Windows fails to produce any audio output and the Windows device manager says the Multimedia Audio Controller is not working properly. Audio is working fine in the host OS. When I open Multimedia Audio Controller properties it says: Device status: The drivers for this device are not installed (Code 28) If I try to reinstall the driver I get the following error message: "Cannot Install this Hardware There was a problem installing this hardware: Multimedia Audio Controller An Error occurred during the installation of the device Driver is not intended for this platform" Has anyone else experienced this problem?

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  • How to make VLC play a .vlm configuration file in "With no interface mode"?

    - by Ole Jak
    How to make VLC play a VLM configuration file (.vlm) in "With no interface mode" on Windows? So I have a VLM configuration file that should stream audio from mic to localhost so no VLC user interface is needed. If I say to Windows "play VLM file with VLC" it plays correctly, starts the server where I need and streams data. But how to do such a thing manually from the command line (so we suppose we can call vlc.exe by VLC and we are now in folder with the vlc.exe and vlcConfig.vlm files)?

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  • How to stop ethernet interface in bridge configuration from obtaining IP address via DHCP

    - by user71061
    Hi! I'm trying to configure openvpn in bridging configuration. First step of doing this requires creating bridge interface (br0), bridging together physical ethernet interface (eth0) and logical tap0 interface. This can be done with simple script but I want to use less popular approach, configuring bridge interface entirely via /etc/network/interfaces file (on Debian linux). So I have removed all eth0 definitions form /etc/network/interfaces and replaced if with following br0 definition: auto br0 iface br0 inet static pre-up openvpn --mktun --dev tap0 address 10.0.0.1 netmask 255.255.255.0 bridge_ports eth0 tap0 post-down openvpn --rmtun --dev tap0 This works as I expected, but there is only one problem: interface eth0 is part of bridge interface br0 AND it also receive it's own IP address from my DHCP server (located on same LAN where eth0 is connected). My questions is: how to stop eth0 interface from obtaining it's own IP address? (It should only be part of br0 bridge).

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  • Where does a Quick-Time powered application store changes to the user interface?

    - by Luke
    I have downloaded an application written with the Quick Time library (for Windows 7). The application does not need an installation: just unzip it in a directory and run the program. It works, but I have a problem: the program allows the user to change a lot of values using its interface but does not have an option to reset them to their default values. What is more problematic is that when I exit the program and run it again, the interface still has the changed values. In the program directory there is no file that stores the changes done to the UI of the program. I suspect that Quick Time records these changes somewhere, but I can't find the right file. I have even deleted the application and re-unzipped it to another location - but the UI values still remain the same values changed by me!

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  • Bsplayer - load audio tracks from external files

    - by torran
    I have a movie file: Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 5 frames Muxing mode : Container [email protected] Codec ID : V_MPEG4/ISO/AVC Duration : 54mn 13s Bit rate : 3 380 Kbps Nominal bit rate : 3 459 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 23.976 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.153 Stream size : 1.28 GiB (88%) Writing library : x264 core 88 r1471 1144615 Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Codec ID : A_AC3 Duration : 54mn 16s Bit rate mode : Constant Bit rate : 384 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Stream size : 149 MiB (10%) and additional audio files in same folder: .mp3 and .ac3. How can I load them with bsplayer? Right click-audio-audio streams is empty. If i open the movie with media players classic I can switch audio files.

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  • How do you setup the Audio plugin for Flowplayer?

    - by codeninja
    I'm having a bit of trouble getting the Audio player to work. Basically I want to initiate an mp3 player doing something like this <a href="path-to-my-audio.mp3" id="player" ></a> and then use the $f() call to initate the player. I've followed the instructions here (http://flowplayer.org/plugins/streaming/audio.html) This doesnt seem to be work and I'm not sure what's wrong because I'm able to play videos in this way. Thanks for your help!

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  • Core Audio on iPhone - any way to change the microphone gain (either for speakerphone mic or headpho

    - by Halle
    After much searching the answer seems to be no, but I thought I'd ask here before giving up. For a project I'm working on that includes recording sound, the input levels sound a little quiet both when the route is external mic + speaker and when it's headphone mic + headphones. Does anyone know definitively whether it is possible to programmatically change mic gain levels on the iPhone in any part of Core Audio? If not, is it possible that I'm not really in "speakerphone" mode (with the external mic at least) but only think I am? Here is my audio session init code: OSStatus error = AudioSessionInitialize(NULL, NULL, audioQueueHelperInterruptionListener, r); [...some error checking of the OSStatus...] UInt32 category = kAudioSessionCategory_PlayAndRecord; // need to play out the speaker at full volume too so it is necessary to change default route below error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); if (error) printf("couldn't set audio category!"); UInt32 doChangeDefaultRoute = 1; error = AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof (doChangeDefaultRoute), &doChangeDefaultRoute); if (error) printf("couldn't change default route!"); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); UInt32 inputAvailable = 0; UInt32 size = sizeof(inputAvailable); error = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, &size, &inputAvailable); if (error) printf("ERROR GETTING INPUT AVAILABILITY! %d\n", (int)error); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioInputAvailable, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); error = AudioSessionSetActive(true); if (error) printf("AudioSessionSetActive (true) failed"); Thanks very much for any pointers.

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  • What is the best service/tool to put short audio clips on a website so users can click and listen im

    - by Edward Tanguay
    I'm making a foreign language flashcard website in which I want to have 100s of short 3-10 second audio files available for users to click and listen. So I am looking for a tool/service such as YouTube or Screenr.com but for audio which e.g.: allows me to easily upload multiple kinds of audio files: mp3, wav, etc. easy to manage them online (delete, replace) has a simple, small player (e.g. flash) that integrates nicely into any site

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  • Html5 Audio plays only once in my Javascript code.

    - by Poul
    I have a dashboard web-app that I want to play an alert sound if its having problems connecting. The site's ajax code will poll for data and throttle down its refresh rate if it can't connect. Once the server comes back up, the site will continue working. In the mean time I would like a sound to play each time it can't connect (so I know to check the server). Here is that code. This code works. var error_audio = new Audio("audio/"+settings.refresh.error_audio); error_audio.load(); //this gets called when there is a connection error. function onConnectionError() { error_audio.play(); } However the 2nd time through the function the audio doesn't play. Digging around in Chrome's debugger the 'played' attribute in the audio element gets set to true. Setting it to false has no results. Any ideas?

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  • Everything has an Interface [closed]

    - by Shane
    Possible Duplicate: Do I need to use an interface when only one class will ever implement it? I am taking over a project where every single real class is implementing an Interface. The vast majority of these interfaces are implemented by a single class that share a similar name and the exact same methods (ex: MyCar and MyCarImpl). Almost no 2 classes in the project implement more than the interface that shares its name. I know the general recommendation is to code to an interface rather than an implementation, but isn't this taking it a bit too far? The system might be more flexible in that it is easier to add a new class that behaves very much like an existing class. However, it is significantly harder to parse through the code and method changes now require 2 edits instead of 1. Personally, I normally only create interfaces when there is a need for multiple classes to have the same behavior. I subscribe to YAGNI, so I don't create something unless I see a real need for it. Am I doing it all wrong or is this project going way overboard?

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  • How accurately (in terms of time) does Windows play audio?

    - by MusiGenesis
    Let's say I play a stereo WAV file with 317,520,000 samples, which is theoretically 1 hour long. Assuming no interruptions of the playback, will the file finish playing in exactly one hour, or is there some occasional tiny variation in the playback speed such that it would be slightly more or slightly less (by some number of milliseconds) than one hour? I am trying to synchronize animation with audio, and I am using a System.Diagnostics.Stopwatch to keep the frames matching the audio. But if the playback speed of WAV audio in Windows can vary slightly over time, then the audio will drift out of sync with the Stopwatch-driven animation. Which leads to a second question: it appears that a Stopwatch - while highly granular and accurate for short durations - runs slightly fast. On my laptop, a Stopwatch run for exactly 24 hours (as measured by the computer's system time and a real stopwatch) shows an elapsed time of 24 hours plus about 5 seconds (not milliseconds). Is this a known problem with Stopwatch? (A related question would be "am I crazy?", but you can try it for yourself.) Given its usage as a diagnostics tool, I can see where a discrepancy like this would only show up when measuring long durations, for which most people would use something other than a Stopwatch. If I'm really lucky, then both Stopwatch and audio playback are driven by the same underlying mechanism, and thus will stay in sync with each other for days on end. Any chance this is true?

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  • How to programmatically generate an audio podcast file with chapters and text track?

    - by adib
    Hi Anybody know how to programmatically generate audio podcast files with bookmarks that can be used in iTunes / iPod / iPhone / iPod touch? Specifically text bookmarks (bookmarks with titles) that the listener can skip to a specific point in time in the audio file. Also how to add the text transcription of the podcast's content. Even better if you have an example Cocoa code or library to write the audio file. Thanks.

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  • Wifi interface changes name seemingly at random

    - by ray_voelker
    I'm currently having some issues getting a wireless interface to work continuously under an install of Ubuntu 12.04.1 LTS. Some of the issues I'm experiencing include Connection will drop out after some time after it has initially worked. Interface will be a different name after a reboot. For example, wlan0 will become wlan4 when using the ifconfig -a command. Ubuntu will take a long time to boot, looking for network adapters. The purpose of this build is to function as a web kiosk in a library. The computer is supposed to boot up into a web browser, and allow for browsing of the catalog. For some reason this interface does not appear to be working as it should. Are there any explanations for some of these problems I'm having, and perhaps some solutions? The wireless card appears as this after doing an lspci ... Ralink corp. RT2561/RT61 802.11g PCI In the /etc/network/interfaces file I have the following configuration for the interface. auto wlan0 iface wlan0 inet dhcp wireless-essid UDwireless wireless-mode Managed Thanks in advance for help on this.

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  • how do i merge two audio files and one video file in to a video file using c# ?

    - by wingdings
    i wrote a program in c# using directshow , that captures all devices' audios , and video from single device (webcam or external camera) , now that my requirement is to merge selected audio files with one video file and i can not get it done in c#. so i need a program or libraries that merges one(or several) audio file(s) and one video file and save it as an avi VIDEO file ,, both audio file and video files are in avi format.

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  • How to speed up drawing of scaled image? Audio playback chokes during window resize.

    - by Paperflyer
    I am writing an audio player for OSX. One view is a custom view that displays a waveform. The waveform is stored as a instance variable of type NSImage with an NSBitmapImageRep. The view also displays a progress indicator (a thick red line). Therefore, it is updated/redrawn every 30 milliseconds. Since it takes a rather long time to recalculate the image, I do that in a background thread after every window resize and update the displayed image once the new image is ready. In the meantime, the original image is scaled to fit the view like this: // The drawing rectangle is slightly smaller than the view, defined by // the two margins. NSRect drawingRect; drawingRect.origin = NSMakePoint(sideEdgeMarginWidth, topEdgeMarginHeight); drawingRect.size = NSMakeSize([self bounds].size.width-2*sideEdgeMarginWidth, [self bounds].size.height-2*topEdgeMarginHeight); [waveform drawInRect:drawingRect fromRect:NSZeroRect operation:NSCompositeSourceOver fraction:1]; The view makes up the biggest part of the window. During live resize, audio starts choking. Selecting the "big" graphic card on my Macbook Pro makes it less bad, but not by much. CPU utilization is somewhere around 20-40% during live resizes. Instruments suggests that rescaling/redrawing of the image is the problem. Once I stop resizing the window, CPU utilization goes down and audio stops glitching. I already tried to disable image interpolation to speed up the drawing like this: [[NSGraphicsContext currentContext] setImageInterpolation:NSImageInterpolationNone]; That helps, but audio still chokes during live resizes. Do you have an idea how to improve this? The main thing is to prevent the audio from choking.

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  • Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web Audio

    Firefox pour Android introduit la « navigation en tant qu'invité » et le support de l'API Web AudioA la suite de la sortie de Firefox 25, Mozilla a également publié une mise à jour de son navigateur pour les possesseurs de terminaux sous Android.Firefox pour Android hérite de quelques fonctionnalités de version desktop, notamment la prise en charge de l'API Web Audio, une spécification du W3C pour les effets audio avancés à partir de HTML5. Cette nouvelle API permettra, par exemple, aux ingénieurs...

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  • Linux based audio prodcuction tutorials

    - by thelinuxer
    I have been searching for a while for Linux based audio production tutorials. All I can find is tool based tutorials. For example I found tutorials on how to use jack, ardour, lmms ..etc. What I need is tutorials that teaches professional audio production with opensource/free tools, like those already available for protools and likes. If any one can guide me to any videos/articles available it would be highly appreciated. Thanks.

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  • Get rid of 0.5s latency when playing audio over Bluetooth with A2DP

    - by brillout.com
    As described in the title I experience a half a second delay when playing audio over Bluetooth with A2DP. This makes watching movies not possible as the sound is not synchronised with the video. I'm not sure if the delay is caused by the Bluetooth connection, the A2PD protocol, or the A2DP implementation on my Ubuntu 12.04. Anyways, is this a normal lag? Is there a way to play audio over Bluetooth without any latency?

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  • New HP dm4 - No audio on ubuntu 11.10 64bits

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work. Thanks in advance

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  • No audio on an HP dm4

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work.

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