Search Results

Search found 16362 results on 655 pages for 'audio interface'.

Page 42/655 | < Previous Page | 38 39 40 41 42 43 44 45 46 47 48 49  | Next Page >

  • C# How to check if a class implements generic interface ?

    - by PaN1C_Showt1Me
    How to get generic interface type for an instance ? Suppose this code: interface IMyInterface<T> { T MyProperty { get; set; } } class MyClass : IMyInterface<int> { #region IMyInterface<T> Members public int MyProperty { get; set; } #endregion } MyClass myClass = new MyClass(); /* returns the interface */ Type[] myinterfaces = myClass.GetType().GetInterfaces(); /* returns null */ Type myinterface = myClass.GetType().GetInterface(typeof(IMyInterface<int>).FullName);

    Read the article

  • How to create a cross-plataform application, doing the interface modules (Mac/Qt/GTK+) in a totally

    - by Somebody still uses you MS-DOS
    I'm amazed at Transmission, a BT client. It has a Mac, a GTK+, a QT, a Web Client and a CLI interface to it. I tried reading some of it's source to understand how he creates all these interfaces, but no luck. Does the developer creates them using a single ide? Or does he create the interface logic in each specific environment (specially mac), "exports" this window code and integrates with the main logic? Is it possible to create that mac interface in another OS using an IDE? How did the developers create this software with so many interfaces, in a independent way?

    Read the article

  • Creating an Interface To a Language's Standard Library?

    - by Nathan Arthur
    In the process of learning test-driven development, I've been introduced to dependency injection and the use of interfaces, and have started using these concepts in my own PHP code in order to make it more testable. There have been times when I've needed to test code that was doing things like calling the PHP time() function. In order to make these tests predictable, it seemed logical to create an interface to the standard PHP functions I use so that I can mock them out in my tests. Is this good software design? What are the pros and cons of doing this? I've found myself groaning at how quickly my PHP interface can stick its fingers into everything I do. Is there a better way to make code that relies on PHP-accessed state and functions more testable?

    Read the article

  • ERROR while getting interface

    - by user284391
    I have installed the latest version of aircrack-ng, but when i run this code, sudo airmon-ng start wlan0 I get this. sudo airmon-ng start wlan0 Found 4 processes that could cause trouble. If airodump-ng, aireplay-ng or airtun-ng stops working after a short period of time, you may want to kill (some of) them! -e PID Name 463 avahi-daemon 475 avahi-daemon 683 NetworkManager 756 wpa_supplicant Interface Chipset Driver wlan0 Broadcom wl - [phy0]mon0: ERROR while getting interface flags: No such device (monitor mode enabled on mon0) Is there anyone who could help me get this problem solved please.

    Read the article

  • virsh XML interface allocation

    - by Kaushik Koneru
    I am trying to launch VM using a XML. This VM will be having 5 interfaces each connected to certain bridge. Issue here is allocation of these interfaces is random. My XML <interface type='bridge'> <mac address='52:54:00:9f:14:b3'/> <source bridge='br0'/> <target dev='vnet1'/> <model type='e1000'/> <alias name='net0'/> <address type='pci' domain='0x0000' bus='0x00' slot='0x03' function='0x0'/> </interface> <interface type='bridge'> <mac address='52:54:00:9f:14:b4'/> <source bridge='br1'/> <target dev='vnet2'/> <model type='e1000'/> <alias name='net1'/> <address type='pci' domain='0x0000' bus='0x00' slot='0x10' function='0x0'/> </interface> <interface type='bridge'> <mac address='52:54:00:9f:14:b5'/> <source bridge='br2'/> <target dev='vnet2'/> <model type='e1000'/> <alias name='net3'/> <address type='pci' domain='0x0000' bus='0x00' slot='0x12' function='0x0'/> </interface> <interface type='bridge'> <mac address='52:54:00:9f:14:c4'/> <source bridge='br3'/> <target dev='vnet3'/> <model type='e1000'/> <alias name='net4'/> <address type='pci' domain='0x0000' bus='0x00' slot='0x18' function='0x0'/> </interface> Allocation of interfaces are random mean e th6 will be connected to br3 ; eth7 -- br4 eth8 -- br2 eth9 -- br0. Is there any way to make it static?? At the same time is there anyway of assigning IP Address to these eth interfaces through XML file itself??

    Read the article

  • Using wireless interface in guest OS with bridged network in VMware fusion 3

    - by Chetan
    I'm running Ubuntu in Snow Leopard with VMware fusion 3, and I want to be able to access the wireless network on eth1 within Ubuntu so I can run tools like aircrack-ng. However, the bridged network that VMware sets up connects my Airport interface in Mac to the wired interface eth0 in Ubuntu. How do I set it up so that the Airport interface is connected to the wireless interface eth1 in Ubuntu?

    Read the article

  • Cisco Switching Module and HSRP interface Tracking

    - by Kyle Brandt
    When using 4 port switching module where each port is configured to switchport access vlan ##, for HRSP should I track the vlan interface or the FastEthernet interface? interface FastEthernet0/0/0 switchport access vlan 10 interface Vlan10 ip address 12.12.12.1 255.255.255.0 int FastEthernet0/1 ip address 192.168.1.2 255.255.255.0 standyby ip 192.168.128.1 standby track ?? ! FastEthernet 0/0/0 or Vlan 10?

    Read the article

  • Split UInt32 (audio frame) into two SInt16s (left and right)?

    - by morgancodes
    Total noob to anything lower-level than Java, diving into iPhone audio, and realing from all of the casting/pointers/raw memory access. I'm working with some example code wich reads a WAV file from disc and returns stereo samples as single UInt32 values. If I understand correctly, this is just a convenient way to return the 32 bits of memory required to create two 16 bit samples. Eventually this data gets written to a buffer, and an audio unit picks it up down the road. Even though the data is written in UInt32-sized chunks, it eventually is interpreted as pairs of 16-bit samples. What I need help with is splitting these UInt32 frames into left and right samples. I'm assuming I'll want to convert each UInt32 into an SInt16, since an audio sample is a signed value. It seems to me that for efficiency's sake, I ought to be able to simply point to the same blocks in memory, and avoid any copying. So, in pseudo-code, it would be something like this: UInt32 myStereoFrame = getFramefromFilePlayer; SInt16* leftChannel = getFirst16Bits(myStereoFrame); SInt16* rightChannel = getSecond16Bits(myStereoFrame); Can anyone help me turn my pseudo into real code?

    Read the article

  • Record 8 separate Line IN Channels from M-Audio Delta 1010 Card

    - by Peter Hoffmann
    I want to record the 8 separate Line IN Channels from my M-Audio Delta 1010 Card. The card is recogniced nicely and a can record a single channel via arecord -d 10 -f cd -t wav -D channel1 out2.wav. I've set up the different channels in ~/.asoundrc. Now if I want to record a second channel in parallel (arecord -d 10 -f cd -t wav -D channel2 out2.wav) I get the error arecord: main:564: audio open error: Device or resource busy As I understand the delta 1010 is a single Access Card, so only one application can access it at a time. Is this correct? The next step was to configure a dual channel input in .asoundrc # envy24 channel 1+2 only pcm.test { type plug ttable.0.0 1 ttable.0.1 1 slave.pcm ice1712 } Which works ok when I do a arecord -d 10 -f cd -t wav -D test -c 2 out.wav (BTW can anyone point me to a tool to split a multi channel wav into a file per channel?) But when I want to record the channels separately with (-I option) arecord -d 10 -f cd -t wav -D test -c 2 -I channel1.wav channel2.wav I get no recordings. Did I miss something with the configuration or what are my options to record all 8 channels via arecord. I've no experience with jackd. Is it an option to install jackd and record the line ins via jackd?

    Read the article

  • bluetooth headset can connect, but not visible in pulse audio

    - by Kim Marivoet
    I have a plantronics bluetooth headset, and until yesterday I could use it without any problem. However, today it suddenly stopped working (maybe related to the last software update I did). I can still connect/disconnect my headset, but it doesn't show up in pulse audio anymore. I read through various posts that describes kind of the same problem, but none of the suggested solutions worked. I get following error in the syslog: Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/HFPAG Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSource Oct 13 16:49:57 desktop bluetoothd[1040]: Endpoint registered: sender=:1.34 path=/MediaEndpoint/A2DPSink Oct 13 16:50:09 desktop kernel: [ 17.340943] input: 48:C1:AC:08:FE:8F as /devices/virtual/input/input14 Oct 13 16:50:09 desktop bluetoothd[1040]: /org/bluez/1040/hci0/dev_48_C1_AC_08_FE_8F/fd0: fd(36) ready Oct 13 16:50:09 desktop rtkit-daemon[1894]: Successfully made thread 2213 of process 1892 (n/a) owned by '1000' RT at priority 5. Oct 13 16:50:09 desktop rtkit-daemon[1894]: Supervising 5 threads of 1 processes of 1 users. Oct 13 16:50:10 desktop bluetoothd[1040]: Badly formated or unrecognized command: AT+XEVENT=USER-AGENT,COM.PLANTRONICS,PLT_VOYAGERPRO,0109,27.90,FFFFFFFFFFFFFFFFFFFFFFFFFFFFFFFF Oct 13 16:50:10 desktop bluetoothd[1040]: Audio connection got disconnected Any help would be much appreciated. I'm using Ubuntu 12.04. Thanks, Kim

    Read the article

  • Setting up Beats audio on HP Pavilion m6

    - by Joel Auterson
    I have an HP Pavilion m6-1054sa laptop, with a Beats subwoofer on the bottom. The normal laptop speakers work fine under Ubuntu but the Beats speaker(s?) does not. Anyone know how to get this working? Here's my lspci output, if it helps... 00:00.0 Host bridge: Intel Corporation Ivy Bridge DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Ivy Bridge PCI Express Root Port (rev 09) 00:02.0 VGA compatible controller: Intel Corporation Ivy Bridge Graphics Controller (rev 09) 00:14.0 USB controller: Intel Corporation Panther Point USB xHCI Host Controller (rev 04) 00:16.0 Communication controller: Intel Corporation Panther Point MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 1 (rev c4) 00:1c.1 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 2 (rev c4) 00:1d.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Panther Point LPC Controller (rev 04) 00:1f.2 RAID bus controller: Intel Corporation 82801 Mobile SATA Controller [RAID mode] (rev 04) 00:1f.3 SMBus: Intel Corporation Panther Point SMBus Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Thames XT/GL [Radeon HD 7600M Series] (rev ff) 07:00.0 Unassigned class [ff00]: Realtek Semiconductor Co., Ltd. Device 5289 (rev 01) 07:00.2 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 0a) 08:00.0 Network controller: Intel Corporation Centrino Wireless-N 2230 (rev c4)

    Read the article

  • Problem Installing Xubuntu 12.04 Audio-Video codecs

    - by Seib
    I used Crouton to install Xubuntu 12.04.4 LTS with the XFCE environment on my HP Chromebook 11. I've gotten it all fully installed and everything; the only thing I'm doing now is the basic setup things, like adding codecs and other things like LibreOffice, VLC, Firefox, Ubuntu Software Center, etc. This information I got from 2 sources: http://www.efytimes.com/e1/fullnews.asp?edid=137269 http://www.binarytides.com/better-xubuntu-14-04/ . I'm currently on the same step at both URLs, which is #6 on Link 1 and #8 on Link 2. Per the articles, which both said the same thing, I typed in sudo apt-get install xubuntu-restricted-extras libavcodec-extra and it didn't do anything. It just kept on saying the same thing: Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package libavcodec-extra I've spend the last hour or so scouring the internet for a solution, for a hint even at what is going on. I don't want to do a clean reinstall, for two reasons: 1) it's takes like 1.5h just to get the croot fully installed, and 2) everything but this out of what I've done so far (up to #6 at Link 1 & up to #8 at Link 2) works except the audio. I've already installed flash, so YouTube works fine. It's just I can't hear any audio. Please help? Thanks in advance. I appreciate all the great help I've been getting from AskUbuntu lately. You all are great.

    Read the article

  • M-Audio Delta starts up at wrong sample rate

    - by steevc
    When the PC starts my M-Audio Delta 66 is using 48000kHz sampling rate when it is set for 44100 in Envy24. This causes audio to play slower than it should. This is in Kubuntu 14.04 on my new PC using an AMD A8 6500 with 8GB. When I first installed it seemed okay, but at some point it went wrong and has been doing this consistently since then. Kernel is 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:31:42 UTC 2014 i686 athlon i686 GNU/Linux steve@slarti:~$ cat /proc/asound/card2/pcm0p/sub0/hw_params access: MMAP_INTERLEAVED format: S32_LE subformat: STD channels: 10 rate: 48000 (48000/1) period_size: 441 buffer_size: 3528 I can get it to switch to 44100 if I disable/enable the Delta in Pulseaudio volume control, but I have to do this every day and the sound still seems distorted. I can't see any issues in any of the config or log files I can find. If I boot the PC with a Mint Live USB it starts at 44100 and sound fine. Originally reported on Youtube is playing at the wrong speed - maybe soundcard related, but I'll close that and have this more relevant question instead.

    Read the article

  • ffmpeg add two audio streams to video

    - by Tossin Hausen
    I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

    Read the article

  • 5.1 sound in Unity3d 3.5

    - by N0xus
    I'm trying to implement 5.1 surround sound in my game. I've set Unity's AudioManager to a default of 5.1 surround and loaded in a 6 channel audio clip that should play a sound in each of the different audio spots. However, when I go to run my game, all I get is flat sound coming out of my front two speakers. Even then, these don't play the sound they should (front speaker should play "front speaker" right should play "right speaker" and so). Both speakers just end up playing the entire sound file. I've tried looking to see if there is a parameter that I have missed, but information on how to set up 5.1 sound in Unity is lacking (or my google skills aren't that good) and I can't get it to work as intended. Could someone please either tell me what I'm missing, or point me in the right direction? My audio source is situated at point (0, 0, 0) with my camera also being in the same point. I've moved about the scene but the same thing happens as I've already described.

    Read the article

  • Alsa devices under Wine

    - by Roberto Aloi
    Hi all, I'm running OpenSuse 11.2 and Wine 1.1.28. Even if audio is perfectly working fine for me (Skype, Banshee, etc), when I try to configure audio for Wine (to use Spotify) I cannot hear anything from the audio test. In the winecfg audio tab, ALSA is checked, but no devices are available. I tried to run alsaconf (it needs root permissions) but it returns: No supported PnP or PCI card found No legacy drivers available, either. Any idea?

    Read the article

  • Implicit and Explicit implementations for Multiple Interface inheritance

    Following C#.NET demo explains you all the scenarios for implementation of Interface methods to classes. There are two ways you can implement a interface method to a class. 1. Implicit Implementation 2. Explicit Implementation. Please go though the sample. using System; namespace ImpExpTest {     class Program     {         static void Main(string[] args)         {             C o3 = new C();             Console.WriteLine(o3.fu());             I1 o1 = new C();             Console.WriteLine(o1.fu());             I2 o2 = new C();             Console.WriteLine(o2.fu());             var o4 = new C();       //var is considered as C             Console.WriteLine(o4.fu());             var o5 = (I1)new C();   //var is considered as I1             Console.WriteLine(o5.fu());             var o6 = (I2)new C();   //var is considered as I2             Console.WriteLine(o6.fu());             D o7 = new D();             Console.WriteLine(o7.fu());             I1 o8 = new D();             Console.WriteLine(o8.fu());             I2 o9 = new D();             Console.WriteLine(o9.fu());         }     }     interface I1     {         string fu();     }     interface I2     {         string fu();     }     class C : I1, I2     {         #region Imicitly Defined I1 Members         public string fu()         {             return "Hello C"         }         #endregion Imicitly Defined I1 Members         #region Explicitly Defined I1 Members         string I1.fu()         {             return "Hello from I1";         }         #endregion Explicitly Defined I1 Members         #region Explicitly Defined I2 Members         string I2.fu()         {             return "Hello from I2";         }         #endregion Explicitly Defined I2 Members     }     class D : C     {         #region Imicitly Defined I1 Members         public string fu()         {             return "Hello from D";         }         #endregion Imicitly Defined I1 Members     } } Output:- Hello C Hello from I1 Hello from I2 Hello C Hello from I1 Hello from I2 Hello from D Hello from I1 Hello from I2 span.fullpost {display:none;}

    Read the article

  • Introducing the Metro User Interface on Windows 2012

    - by andywe
    Although I am a big fan of using PowerShell to do many of my server operations, that aspect is well covered by those far more knowledgeable than I, and there is vast information around the web already on that. The new Metro interface, and getting around both Windows 8 and Windows Server 2012 though is relatively new, even for those whop ran the previews. What is this? A blank Desktop!   Where did the start button go? Well, it is still there...sort of. It is hidden, and acts like an auto hidden component that appear only when the mouse is hovered over the lower left corner of the screen. Those familiar with Gnome or OSX can relate this to the "Hot Corners" functions. To get to the start button, hover your mouse in the very left corner of the task bar. Let it sit there a moment, and a small blue square with colored tiles in it called start will appear. Click it. I clicked it and now I have all the tiles..What is this?   Welcome to the Metro interface. This is a much more modern look, and although at first seems weird and cumbersome, I have actually found that it is a bit more extensible, allowing greater organization and customization than the older explorer desktop. If you look closely, you'll see each box represents either a program, or program group. First, a few basics about using the start view. First and foremost, a right mouse click will bring up a bar on the bottom, with an icon towards the right. Notice it is titled “All Apps”. An even easier way in many places is to hover your mouse in the exact opposite corner, in the upper right. A sidebar will open and expose what used to be a widget bar (remember Vista?), and there are options for Search, Start, and Settings.   Ok Great, but where is everything? It’s all there…Click the All Apps icon.   Look better? Notice the scroll bar at the bottom. Move it right..your desktop is sized to your content..so you can have a smaller, or larger amount of programs exposed. Each icon can be secondary clicked (right mouse click for most of us, and an options bar at the bottom, rather than the old small context menu, is opened with some very familiar options.   Notice the top of the Windows Explorer window has some new features. You still have your right mouse click functions, but since the shortcuts for these items already exist..just copy them. There are many ways, but here is a long way to show you more of the interface. 1. Right mouse click a program icon, and select the Open File Location option. 2. Trusty file manager opens…but if you look closely up at top edge of the window, you’ll see a nifty enhancement. An orange colored box that is titled Shortcut Tools and another lavender box Title Application tools. Each of these adds options at the top of the file manager window to make selection easy. Of course, you can still secondary click an item in the listing window too. 3. Click shortcut tools, right click your app shortcut and copy it. Then simply paste it into the desktop outside the File Explorer window Also note some of the newer features. The large icons up top below the menu that has many common operations. The options change as you select each menu item. Well, that’s it for this installment. I hope this helps you out.

    Read the article

  • Scriptable user-interfaces/frameworks for automated UI testing

    - by AareP
    I'm planning on using scripting for automated UI testing. Main application is written in c#, and I want it to be scriptable, so I can do everything end-user can do, but programmatically. What do you think of software that provides an interface for scripting, like VBA macros in Excel? Can this be future of all programming, big and small? What is the best way to build such an interface for your own application, dll-based or by parsing own scripting language?

    Read the article

  • Audio stream mangement in Linux

    - by User1
    I have a very complicated audio setup for a project. Here's what we have: 3 applications playing sound 2 applications recording sound 2 sound cards I really don't really have the code to any of these applications. All I want to do is monitor and control the audio streams. Here are a few examples of operations I'd like to do while the applications are running: Mute one of the incoming audio streams. Have one of the incoming audio streams do a "solo" (be the only stream that can "talk"). Get a graph (about 30 seconds worth) of the audio that each stream produced. Send one of the audio streams to soundcard #1, but all three audio streams to soundcard #2. I would likely switch audio streams every 2 minutes or so with one of the operations listed above. A GUI would be preferred. I started looking at the sound systems in Linux and it gets extremely complex and I feel like there have been many new advances in the past few years. I see jack, pulseaudio, artsd, and several other packages. They all have some promise but where should I start? Is there something someone already built that can help?

    Read the article

  • Cannot Access Web Interface on HP 2510G

    - by Stephen
    I am currently setting up a new infrastructure with HP 2510s as edge switches and an HP E5406 as the main switch. I also have a DHCP and DNS server running on the same network. When i first set up one of my 2510 switches, I gave it a static IP through the console and then went to the web interface to continue my configuration. Later, I realized that I assigned it the wrong IP address, so i went through the web interface and changed the IP address to the correct one. Now, I can't access the web interface. I can telnet to the switch on the new IP address, but the web interface will not load. If I switch from static IP to DHCP, it loads the web interface. Any ideas on what could be causing the web server in the 2510 not to load with the new static IP address?

    Read the article

  • Is there an easy way to copy an audio CD in Mac OS X?

    - by Bob D
    (not a commercial CD). I did some recordings of a band years ago and ran into one of the band members who asked me if I could make copies. I assumed that this would be easy. I know that I can rip the CD into iTunes and then burn a new CD, but I have two optical drives available, is there a way to simply copy the CD from one drive to the other in one step?

    Read the article

  • Ubuntu: how to get audio to work in both Spotify (under Wine) and Flash (in Firefox)?

    - by Jonik
    I'm running Spotify on Linux using Wine. Sound worked great (even though the sound test in winecfg failed!), until I installed alsa-oss package yesterday to get Flash sound working in Firefox. Now Spotify says: "There is a problem with your sound card. Spotify can't play music." So the question is, how to get the sound in Spotify working again, so that it also keeps working in Flash & Firefox? Tweak some ALSA settings? Spotify settings? Add/remove some packages? By the way, curiously, now that sound doesn't work in Spotify, winecfg's "Test Sound" does work! This is Ubuntu 8.04 (Hardy). Sound card / driver is probably an integrated AC'97. Please mention if any additional information about the system is needed! Update: I have Flash 10 installed (outside the packaging system, using $MOZ_PLUGIN_PATH env variable), but also had Flash 9 from flashplugin-nonfree package - and the earlier version was being used by Firefox! Based on what Mike Arthur said about Flash and alsa-oss, I removed the older Flash (flashplugin-nonfree package) and alsa-oss - and Flash sound still works, which is nice. But for some reason Spotify still doesn't play sound, even though things should now be like they were originally... Update 2: Got it working, all smoothly, finally.

    Read the article

< Previous Page | 38 39 40 41 42 43 44 45 46 47 48 49  | Next Page >