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  • Visual Studio 2013 Static Code Analysis in depth: What? When and How?

    - by Hosam Kamel
    In this post I'll illustrate in details the following points What is static code analysis? When to use? Supported platforms Supported Visual Studio versions How to use Run Code Analysis Manually Run Code Analysis Automatically Run Code Analysis while check-in source code to TFS version control (TFSVC) Run Code Analysis as part of Team Build Understand the Code Analysis results & learn how to fix them Create your custom rule set Q & A References What is static Rule analysis? Static Code Analysis feature of Visual Studio performs static code analysis on code to help developers identify potential design, globalization, interoperability, performance, security, and a lot of other categories of potential problems according to Microsoft's rules that mainly targets best practices in writing code, and there is a large set of those rules included with Visual Studio grouped into different categorized targeting specific coding issues like security, design, Interoperability, globalizations and others. Static here means analyzing the source code without executing it and this type of analysis can be performed through automated tools (like Visual Studio 2013 Code Analysis Tool) or manually through Code Review which already supported in Visual Studio 2012 and 2013 (check Using Code Review to Improve Quality video on Channel9) There is also Dynamic analysis which performed on executing programs using software testing techniques such as Code Coverage for example. When to use? Running Code analysis tool at regular intervals during your development process can enhance the quality of your software, examines your code for a set of common defects and violations is always a good programming practice. Adding that Code analysis can also find defects in your code that are difficult to discover through testing allowing you to achieve first level quality gate for you application during development phase before you release it to the testing team. Supported platforms .NET Framework, native (C and C++) Database applications. Support Visual Studio versions All version of Visual Studio starting Visual Studio 2013 (except Visual Studio Test Professional) check Feature comparisons Create and modify a custom rule set required Visual Studio Premium or Ultimate. How to use? Code Analysis can be run manually at any time from within the Visual Studio IDE, or even setup to automatically run as part of a Team Build or check-in policy for Team Foundation Server. Run Code Analysis Manually To run code analysis manually on a project, on the Analyze menu, click Run Code Analysis on your project or simply right click on the project name on the Solution Explorer choose Run Code Analysis from the context menu Run Code Analysis Automatically To run code analysis each time that you build a project, you select Enable Code Analysis on Build on the project's Property Page Run Code Analysis while check-in source code to TFS version control (TFSVC) Team Foundation Version Control (TFVC) provides a way for organizations to enforce practices that lead to better code and more efficient group development through Check-in policies which are rules that are set at the team project level and enforced on developer computers before code is allowed to be checked in. (This is available only if you're using Team Foundation Server) Require permissions on Team Foundation Server: you must have the Edit project-level information permission set to Allow typically your account must be part of Project Administrators, Project Collection Administrators, for more information about Team Foundation permissions check http://msdn.microsoft.com/en-us/library/ms252587(v=vs.120).aspx In Team Explorer, right-click the team project name, point to Team Project Settings, and then click Source Control. In the Source Control dialog box, select the Check-in Policy tab. Click Add to create a new check-in policy. Double-click the existing Code Analysis item in the Policy Type list to change the policy. Check or Uncheck the policy option based on the configurations you need to perform as illustrated below: Enforce check-in to only contain files that are part of current solution: code analysis can run only on files specified in solution and project configuration files. This policy guarantees that all code that is part of a solution is analyzed. Enforce C/C++ Code Analysis (/analyze): Requires that all C or C++ projects be built with the /analyze compiler option to run code analysis before they can be checked in. Enforce Code Analysis for Managed Code: Requires that all managed projects run code analysis and build before they can be checked in. Check Code analysis rule set reference on MSDN What is Rule Set? Rule Set is a group of code analysis rules like the example below where Microsoft.Design is the rule set name where "Do not declare static members on generic types" is the code analysis rule Once you configured the Analysis rule the policy will be enabled for all the team member in this project whenever a team member check-in any source code to the TFSVC the policy section will highlight the Code Analysis policy as below TFS is a very extensible platform so you can simply implement your own custom Code Analysis Check-in policy, check this link for more details http://msdn.microsoft.com/en-us/library/dd492668.aspx but you have to be aware also about compatibility between different TFS versions check http://msdn.microsoft.com/en-us/library/bb907157.aspx Run Code Analysis as part of Team Build With Team Foundation Build (TFBuild), you can create and manage build processes that automatically compile and test your applications, and perform other important functions. Code Analysis can be enabled in the Build Definition file by selecting the correct value for the build process parameter "Perform Code Analysis" Once configure, Kick-off your build definition to queue a new build, Code Analysis will run as part of build workflow and you will be able to see code analysis warning as part of build report Understand the Code Analysis results & learn how to fix them Now after you went through Code Analysis configurations and the different ways of running it, we will go through the Code Analysis result how to understand them and how to resolve them. Code Analysis window in Visual Studio will show all the analysis results based on the rule sets you configured in the project file properties, let's dig deep into what each result item contains: 1 Check ID The unique identifier for the rule. CheckId and Category are used for in-source suppression of a warning.       2 Title The title of warning message       3 Description A description of the problem or suggested fix 4 File Name File name and the line of code number which violate the code analysis rule set 5 Category The code analysis category for this error 6 Warning /Error Depend on how you configure it in the rule set the default is Warning level 7 Action Copy: copy the warning information to the clipboard Create Work Item: If you're connected to Team Foundation Server you can create a work item most probably you may create a Task or Bug and assign it for a developer to fix certain code analysis warning Suppress Message: There are times when you might decide not to fix a code analysis warning. You might decide that resolving the warning requires too much recoding in relation to the probability that the issue will arise in any real-world implementation of your code. Or you might believe that the analysis that is used in the warning is inappropriate for the particular context. You can suppress individual warnings so that they no longer appear in the Code Analysis window. Two options available: In Source inserts a SuppressMessage attribute in the source file above the method that generated the warning. This makes the suppression more discoverable. In Suppression File adds a SuppressMessage attribute to the GlobalSuppressions.cs file of the project. This can make the management of suppressions easier. Note that the SuppressMessage attribute added to GlobalSuppression.cs also targets the method that generated the warning. It does not suppress the warning globally.       Visual Studio makes it very easy to fix Code analysis warning, all you have to do is clicking on the Check Id hyperlink if you are not aware how to fix the warring and you'll be directed to MSDN online or local copy based on the configuration you did while installing Visual Studio and you will find all the information about the warring including how to fix it. Create a Custom Code Analysis Rule Set The Microsoft standard rule sets provide groups of rules that are organized by function and depth. For example, the Microsoft Basic Design Guidelines Rules and the Microsoft Extended Design Guidelines Rules contain rules that focus on usability and maintainability issues, with added emphasis on naming rules in the Extended rule set, you can create and modify a custom rule set to meet specific project needs associated with code analysis. To create a custom rule set, you open one or more standard rule sets in the rule set editor. Create and modify a custom rule set required Visual Studio Premium or Ultimate. You can check How to: Create a Custom Rule Set on MSDN for more details http://msdn.microsoft.com/en-us/library/dd264974.aspx Q & A Visual Studio static code analysis vs. FxCop vs. StyleCpp http://www.excella.com/blog/stylecop-vs-fxcop-difference-between-code-analysis-tools/ Code Analysis for SharePoint Apps and SPDisposeCheck? This post lists some of the rule set you can run specifically for SharePoint applications and how to integrate SPDisposeCheck as well. Code Analysis for SQL Server Database Projects? This post illustrate how to run static code analysis on T-SQL through SSDT ReSharper 8 vs. Visual Studio 2013? This document lists some of the features that are provided by ReSharper 8 but are missing or not as fully implemented in Visual Studio 2013. References A Few Billion Lines of Code Later: Using Static Analysis to Find Bugs in the Real World http://cacm.acm.org/magazines/2010/2/69354-a-few-billion-lines-of-code-later/fulltext What is New in Code Analysis for Visual Studio 2013 http://blogs.msdn.com/b/visualstudioalm/archive/2013/07/03/what-is-new-in-code-analysis-for-visual-studio-2013.aspx Analyze the code quality of Windows Store apps using Visual Studio static code analysis http://msdn.microsoft.com/en-us/library/windows/apps/hh441471.aspx [Hands-on-lab] Using Code Analysis with Visual Studio 2012 to Improve Code Quality http://download.microsoft.com/download/A/9/2/A9253B14-5F23-4BC8-9C7E-F5199DB5F831/Using%20Code%20Analysis%20with%20Visual%20Studio%202012%20to%20Improve%20Code%20Quality.docx Originally posted at "Hosam Kamel| Developer & Platform Evangelist" http://blogs.msdn.com/hkamel

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  • Recording Audio through M-Audio Keystudio

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • No audio with headphones, but audio works with integrated speakers

    - by Pedro
    My speakers work correctly, but when I plug in my headphones, they don't work. I am running Ubuntu 10.04. My audio card is Realtek ALC259 My laptop model is a HP G62t a10em In another thread someone fixed a similar issue (headphones work, speakers not) folowing this: sudo vi /etc/modprobe.d/alsa-base.conf (or some other editor instead of Vi) Append the following at the end of the file: alias snd-card-0 snd-hda-intel options snd-hda-intel model=auto Reboot but it doesnt work for me. Before making and changes to alsa, this was the output: alsamixer gives me this: Things I did: followed this HowTo but now no hardware seems to be present (before, there were 2 items listed): Now, alsamixer gives me this: alsamixer: relocation error: alsamixer: symbol snd_mixer_get_hctl, version ALSA_0.9 not defined in file libasound.so.2 with link time reference I guess there was and error in the alsa-driver install so I began reinstalling it. cd alsa-driver* //this works fine// sudo ./configure --with-cards=hda-intel --with-kernel=/usr/src/linux-headers-$(uname -r) //this works fine// sudo make //this doesn't work. see ouput error below// sudo make install Final lines of sudo make: hpetimer.c: In function ‘snd_hpet_open’: hpetimer.c:41: warning: implicit declaration of function ‘hpet_register’ hpetimer.c:44: warning: implicit declaration of function ‘hpet_control’ hpetimer.c:44: error: expected expression before ‘unsigned’ hpetimer.c: In function ‘snd_hpet_close’: hpetimer.c:51: warning: implicit declaration of function ‘hpet_unregister’ hpetimer.c:52: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: In function ‘hpetimer_init’: hpetimer.c:88: error: ‘EINVAL’ undeclared (first use in this function) hpetimer.c:99: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c:100: error: invalid use of undefined type ‘struct hpet_task’ hpetimer.c: At top level: hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) hpetimer.c:121: error: extra brace group at end of initializer hpetimer.c:121: error: (near initialization for ‘__param_frequency’) hpetimer.c:121: warning: excess elements in struct initializer hpetimer.c:121: warning: (near initialization for ‘__param_frequency’) make[1]: *** [hpetimer.o] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [compile] Error 1 And then sudo make install gives me: rm -f /lib/modules/0.0.0/misc/snd*.*o /lib/modules/0.0.0/misc/persist.o /lib/modules/0.0.0/misc/isapnp.o make[1]: Entering directory `/usr/src/alsa/alsa-driver-1.0.9/acore' mkdir -p /lib/modules/0.0.0/misc cp snd-hpet.o snd-page-alloc.o snd-pcm.o snd-timer.o snd.o /lib/modules/0.0.0/misc cp: cannot stat `snd-hpet.o': No such file or directory cp: cannot stat `snd-page-alloc.o': No such file or directory cp: cannot stat `snd-pcm.o': No such file or directory cp: cannot stat `snd-timer.o': No such file or directory cp: cannot stat `snd.o': No such file or directory make[1]: *** [_modinst__] Error 1 make[1]: Leaving directory `/usr/src/alsa/alsa-driver-1.0.9/acore' make: *** [install-modules] Error 1 [SOLUTION] After screwing it all up, someone mentioned why not trying using the packages in Synaptic - so I did. I have reinstalled the following packages and rebooter: -alsa-hda-realtek-ignore-sku-dkms -alsa-modules-2.6.32-25-generic -alsa-source -alsa-utils -linux-backports-modules-alsa-lucid-generic -linux-backports-modules-alsa-lucid-generic-pae -linux-sound-base -(i think i listed them all) After rebooting, the audio worked, both in speakers and headphones. I have no idea which is the package that made my audio work, but it certainly was one of them. [/SOLUTION]

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  • Extract Audio from a Video File with Pazera Free Audio Extractor

    - by DigitalGeekery
    Have you ever wanted to extract some or all of the audio from a video file?  Today we’ll take a look at Pazera Free Audio Extractor. A simple audio converter that specializes in that very task. Download the Pazera Free Audio Extractor. (See download link below) You’ll need to unzip the download folder, but there is no need to install the application. Simply double-click the AudioExtractor.exe file to run the application. To add your video files to the queue to be converted, click on the Add files  button at the top left. You can add multiple files to the queue and convert them all at one time. Browse for your video file, and click Open.   Your video will be added to the Queue for processing.   Under Output directory you can choose to output to a folder of your choice. Outputting to the same folder as the input folder is the default.   Pazera Free Audio Extractor includes pre-configured profiles that will simplify the process of choosing conversion settings. To load a profile, choose one from the Profile drop down list and then click the Load button. You can choose to output to MP3, AAC, AC3, WMA, FLAC, OGG or WAV file format.   You will see the profile update the Audio settings in the panels at the lower left of the application. If you wish, you may also select your own custom settings. Advanced Settings The Advanced settings can be used if you want to extract only a portion of the the audio, such as a clip of dialog or a song from a movie. To extract only a portion of the audio, set the start time by selecting the Start time offset check box, then entering the time in the video clip where the audio begins. To set the end time, begin by selecting the Duration check box. Now, you can either select the Duration radio button and enter the amount of time for which you would like to extract the audio, or you can select the End time offset radio button and enter the time in the video clip where the audio ends. When you are ready to convert, click the CONVERT button on the menu at the top of the screen.   An output box will open and display the conversion progress. When finished, click Close.   Now you are ready to enjoy your audio clip. Pazera Free Audio Extractor is a basic audio tool that is easy enough for everyone to use. It runs on Windows only and supports most common video formats including AVI, FLV, MP4, MPG, MOV, 3GP, and WMV. Download Free Audio Extractor 1.3 Similar Articles Productive Geek Tips Eufony Free Audio Player – Resource Gentle Audio PlayerConvert .3GP and .3G2 Files to AVI / MPEG for FreeTurn Off Auto-Play of Audio and Video CDs and DVDs in UbuntuHow to Make/Edit a movie with Windows Movie Maker in Windows VistaEasily Change Audio File Formats with XRECODE TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Use Printflush to Solve Printing Problems Icelandic Volcano Webcams Open Multiple Links At One Go NachoFoto Searches Images in Real-time Office 2010 Product Guides Google Maps Place marks – Pizza, Guns or Strip Clubs

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  • guvcview recording video and audio out of synchronisation in Ubuntu 10.10

    - by SIJAR
    I finally got Guvcview, a great software for Logitech webcam and it does all the stuff that one wants out of it. But I'm not satisfy with the video recording, video and audio out of synchronisation also video seems to be in slow motion. Please help so that I can tweak in and get a good video recording with the webcam. Below is the log of Guvcview ------------------------------------------------------------------------------- guvcview 1.4.1 video_device: /dev/video0 vid_sleep: 0 cap_meth: 1 resolution: 640 x 480 windowsize: 1024 x 715 vert pane: 578 spin behavior: 0 mode: mjpg fps: 1/25 Display Fps: 0 bpp: 0 hwaccel: 1 avi_format: 4 sound: 1 sound Device: 4 sound samp rate: 0 sound Channels: 0 Sound delay: 0 nanosec Sound Format: 85 Pan Step: 2 degrees Tilt Step: 2 degrees Video Filter Flags: 0 image inc: 0 profile(default):/home/sijar/default.gpfl starting portaudio... bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) bt_audio_service_open: connect() failed: Connection refused (111) Cannot connect to server socket err = No such file or directory Cannot connect to server socket jack server is not running or cannot be started language catalog= dir:/usr/share/locale type:UTF-8 lang:en_US.utf8 cat:guvcview.mo mjpg: setting format to 1196444237 capture method = 1 video device: /dev/video0 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! /dev/video0 - device 1 libv4lconvert: warning more framesizes then I can handle! libv4lconvert: warning more framesizes then I can handle! Init. UVC Camera (046d:0825) (location: usb-0000:00:1d.7-5) { pixelformat = 'YUYV', description = 'YUV 4:2:2 (YUYV)' } { discrete: width = 640, height = 480 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 160, height = 120 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 176, height = 144 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 176 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 320, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 352, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 432, height = 240 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 544, height = 288 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, { discrete: width = 640, height = 360 } Time interval between frame: 1/30, 1/25, 1/20, 1/15, 1/10, 1/5, ... repeats a couple of times ... vid:046d pid:0825 driver:uvcvideo Adding control for Pan (relative) UVCIOC_CTRL_ADD - Error: Operation not permitted checking format: 1196444237 VIDIOC_G_COMP:: Invalid argument compression control not supported fps is set to 1/25 drawing controls control[0]: 0x980900 Brightness, 0:255:1, default 128 control[0]: 0x980901 Contrast, 0:255:1, default 32 control[0]: 0x980902 Saturation, 0:255:1, default 32 control[0]: 0x98090c White Balance Temperature, Auto, 0:1:1, default 1 control[0]: 0x980913 Gain, 0:255:1, default 0 control[0]: 0x980918 Power Line Frequency, 0:2:1, default 2 control[0]: 0x98091a White Balance Temperature, 0:10000:10, default 4000 control[0]: 0x98091b Sharpness, 0:255:1, default 24 control[0]: 0x98091c Backlight Compensation, 0:1:1, default 1 control[0]: 0x9a0901 Exposure, Auto, 0:3:1, default 3 control[0]: 0x9a0902 Exposure (Absolute), 1:10000:1, default 166 control[0]: 0x9a0903 Exposure, Auto Priority, 0:1:1, default 0 resolutions of format(2) = 19 frame rates of 1º resolution=6 Def. Res: 0 numb. fps:6 --------------------------------------- device #0 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 (hw:0,0) Host API = ALSA Max inputs = 2, Max outputs = 2 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #1 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC ADC (hw:0,1) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #2 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - MIC2 ADC (hw:0,2) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #3 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - ADC2 (hw:0,3) Host API = ALSA Max inputs = 2, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #4 Name = Intel 82801DB-ICH4: Intel 82801DB-ICH4 - IEC958 (hw:0,4) Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #5 Name = USB Device 0x46d:0x825: USB Audio (hw:1,0) Host API = ALSA Max inputs = 1, Max outputs = 0 Def. low input latency = 0.011 Def. low output latency = -1.000 Def. high input latency = 0.043 Def. high output latency = -1.000 Def. sample rate = 48000.00 --------------------------------------- device #6 Name = front Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.012 Def. high input latency = -1.000 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #7 Name = iec958 Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #8 Name = spdif Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.011 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #9 Name = pulse Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 --------------------------------------- device #10 Name = dmix Host API = ALSA Max inputs = 0, Max outputs = 2 Def. low input latency = -1.000 Def. low output latency = 0.043 Def. high input latency = -1.000 Def. high output latency = 0.043 Def. sample rate = 48000.00 --------------------------------------- device #11 [ Default Input, Default Output ] Name = default Host API = ALSA Max inputs = 32, Max outputs = 32 Def. low input latency = 0.012 Def. low output latency = 0.012 Def. high input latency = 0.046 Def. high output latency = 0.046 Def. sample rate = 44100.00 ---------------------------------------------- SampleRate:0 Channels:0 Video driver: x11 A window manager is available VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_CTRL for user class controls control(0x0098091a) "White Balance Temperature" failed to set (error -1) VIDIOC_S_EXT_CTRLS for multiple controls failed (error -1) using VIDIOC_S_EXT_CTRLS on single controls for class: 0x009a0000 control(0x009a0902) "Exposure (Absolute)" failed to set (error -1) Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25371756K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cbd8b0]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cbd8b0]profile Baseline, level 3.0 [libx264 @ 0x8cbd8b0]non-strictly-monotonic PTS shift sound by -9 ms shift sound by -9 ms shift sound by -9 ms AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... AUDIO: droping audio data (/home/sijar/Videos/Webcam) 25371748K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... Cap Video toggled: 0 Shuting Down IO Thread AUDIO: droping audio data stop= 4426644744000 start=4416533023000 VIDEO: 146 frames in 10111.000000 ms = 14.439719 fps Stoping audio stream Closing audio stream... close avi Last message repeated 145 times [libx264 @ 0x8cbd8b0]frame I:2 Avg QP:14.10 size: 24492 [libx264 @ 0x8cbd8b0]frame P:103 Avg QP:16.06 size: 20715 [libx264 @ 0x8cbd8b0]mb I I16..4: 48.4% 0.0% 51.6% [libx264 @ 0x8cbd8b0]mb P I16..4: 57.5% 0.0% 0.0% P16..4: 40.2% 0.0% 0.0% 0.0% 0.0% skip: 2.3% [libx264 @ 0x8cbd8b0]final ratefactor: 62.05 [libx264 @ 0x8cbd8b0]coded y,uvDC,uvAC intra: 79.7% 92.2% 68.4% inter: 62.4% 87.5% 48.0% [libx264 @ 0x8cbd8b0]i16 v,h,dc,p: 23% 17% 41% 19% [libx264 @ 0x8cbd8b0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 30% 24% 26% 2% 5% 3% 3% 3% 4% [libx264 @ 0x8cbd8b0]i8c dc,h,v,p: 53% 20% 23% 4% [libx264 @ 0x8cbd8b0]ref P L0: 63.0% 37.0% [libx264 @ 0x8cbd8b0]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Cap Video toggled: 1 (/home/sijar/Videos/Webcam) 25379744K bytes free on a total of 39908968K (used: 36 %) treshold=51200K using audio codec: 0x0055 Audio frame size is 1152 samples for selected codec IO thread started...OK [libx264 @ 0x8cfba20]using cpu capabilities: MMX2 SSE2 Cache64 [libx264 @ 0x8cfba20]profile Baseline, level 3.0 [libx264 @ 0x8cfba20]non-strictly-monotonic PTS shift sound by -236 ms shift sound by -236 ms shift sound by -236 ms (/home/sijar/Videos/Webcam) 25377044K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25373408K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25370696K bytes free on a total of 39908968K (used: 36 %) treshold=51200K AUDIO: droping audio data AUDIO: droping audio data AUDIO: droping audio data ... repeats a couple of times ... (/home/sijar/Videos/Webcam) 25367680K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25364052K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25360312K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25356628K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25352908K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25349316K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25345552K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25341828K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25338092K bytes free on a total of 39908968K (used: 36 %) treshold=51200K (/home/sijar/Videos/Webcam) 25334412K bytes free on a total of 39908968K (used: 36 %) treshold=51200K Cap Video toggled: 0 Shuting Down IO Thread stop= 4708817235000 start=4578624714000 VIDEO: 1604 frames in 130192.000000 ms = 12.320265 fps Stoping audio stream Closing audio stream... close avi Last message repeated 1603 times [libx264 @ 0x8cfba20]frame I:16 Avg QP:14.78 size: 42627 [libx264 @ 0x8cfba20]frame P:1547 Avg QP:16.44 size: 28599 [libx264 @ 0x8cfba20]mb I I16..4: 21.6% 0.0% 78.4% [libx264 @ 0x8cfba20]mb P I16..4: 28.1% 0.0% 0.0% P16..4: 70.5% 0.0% 0.0% 0.0% 0.0% skip: 1.4% [libx264 @ 0x8cfba20]final ratefactor: 88.17 [libx264 @ 0x8cfba20]coded y,uvDC,uvAC intra: 74.4% 95.8% 83.2% inter: 75.2% 94.6% 69.2% [libx264 @ 0x8cfba20]i16 v,h,dc,p: 27% 17% 40% 16% [libx264 @ 0x8cfba20]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 25% 25% 21% 3% 6% 4% 5% 4% 7% [libx264 @ 0x8cfba20]i8c dc,h,v,p: 61% 18% 18% 4% [libx264 @ 0x8cfba20]ref P L0: 64.0% 36.0% [libx264 @ 0x8cfba20]kb/s:-0.00 total frames encoded: 0 total audio frames encoded: 0 IO thread finished...OK IO Thread finished enabling controls Shuting Down Thread Thread terminated... cleaning Thread allocations: 100% SDL Quit Video Thread finished write /home/sijar/.guvcviewrc OK free audio mutex closed v4l2 strutures free controls free controls - vidState cleaned allocations - 100% Closing portaudio ...OK Closing GTK... OK

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • How to create a virtual audio device and stream audio input with it

    - by Steven Rosato
    Is it possible to create another audio device and redirect only wanted input streams to it? Here's my concrete problem: I am broadcasting a game via XFire and it uses the Windows audio device to capture any audio I receive. As I am broadcasting, other users who watch the video stream are communicating with me over Skype, and they hear themselves back within the video stream and it is entirely logical since I am broadcasting the audio I hear. What I want to do is create another audio device within Windows and redirect (pipe) ONLY the audio input from that game and not the input reveived from Skype. I would then tell XFire to use that newly created "virtual" audio device to broadcast and therefore my partners won't hear themselves back. Is there any software that can do that or can it be achieved natively with Windows? (I am under Windows 7).

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  • what is the best way to stream a audio file to website users/listners

    - by Naveen Chamikara Gamage
    I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage.. As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one? I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality.. I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers? if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use? I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files? Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files? Any known best softwares for converting audio files while keeping quality in a good level? Note** - I know that I will not need complex requirements at the beginning of the site but I wanted to what are the best ways like they are using for soundcloud.com

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  • Combine Multiple Audio Files into a single higher-quality audio File

    - by namenlos
    BACKGROUND My team gave a demo to a large audience - we recorded the audio of the demo in multiple locations in the room (3) the audio was recorded using cheap laptop microphones I was not involved in the recording of the audio or the demo Both audio files suck in some form the first one is of a recording near the speaker - which clearly gets his voice but the the audience is audience is muffled - also this one is slightly noisy The second recording was done in the middle of the audience - it gets the audience questions clearly but actually gets the speaker rather sometimes well and sometimes poorly (not all the speakers spoke loudly enough to be heard) MY QUESTION Is there any techinque or software which can be used to merge these audio files in such a way that the best qualities of each are preserved. I am NOT asking now to simply merge them together in one track - I've already done that in Audacity and it is certainly better - what I am looking for could be considered closer to how HDR images are created - multiple exposures combined into an enhanced new version which is not simply an average of the inputs. NOTE Am not an "Audio" guy - just a normal user

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  • Two audio streams - headphones and speakers

    - by Sylvester
    What I want (this is probably hard for most to answer, as this is a very unique setup) is to have two different streams (this means audio splitter is not an option, as it will still only be one stream) of audio - one through the headphones and one through the main speakers. I can do the audio rerouting using virtual audio cables, however the problem is this: i cannot get both headphones AND speakers to play even just one stream, let alone two seperate ones. using "split front and back audio into seperate streams is not an option, as the actual MB F_PANEL is faulty (nothing to do with the case front panel, just so you know. that works fine) So, first things first. I need it to recognise the headphones as a seperate audio device so that Virtual Audio Cables will detect it and allow me to route the necessary audio to the headphones only. I also need to be able have sound play through speakers and headphones together what i want to achieve overall, is this: have the ENTIRE computer's sounds picked up by VAC, and stream them to Line1. then have Line1 stream to the headphones. that way whatever's being streamed is heard through the headphones, while the entire system sounds (including those not streamed) are played through speakers.

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  • Why are the analoge stereo input and output of my M-Audio 24/96 soundcard not available to me in Ubu

    - by user37968
    I have installed Lucid on an old Mac PowerPC G4 desktop with a M-Audio Audiophile 24/96 soundcard. The only inputs and outputs I can select in the audio preferences are digital ones for the digital input and output. "lspci -v" shows the card as so: 0001:10:13.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. Device d634 Flags: bus master, medium devsel, latency 16, IRQ 53 I/O ports at 0440 [size=32] I/O ports at 04b0 [size=16] I/O ports at 04a0 [size=16] I/O ports at 0400 [size=64] Capabilities: <access denied> Kernel driver in use: ICE1712 Kernel modules: snd-ice1712 "cat /proc/asound/cards" as so: 0 [Tumbler ]: PMac Tumbler - PowerMac Tumbler PowerMac Tumbler (Dev 21) Sub-frame 0 1 [M2496 ]: ICE1712 - M Audio Audiophile 24/96 M Audio Audiophile 24/96 at 0x440, irq 53 "aplay -L" shows these as listed: pulse Playback/recording through the PulseAudio sound server front:CARD=Tumbler,DEV=0 PowerMac Tumbler, PowerMac Tumbler Front speakers front:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi Front speakers surround40:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.0 Surround output to Front and Rear speakers surround41:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi IEC958 (S/PDIF) Digital Audio Output I believe it is a problem with detecting the analogue input/output. Sometimes I can get sound from the device but it is a sheet of white noise and tinkering makes it go away again I don't know if that is a separate problem or if it is linked to not being able to see the analogue input/outputs in the sound preferences. Any help would be greatly appreciated As for the white noise I have installed the Envy24 control panel and spend lots of time playing with the settings but when I can get the white noise I can never get it to an quality where I can actually hear what is being played. The internal speaker plays audio fine and plugging in a NI Audio 4DJ via usb also plays sound, although with some static but I believe that is due to an underpowered usb2 pci expansion card not being able to get enough electricity to the device. Alternatively I have seen other people with problems with this device so it may be a bug in the driver but that is another matter. I would like to get the M-Audio card working so I can begin to enjoy my music once again. As a note, I do not currently have any audio equipment capable of sending or receiving audio via the digital inputs and output so I can not check if they are working. The sound preferences show a wide range of digital in and out options with various surround sound options but no analogue ins and outs.

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  • Why are the analoge stereo input and output of my M-Audio 24/96 soundcard not available to me in Ubuntu Lucid

    - by MIDoubleKO
    I have installed Lucid on an old Mac PowerPC G4 desktop with a M-Audio Audiophile 24/96 soundcard. The only inputs and outputs I can select in the audio preferences are digital ones for the digital input and output. "lspci -v" shows the card as so: 0001:10:13.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. Device d634 Flags: bus master, medium devsel, latency 16, IRQ 53 I/O ports at 0440 [size=32] I/O ports at 04b0 [size=16] I/O ports at 04a0 [size=16] I/O ports at 0400 [size=64] Capabilities: <access denied> Kernel driver in use: ICE1712 Kernel modules: snd-ice1712 "cat /proc/asound/cards" as so: 0 [Tumbler ]: PMac Tumbler - PowerMac Tumbler PowerMac Tumbler (Dev 21) Sub-frame 0 1 [M2496 ]: ICE1712 - M Audio Audiophile 24/96 M Audio Audiophile 24/96 at 0x440, irq 53 "aplay -L" shows these as listed: pulse Playback/recording through the PulseAudio sound server front:CARD=Tumbler,DEV=0 PowerMac Tumbler, PowerMac Tumbler Front speakers front:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi Front speakers surround40:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.0 Surround output to Front and Rear speakers surround41:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=M2496,DEV=0 M Audio Audiophile 24/96, ICE1712 multi IEC958 (S/PDIF) Digital Audio Output I believe it is a problem with detecting the analogue input/output. Sometimes I can get sound from the device but it is a sheet of white noise and tinkering makes it go away again I don't know if that is a separate problem or if it is linked to not being able to see the analogue input/outputs in the sound preferences. Any help would be greatly appreciated As for the white noise I have installed the Envy24 control panel and spend lots of time playing with the settings but when I can get the white noise I can never get it to an quality where I can actually hear what is being played. The internal speaker plays audio fine and plugging in a NI Audio 4DJ via usb also plays sound, although with some static but I believe that is due to an underpowered usb2 pci expansion card not being able to get enough electricity to the device. Alternatively I have seen other people with problems with this device so it may be a bug in the driver but that is another matter. I would like to get the M-Audio card working so I can begin to enjoy my music once again. As a note, I do not currently have any audio equipment capable of sending or receiving audio via the digital inputs and output so I can not check if they are working. The sound preferences show a wide range of digital in and out options with various surround sound options but no analogue ins and outs.

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  • Audio Static/Interference regardless of audio interface?

    - by Tom
    I currently am running a media center/server on a Lubuntu machine. The machine specs: Core 2 Duo Extreme EVGA SLI 680i MotherBoard 2 GB DDR2 Ram 3 Hard Drives no raid - WD Caviar Black, Green, and Samsung Spinpoint Galaxy GTX 220 1GB External USB Creative XI-FI Extreme Card 550W Power Supply This machine is hooked up through an optical cable to an ONKYO HTR340 Receiver through the XIFI card. Whenever I play any audio regardless if it is through XBMC, the default audio player, a flash video, etc, I get a horrible static sound that randomly gets louder. Here is a video of the sound: http://www.youtube.com/watch?v=SqKQkxYRVA4 This static comes in randomly, sometimes going away for short periods, but eventually always comes back. So far I have tried everything I could think of: Reinstalling OS Installing/upgrading/repairing PulseAudio/Alsa Installing alternate OSes, straight Ubuntu, Lubuntu, Xubuntu, Arch, Mint, Windows 7 Switching audio from the external card to internal Optical, audio out through HDMI, audio out through headphones Different ports on receiver (my main desktop sounds fine on the same sound system) Different optical cables Unplugging everything unnecessary from the motherboard (1 HD, 1 Stick of Ram, 1 Keyboard) Swapping out ram Swapping out the motherboard Replacing the Graphics Card (was replaced due to fan being noisy, not specifically for this problem) Different harddrives Swapping power supply Disabling onboard audio Pretty much everything short of swapping the CPU. I haven't been able to narrow down the problem and it is getting frustrating. Is it possible that the CPU is faulty and might cause a problem such as this, or that the PC case is shorting out the motherboard? Any kind of suggestions will be appreciated.

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • No digital audio output with Asus Xonar DG

    - by Lunatik
    I've purchased an Asus Xonar DG as replacement for faulty onboard audio in a Medion 8822 as it has an optical output which is all I really need to feed my HTPC. I uninstalled the previous drivers/devices, switched the PC off, inserted the Asus card, powered up, disabled the onboard audio in the BIOS, then installed the driver that came on the CD (same version as on Asus' website as of today) and everything went perfectly - no errors. I set the audio devices up in Windows and in the Asus utility (SPDIF enabled, 6-ch audio) as I would expect to see them work, but the only thing is I have no digital audio from test tones within Windows/the Asus utility, PCM audio or Dolby Digital from DVD. Analogue audio is fine. I've uninstalled things and reinstalled a couple of times now, as well as trying almost all combinations of analogue/digital outputs but can't get it sorted. Does anyone have any tips on how to get this working? This card has just been released so there isn't much out there to go on. Notes: The light on the toslink port is lit. OS is Vista 32-bit SP2 and all up to date, pretty much a fresh install with almost no 3rd party applications installed This page seems to suggest that a digital output device in Windows is not needed with Xonar cards as it was with the previous Realtek so I have it set to Analog. The only other output device is S/PDIF pass-thru

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  • Digital audio input on Macbook?

    - by Ken
    I have: a Macbook (not Pro), don't know the exact model but it's a Core 2 Duo 2.0GHz and probably what Wikipedia calls the "Late 2006" or "Mid 2007" model a DVD player, region-free, that has "Coax and TosLink optical digital audio outputs" I want to make an MP3 of the audio track of some DVDs (for learning a new language), and I can't use the Macbook's built-in DVD drive because it's a different region (ugh!). I'm sure I can connect the DVD player to the Macbook with an analog audio cable. However, if it's possible I'd prefer to keep the signal digital. I'm not even positive if my old Macbook has digital audio in, and if so what I need to connect to it. (I've done plenty of home audio geeking, but always in analog!) Will a "Toslink cable" plus a "Toslink Female to Mini-Plug Male Adapter" (found on Amazon) let me connect my things together? It looks like the pieces will fit but I'd like to hear someone confidently knowledgeable on the matter before I buy something. Thanks!

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  • Configure TV Capture card to not use external audio jack for TV audio output

    - by Adam D.
    I had this working with MythTV on Ubuntu 9.1. Then a power surge killed the motherboard. After replacing the motherboard, ram and cpu, the card does not produce any audio except through the output jack on the back of the card. I do not want to use a cable to go from the back of the card to the audio in on the built in sound card of the new mother board. FYI, the old motherboard did not have an on-board sound card. There was a separate audio card installed. There's some configuration that has to be done to have it work the same way again. I just have no idea where to start. This is regarding wintv hauppauge mythtv linux ubuntu 9.10 audio

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  • White Paper on Analysis Services Tabular Large-scale Solution #ssas #tabular

    - by Marco Russo (SQLBI)
    Since the first beta of Analysis Services 2012, I worked with many companies designing and implementing solutions based on Analysis Services Tabular. I am glad that Microsoft published a white paper about a case-study using one of these scenarios: An Analysis Services Case Study: Using Tabular Models in a Large-scale Commercial Solution. Alberto Ferrari is the author of the white paper and many people contributed to it. The final result is a very technical document based on a case study, which provides a level of detail that I don’t see often in other case studies (which are usually more marketing-oriented). This white paper has the following structure: Requirements (data model, capacity planning, client tool) Options considered (SQL Server Columnstore Indexes, SSAS Multidimensional, SSAS Tabular) Data Model optimizations (memory compression, query performance, scalability) Partitioning and Processing strategy for near real-time latency Hardware selection (NUMA analysis, Azure VM tests) Scalability tests (estimation of maximum users per node) If you are in charge of evaluating Tabular as analytical engine, or if you have to design your solution based on Tabular, this white paper is a must read. But if you just want to increase your knowledge of Analysis Services, you will find a lot of useful technical information. That said, my favorite quote of the document is the following one, funny but true: […] After several trials, the clear winner was a video gaming machine that one guy on the team used at home. That computer outperformed any available server, running twice as fast as the server-class machines we had in house. At that point, it was clear that the criteria for choosing the server would have to be expanded a bit, simply because it would have been impossible to convince the boss to build a cluster of gaming machines and trust it to serve our customers.  But, honestly, if a business has the flexibility to buy gaming machines (assuming the machines can handle capacity) – do this. Owen Graupman, inContact I want to write a longer discussion about how companies are adopting Tabular in scenarios where it is the hidden engine of a more complex solution (and not the classical “BI system”), because it is more frequent than you might expect (and has several advantages over many alternative approaches).

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  • USB Audio device not selectable in settings after Camtasia Studio

    - by DJDavid98
    I tried to use Camtasia Studio 7 Recorder today, and when I selected the "Record Microphone and System audio" option, the recorder crashed, and then my sound settings icon disappeared. If I tried to open the Camtasia Recorder's Options or tried to select the option again, the program crashed. After holding the power button to force restart (so My settings wouldn't save, theoretically) the audio device was still not showing up in the sound settings, although it still is displayed under the "Hardware" tab, and is Functional. Or at least that's what windows thinks. I tried uninstalling Camtasia, and no sucess either. The USB audio device is labeled "C-Media USB Headphone Set". OS: Windows XP SP3 hu-HU Is there any way I could make it work again, like registry editing or something?

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  • Basket Analysis with #dax in #powerpivot and #ssas #tabular

    - by Marco Russo (SQLBI)
    A few days ago I published a new article on DAX Patterns web site describing how to implement Basket Analysis in DAX. This topic is a very classical one and is also covered in the many-to-many revolution white paper. It has been also discussed in several blog posts, listed here in historical order: Simple Basket Analysis in DAX by Chris Webb PowerPivot, basket analysis and the hidden many to many by Alberto Ferrari Applied Basket Analysis in Power Pivot using DAX by Gerhard Brueckl As usual, in DAX Patterns we try to present the required DAX formulas in a way that is easy to adapt to specific models. We also try to show a good implementation from a performance point of view. Further optimizations are always possible in DAX. However, in order to keep the model simple to adapt in different scenarios, we avoid presenting optimizations that would require particular assumptions or restrictions on the data model. I hope you will find the Basket Analysis pattern useful. Even if you do not need it today, reading the DAX formula is a good exercise to check your knowledge of evaluation contexts in DAX. For example, describing how does it work the following expression is not a trivial task! [Orders with Both Products] := CALCULATE (     DISTINCTCOUNT ( Sales[SalesOrderNumber] ),     CALCULATETABLE (         SUMMARIZE ( Sales, Sales[SalesOrderNumber] ),         ALL ( Product ),         USERELATIONSHIP ( Sales[ProductCode], 'Filter Product'[Filter ProductCode] )     ) ) The good news is that you can use the patterns even if you do not really understand all the details of the DAX formulas you are using! Any feedback on this new pattern is very welcome.

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  • Prevent audio recording applications from picking up non-microphone audio output

    - by hheimbuerger
    When talking to people over Mumble/Teamspeak/Ventrilo, I have a few who are broadcasting all of their audio output (e.g. music playing in the background, game sounds, etc.) whenever they are sending -- in addition to the microphone signal. I don't have this problem myself, so it's hard for me to troubleshoot remotely, but I'm interested in collecting solutions for this problem for Windows XP, Vista and 7, so that I can link them to this question. Links to other related SU questions are highly welcome as well, but I couldn't find any. Most people seem to try to either get microphone to feed back into the audio output or turn exactly that off, which is different from this problem. I'm talking about the opposite problem, the output being fed into the input.

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  • Digital audio does not work on MacBook Pro

    - by mathk
    I have a MacBook Pro (8,2). Using a TOSLINK cable I have no digital output. Oddly enough, sometime I can hear a glitch when I plug in the cable or when I give it a gentle wiggle. My guess is that the output is not correctly detecting that I have a digital link. So is there a way to force digital audio output on a MacBook Pro? Some say that in the Audio MIDI Setup there is an option but I can't find it. I am running OS X 10.7.5.

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