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  • How to release audio properly? (AVAudioPlayer)

    - by Aluminum
    Hello everyone! I need help with my iOS application ^^,. I want to know if I'm releasing AVAudioPlayer correctly. MyViewController.h #import <UIKit/UIKit.h> @interface MyViewController : UIViewController { NSString *Path; } - (IBAction)Playsound; @end MyViewController.m #import <AVFoundation/AVAudioPlayer.h> #import "MyViewController.h" @implementation MyViewController AVAudioPlayer *Media; - (IBAction)Playsound { Path = [[NSBundle mainBundle] pathForResource:@"Sound" ofType:@"wav"]; Media = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:Path] error:NULL]; [Media play]; } - (void)dealloc { [Media release]; [super viewDidUnload]; } @end

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  • Non intrusive notification without audio?

    - by acidzombie24
    i have a C# app that registers a protocol. When you click BLAH://djfhgjfdghjkd in a browser it launches my app. However you can click multiple links and each link is a note added into the app. How can i inform the user that he did fully click the link? Right now i have a console app showing up for 1sec (basically pops up and goes away as fast as possible) which felt better then a hidden console since you are unsure if it went through. The 1 second takes a lot of time when you are trying to rapidly click many notes/links and the console gets in the way. What can i do that is noticeable? I'm thinking have a box that comes up (and is semi transparent) but the click passes through it. Maybe there is a better way? Also i wouldnt know where to start with transparent windows or pass through clicks

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  • absolute audio synchronization

    - by user1780526
    I would like to synchronize my computer with an external camcorder recording so that I can know exactly (to the millisecond) when certain recored events happen with respect to other sensors logged by the computer. One idea is to playback short sound pulses or chirps every second from the computer that get picked up by the microphone on the camcorder. But the accuracy of a simple cron job playing a sound clip is not precise enough. I was thinking of using something like gstreamer, but how does one get it to playback a clip at precisely a certain time according to the system clock?

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  • Vmware software installation error

    - by Perry
    I am trying to install Vmware software, but I am facing the following error: Selecting previously unselected package vmware-view-client:i386. (Reading database ... 239594 files and directories currently installed.) Unpacking vmware-view-client:i386 (from .../vmware-view-client_2.1.0-0ubuntu0.12.04_i386.deb) ... Processing triggers for desktop-file-utils ... Processing triggers for bamfdaemon ... Rebuilding /usr/share/applications/bamf.index... Processing triggers for gnome-menus ... Setting up icaclient:i386 (12.1.0) ... dpkg: error processing icaclient:i386 (--configure): subprocess installed post-installation script returned error exit status 2 Setting up vmware-view-client:i386 (2.1.0-0ubuntu0.12.04) ... Processing triggers for libc-bin ... ldconfig deferred processing now taking place Errors were encountered while processing: icaclient:i386 E: Sub-process /usr/bin/dpkg returned an error code (1) A package failed to install. Trying to recover: Setting up icaclient:i386 (12.1.0) ... dpkg: error processing icaclient:i386 (--configure): subprocess installed post-installation script returned error exit status 2 Errors were encountered while processing: icaclient:i386 Any suggestions on how to fix this issue? Thanks in advance

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  • Error when I try to installth desktop integration features for Openoffice

    - by PENG TENG
    peng@peng-ThinkPad-SL410:~$ cd '/home/peng/Downloads/en-US/DEBS/desktop-integration' peng@peng-ThinkPad-SL410:~/Downloads/en-US/DEBS/desktop-integration$ sudo dpkg -i *.deb (Reading database ... 357248 files and directories currently installed.) Unpacking openoffice.org-debian-menus (from openoffice.org3.4-debian-menus_3.4-9593_all.deb) ... dpkg: error processing openoffice.org3.4-debian-menus_3.4-9593_all.deb (--install): trying to overwrite '/usr/bin/soffice', which is also in package libreoffice-common 1:3.6.2~rc2-0ubuntu3 /usr/bin/gtk-update-icon-cache gtk-update-icon-cache: Cache file created successfully. /usr/bin/gtk-update-icon-cache gtk-update-icon-cache: Cache file created successfully. Processing triggers for menu ... Processing triggers for hicolor-icon-theme ... Processing triggers for gnome-icon-theme ... Processing triggers for shared-mime-info ... Unknown media type in type 'all/all' Unknown media type in type 'all/allfiles' Unknown media type in type 'uri/mms' Unknown media type in type 'uri/mmst' Unknown media type in type 'uri/mmsu' Unknown media type in type 'uri/pnm' Unknown media type in type 'uri/rtspt' Unknown media type in type 'uri/rtspu' Processing triggers for bamfdaemon ... Rebuilding /usr/share/applications/bamf.index... Processing triggers for desktop-file-utils ... Processing triggers for gnome-menus ... Errors were encountered while processing: openoffice.org3.4-debian-menus_3.4-9593_all.deb Can anyone solve the problem?

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  • aplay -l says no soundcards found; alsaconf says no supported cords; yet /proc/asound contains cards

    - by nimasmi
    I am trying to get HDMI output using a Gainward Nvidia 210 512 MB on Ubuntu 10.04 Lucid Lynx. I have upgraded alsa-driver, alsa-lib and alsa-utils to 1.0.24 by building from source, thanks to this blog post. Some relevant output... user@box:~$ lspci | grep Audio 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 01:09.0 Multimedia video controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder (rev 05) 01:09.2 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [MPEG Port] (rev 05) 01:09.4 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [IR Port] (rev 05) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) user@box:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Sep 15 2012 for kernel 2.6.32-42-generic (SMP). user@box:~$ ls /proc/asound` card0 cards hwdep NVidia oss seq version card1 devices modules NVidia_1 pcm timers user@box:~$ aplay -l aplay: device_list:240: no soundcards found... user@box:~$ sudo /sbin/alsa-utils start * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1403: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... amixer: Invalid command! ...done. Any help appreciated. PS my video card is connected only through the PCI-E slot. I assume there is no extra audio connection required.

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  • NAudio demos not working anymore

    - by Kurru
    I just tried to run the NAudio demos and I'm getting a weird error: System.BadImageFormatException: Could not load file or a ssembly 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' or one o f its dependencies. An attempt was made to load a program with an incorrect form at. File name: 'NAudio, Version=1.3.8.0, Culture=neutral, PublicKeyToken=null' at NAudioWpfDemo.AudioGraph..ctor() at NAudioWpfDemo.ControlPanelViewModel..ctor(IWaveFormRenderer waveFormRender er, SpectrumAnalyser analyzer) in C:\Users\Admin\Downloads\NAudio-1.3\NAudio-1-3 \Source Code\NAudioWpfDemo\ControlPanelViewModel.cs:line 23 at NAudioWpfDemo.MainWindow..ctor() in C:\Users\Admin\Downloads\NAudio-1.3\NA udio-1-3\Source Code\NAudioWpfDemo\MainWindow.xaml.cs:line 15 WRN: Assembly binding logging is turned OFF. To enable assembly bind failure logging, set the registry value [HKLM\Software\M icrosoft\Fusion!EnableLog] (DWORD) to 1. Note: There is some performance penalty associated with assembly bind failure lo gging. To turn this feature off, remove the registry value [HKLM\Software\Microsoft\Fus ion!EnableLog]. Since the last time I used NAudio demos I have changed from 32bit Windows XP to 64bit Windows 7. Would this cause this issue? Its very annoying as I was about to try my hand at audio in C# again

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  • Eigenvector computation using OpenCV

    - by Andriyev
    Hi I have this matrix A, representing similarities of pixel intensities of an image. For example: Consider a 10 x 10 image. Matrix A in this case would be of dimension 100 x 100, and element A(i,j) would have a value in the range 0 to 1, representing the similarity of pixel i to j in terms of intensity. I am using OpenCV for image processing and the development environment is C on Linux. Objective is to compute the Eigenvectors of matrix A and I have used the following approach: static CvMat mat, *eigenVec, *eigenVal; static double A[100][100]={}, Ain1D[10000]={}; int cnt=0; //Converting matrix A into a one dimensional array //Reason: That is how cvMat requires it for(i = 0;i < affnDim;i++){ for(j = 0;j < affnDim;j++){ Ain1D[cnt++] = A[i][j]; } } mat = cvMat(100, 100, CV_32FC1, Ain1D); cvEigenVV(&mat, eigenVec, eigenVal, 1e-300); for(i=0;i < 100;i++){ val1 = cvmGet(eigenVal,i,0); //Fetching Eigen Value for(j=0;j < 100;j++){ matX[i][j] = cvmGet(eigenVec,i,j); //Fetching each component of Eigenvector i } } Problem: After execution I get nearly all components of all the Eigenvectors to be zero. I tried different images and also tried populating A with random values between 0 and 1, but the same result. Few of the top eigenvalues returned look like the following: 9805401476911479666115491135488.000000 -9805401476911479666115491135488.000000 -89222871725331592641813413888.000000 89222862280598626902522986496.000000 5255391142666987110400.000000 I am now thinking on the lines of using cvSVD() which performs singular value decomposition of real floating-point matrix and might yield me the eigenvectors. But before that I thought of asking it here. Is there anything absurd in my current approach? Am I using the right API i.e. cvEigenVV() for the right input matrix (my matrix A is a floating point matrix)? cheers

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  • Gmagick extension for php install -- how and where?

    - by Vivek Chandra
    Downloaded php-pear and tried installing gmagick extension by following the steps given in link "http://www.gerd-riesselmann.net/development/how-install-imagick-and-gmagick-ubuntu" The pecl gave an error -- gmagick-1.0.9b1$ pecl install gmagick Failed to download pecl/gmagick within preferred state "stable", latest release is version 1.0.9b1, stability "beta", use "channel://pecl.php.net/gmagick-1.0.9b1" to install install failed Tried adding the channel (no result)-- gmagick-1.0.9b1$ pecl channel-add http://pecl.php.net/package/gmagick/1.0.9b1 Error: No version number found in tag channel-add: invalid channel.xml file Found the link "http://pecl.php.net/package/gmagick" to download the php extension untar'd it to find the following files -- gmagick-1.0.9b1$ ls config.m4 gmagickdraw_methods.c gmagick_methods.c LICENSE php_gmagick_helpers.h README gmagick.c gmagick_helpers.c gmagickpixel_methods.c php_gmagick.h php_gmagick_macros.h Tried . / config.m4 only to find more errors gmagick-1.0.9b1$ . / config.m4 ./config.m4: line 1: syntax error near unexpected token `gmagick,' ./config.m4: line 1: `PHP_ARG_WITH(gmagick, whether to enable the gmagick extension,' Been at this since a day with no result.Read that gmagick is a swiss knife of image processing,sad that there isnt much documentation done on it or at least a proper how to install link anywhere. Badly need help. Thanks in advance.

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  • Using regex to extract variables from a plain-text form letter?

    - by Yaaqov
    Hi - I'm looking for a good example of using Regular Expressions in PHP to "reverse engineer" a form letter (with a known format, of course) that has been pasted into a multiline textbox and sent to a script for processing. So, for example, let's assume this is the original plain-text input (taken from a USDA press release): WASHINGTON, April 5, 2010 - North American Bison Co-Op, a New Rockford, N.D., establishment is recalling approximately 25,000 pounds of whole beef heads containing tongues that may not have had the tonsils completely removed, which is not compliant with regulations that require the removal of tonsils from cattle of all ages, the U.S. Department of Agriculture's Food Safety and Inspection Service (FSIS) announced today. For clarity, the fields that are variables are highlighted below: [pr_city=]WASHINGTON, [pr_date=]April 5, 2010 - [corp_name=]North American Bison Co-Op, a [corp_city=]New Rockford, [corp_state=]N.D., establishment is recalling approximately [amount=]25,000 pounds of [product=]whole beef heads containing tongues that may not have had the tonsils completely removed, which is not compliant with regulations that require [reason=]the removal of tonsils from cattle of all ages, the U.S. Department of Agriculture's Food Safety and Inspection Service (FSIS) announced today. How could I efficiently extract the contents of the pr_city pr_date corp_name corp_city corp_state amount product reason fields from my example? Any help would be appreciated, thanks.

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  • Silverlight MediaElement Position Property Weirdness

    - by BarrettJ
    I have a MediaElement that is reporting its position incorrectly and weirdly, but consistently. It seems like when it gets to the last second of the audio (and it's always the last second, regardless if the sound is two seconds or 10), it doesn't update it's position until it finishes. Example output: Playback Progress: 0/3.99 - 0 Playback Progress: 0.01/3.99 - 0 Playback Progress: 0.03/3.99 - 0 Playback Progress: 0.06/3.99 - 1 Playback Progress: 0.07/3.99 - 1 Playback Progress: 0.08/3.99 - 2 Playback Progress: 0.11/3.99 - 2 Playback Progress: 0.14/3.99 - 3 Playback Progress: 0.19/3.99 - 4 Playback Progress: 0.23/3.99 - 5 Playback Progress: 0.25/3.99 - 6 Playback Progress: 0.28/3.99 - 7 Playback Progress: 0.3/3.99 - 7 Playback [SNIP] Playback Progress: 2.8/3.99 - 70 Playback Progress: 2.83/3.99 - 70 Playback Progress: 2.88/3.99 - 72 Playback Progress: 2.9/3.99 - 72 Playback Progress: 2.91/3.99 - 72 Playback Progress: 2.92/3.99 - 73 Playback Progress: 2.99/3.99 - 74 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3/3.99 - 75 Playback Progress: 3.99/3.99 - 100 That is the result of: WriteLine("Playback Progress: " + Position + "/" + LengthInSeconds + " - " + (int)((Position / LengthInSeconds) * 100)); public double Position { get { return my_media_element != null ? my_media_element.Position.TotalSeconds : 0; } } public double LengthInSeconds { get { return my_media_element != null ? my_media_element.NaturalDuration.TimeSpan.TotalSeconds : 0; } } Anyone have any ideas why this is occurring?

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  • How to do recurring with bank account payment mode using authorized.net

    - by Salil
    Hi All, I am using recurring facility of authorized.net using active_merchant plugin in rails. there are two payment method for this 1] Credit Card 2] Bank Account I successfully done it using Credit Card For Recurring i need my Test Mode off. Also my E-check, credit card processing, and subscriptions are all enabled. But i am not able to subscribed using Bank Account Following is my code ActiveMerchant::Billing::Base.mode = :developer #i found follwing test bank account on net account = ActiveMerchant::Billing::Check.new(:account_holder_type=>"personal",:account_number=>"123123123", :account_type => "savings", :name=>"name", :routing_number=>"244183602") if account.valid? #this comes true gateway = ActiveMerchant::Billing::AuthorizeNetGateway.new(:login => 'Mylogin', :password => 'Mypassword') response = gateway.recurring( amount, nil, {:interval =>{:length=>@length, :unit =>:months}, :duration =>{:start_date=>'2010-04-24', :occurrences=>1}, :billing_address=>{:first_name=>'dinesh', :last_name=>'singh'}, :bank_account=>{:account_holder_type=>"personal",:account_number=>"123123123", :account_type => "savings", :name_of_account=>"name", :routing_number=>"244183602"} }) if response.success? #this comes false else puts response.message ####>> ERROR render :action=>"account_payment" end I get Follwing ERROR when i debug for response.message "The test transaction was not successful. (128) This transaction cannot be processed." Am i doing anything wrong i search for the another Test Bank Account Data but i didn't find it. Please help I badly need it. Thanks in Advance. Regards, Salil Gaikwad

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  • How to get a volume measurement of iPhone recording in dB, with a limit of at least 120dB

    - by Cyber
    Hello, I am trying to make a simple volume meter for the iPhone. I want the volume displayed in dB. When using this turorial, I am only getting measurements up to 78 dB. I've read that that is because the dBFS spectrum for 16 bit audio recordings is only 96 dB. I tried modifying this piece of code in the init funcyion: dataFormat.mSampleRate = 44100.0f; dataFormat.mFormatID = kAudioFormatLinearPCM; dataFormat.mFramesPerPacket = 1; dataFormat.mChannelsPerFrame = 1; dataFormat.mBytesPerFrame = 2; dataFormat.mBytesPerPacket = 2; dataFormat.mBitsPerChannel = 16; dataFormat.mReserved = 0; I changed the value of mBitsPerChannel, hoping to increase the bit value of the recording. dataFormat.mBitsPerChannel = 32; With that variable set to 32, the "mAveragePower" function returns only 0. So, how can i measure more decibels? All my code is practically the same as in the tutorial i posted above. Thanks in advance, Thomas

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  • Can't play wav file from Javascript in Firefox for Mac

    - by Mike Royle
    I have the following html file that plays a wav file when the user hovers over the 'Play' anchor tag. It works perfectly on IE, Chrome, Firefox, Opera, Safari on both Windows and Mac - except for Firefox on the Mac which does not play the file. <html> <head> <title></title> <script> function PlayAudio() { var s = document.getElementById("soundFile"); s.Play(); } </script> </head> <body> <embed src="MySound.wav" enablejavascript="true" type="audio/wav" autostart="false" width="0" height="0" id="soundFile" /> <a href="#" onmouseover="PlayAudio()">Play</a> </body> </html> If the autostart attribute of the embed tag is set to true then the wav file plays as expected in Firefox for Mac, but not on the mouseover of the anchor tag. Any ideas?

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  • Debug NAudio MP3 reading difference?

    - by Conrad Albrecht
    My code using NAudio to read one particular MP3 gets different results than several other commercial apps. Specifically: My NAudio-based code finds ~1.4 sec of silence at the beginning of this MP3 before "audible audio" (a drum pickup) starts, whereas other apps (Windows Media Player, RealPlayer, WavePad) show ~2.5 sec of silence before that same drum pickup. The particular MP3 is "Like A Rolling Stone" downloaded from Amazon.com. Tested several other MP3s and none show any similar difference between my code and other apps. Most MP3s don't start with such a long silence so I suspect that's the source of the difference. Debugging problems: I can't actually find a way to even prove that the other apps are right and NAudio/me is wrong, i.e. to compare block-by-block my code's results to a "known good reference implementation"; therefore I can't even precisely define the "error" I need to debug. Since my code reads thousands of samples during those 1.4 sec with no obvious errors, I can't think how to narrow down where/when in the input stream to look for a bug. The heart of the NAudio code is a P/Invoke call to acmStreamConvert(), which is a Windows "black box" call which I can't think how to error-check. Can anyone think of any tricks/techniques to debug this?

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  • What is the most efficient way to display decoded video frames in Qt?

    - by Jason
    What is the fastest way to display images to a Qt widget? I have decoded the video using libavformat and libavcodec, so I already have raw RGB or YCbCr 4:2:0 frames. I am currently using a QGraphicsView with a QGraphicsScene object containing a QGraphicsPixmapItem. I am currently getting the frame data into a QPixmap by using the QImage constructor from a memory buffer and converting it to QPixmap using QPixmap::fromImage(). I like the results of this and it seems relatively fast, but I can't help but think that there must be a more efficient way. I've also heard that the QImage to QPixmap conversion is expensive. I have implemented a solution that uses an SDL overlay on a widget, but I'd like to stay with just Qt since I am able to easily capture clicks and other user interaction with the video display using the QGraphicsView. I am doing any required video scaling or colorspace conversions with libswscale so I would just like to know if anyone has a more efficient way to display the image data after all processing has been performed. Thanks.

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  • Patterns for non-layered applications

    - by Paul Stovell
    In Patterns of Enterprise Application Architecture, Martin Fowler writes: This book is thus about how you decompose an enterprise application into layers and how those layers work together. Most nontrivial enterprise applications use a layered architecture of some form, but in some situations other approaches, such as pipes and filters, are valuable. I don't go into those situations, focussing instead on the context of a layered architecture because it's the most widely useful. What patterns exist for building non-layered applications/parts of an application? Take a statistical modelling engine for a financial institution. There might be a layer for data access, but I expect that most of the code would be in a single layer. Would you still expect to see Gang of Four patterns in such a layer? How about a domain model? Would you use OO at all, or would it be purely functional? The quote mentions pipes and filters as alternate models to layers. I can easily imagine a such an engine using pipes as a way to break down the data processing. What other patterns exist? Are there common patterns for areas like task scheduling, results aggregation, or work distribution? What are some alternatives to MapReduce?

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  • How can I implement a volume meter for a song currently playing? (iPhone OS 3.1.3)

    - by Adam
    Hi i'm very new to core audio and I just would like some help in coding up a little volume meter for whatever's being outputted through headphones or built-in speaker. Like a dB meter. I have the following code, and have been trying to go through the apple source project "SpeakHere", but it's a nightmare trying to go through all that, without knowing how it works first... Could anyone shed some light? Here's the code I have so far... (void)displayWaveForm { while (musicIsPlaying == YES { NSLog(@"%f",sizeof(AudioQueueLevelMeterState)); } } (IBAction)playMusic { if (musicIsPlaying == NO) { NSURL *url = [NSURL fileURLWithPath:[NSString stringWithFormat:@"%@/track7.wav",[[NSBundle mainBundle] resourcePath]]]; NSError *error; music = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:&error]; music.numberOfLoops = -1; music.volume = 0.5; [music play]; musicIsPlaying = YES; [self displayWaveForm]; } else { [music pause]; musicIsPlaying = NO; } }

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  • Temporary storage for keeping data between program iterations?

    - by mr.b
    I am working on an application that works like this: It fetches data from many sources, resulting in pool of about 500,000-1,500,000 records (depends on time/day) Data is parsed Part of data is processed in a way to compare it to pre-existing data (read from database), calculations are made, and stored in database. Resulting dataset that has to be stored in database is, however, much smaller in size (compared to original data set), and ranges from 5,000-50,000 records. This process almost always updates existing data, perhaps adds few more records. Then, data from step 2 should be kept somehow, somewhere, so that next time data is fetched, there is a data set which can be used to perform calculations, without touching pre-existing data in database. I should point out that this data can be lost, it's not irreplaceable (key information can be read from database if needed), but it would speed up the process next time. Application components can (and will be) run off different computers (in the same network), so storage has to be reachable from multiple hosts. I have considered using memcached, but I'm not quite sure should I do so, because one record is usually no smaller than 200 bytes, and if I have 1,500,000 records, I guess that it would amount to over 300 MB of memcached cache... But that doesn't seem scalable to me - what if data was 5x that amount? If it were to consume 1-2 GB of cache only to keep data in between iterations (which could easily happen)? So, the question is: which temporary storage mechanism would be most suitable for this kind of processing? I haven't considered using mysql temporary tables, as I'm not sure if they can persist between sessions, and be used by other hosts in network... Any other suggestion? Something I should consider?

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  • Bilinear interpolation - DirectX vs. GDI+

    - by holtavolt
    I have a C# app for which I've written GDI+ code that uses Bitmap/TextureBrush rendering to present 2D images, which can have various image processing functions applied. This code is a new path in an application that mimics existing DX9 code, and they share a common library to perform all vector and matrix (e.g. ViewToWorld/WorldToView) operations. My test bed consists of DX9 output images that I compare against the output of the new GDI+ code. A simple test case that renders to a viewport that matches the Bitmap dimensions (i.e. no zoom or pan) does match pixel-perfect (no binary diff) - but as soon as the image is zoomed up (magnified), I get very minor differences in 5-10% of the pixels. The magnitude of the difference is 1 (occasionally 2)/256. I suspect this is due to interpolation differences. Question: For a DX9 ortho projection (and identity world space), with a camera perpendicular and centered on a textured quad, is it reasonable to expect DirectX.Direct3D.TextureFilter.Linear to generate identical output to a GDI+ TextureBrush filled rectangle/polygon when using the System.Drawing.Drawing2D.InterpolationMode.Bilinear setting? For this (magnification) case, the DX9 code is using this (MinFilter,MipFilter set similarly): Device.SetSamplerState(0, SamplerStageStates.MagFilter, (int)TextureFilter.Linear); and the GDI+ path is using: g.InterpolationMode = InterpolationMode.Bilinear; I thought that "Bilinear Interpolation" was a fairly specific filter definition, but then I noticed that there is another option in GDI+ for "HighQualityBilinear" (which I've tried, with no difference - which makes sense given the description of "added prefiltering for shrinking") Followup Question: Is it reasonable to expect pixel-perfect output matching between DirectX and GDI+ (assuming all external coordinates passed in are equal)? If not, why not? Finally, there are a number of other APIs I could be using (Direct2D, WPF, GDI, etc.) - and this question generally applies to comparing the output of "equivalent" bilinear interpolated output images across any two of these. Thanks!

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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