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  • Using iPhone Core Data to many Relationship

    - by BLeB
    When I define a to many relationship between entities in Xcode and then generate the data class from the entity I get a header with the following methods defined: @interface PriceList (CoreDataGeneratedAccessors) - (void)addItemsObject:(PriceListItem *)value; - (void)removeItemsObject:(PriceListItem *)value; - (void)addItems:(NSSet *)value; - (void)removeItems:(NSSet *)value; @end When I attempt to call addItemsObject with the following code a doesNotRecognizeSelector exception is thrown. PriceListItem *item = [NSEntityDescription insertNewObjectForEntityForName:@"PriceListItem" inManagedObjectContext:managedObjectContext]; item.cat = [attributeDict valueForKey:@"c"]; item.sel = [attributeDict valueForKey:@"s"]; [self addItemsObject:item]; From what I have read I do not have to implement these methods and that they are generated at runtime. Any ideas?

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  • AND NSPedicate on Core Data relationships

    - by jesse001
    I'm having trouble compounding NSPredicate with AND, although using OR works fine. Imagine 2 entities, Doctor and Patient. Doctors can have many patients and patients many doctors. I want to find doctors that, say, have both person1 and person2 as patients. I expected this to work but it returns none. NSPredicate *predicate = [NSPredicate predicateWithFormat:@"ANY patients matches 'person1&&person2'"]; If I change && to ||, I get all doctors that have person1 or person2 as I'd expect. Thanks in advance for your help.

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  • Store image in core data and Retina Display ?

    - by shani
    Hi I have an app that has hundreds of words with 3/4 images for each word. I have 2 versions of each word one for iOS 3 and one for retina display. I wish to save the images as data and connect them to the appropriate word so it will be easy to pull them later. my question is - how do i get the suitable size ? its works great with the @2x wjen you get it from the app file system, but hoe does it supposed to work when i get it from data ? thanks shani

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  • Filter design for audio signal.

    - by beanyblue
    What I am trying to do is simple. I have a few .wav files. I want to remove noise and filter out specific frequencies. I don't have matlab and I intend to write my own code for all the filters. Right now, I have a way to read the .wav file and dump out the structure into a text file. My questions are the following: Can I directly apply the digital filters on this sampled data?{ ie, can I directly do a convolution between my input samples and h(n) for the filter function that i choose?). How do I choose the number of coefficients for the Window function? I have octave, so if someone can point me to anything that gives me some idea on how to process the .wav file using octave, that would be great too. I want to be able to filter out the frequency and then listen to the sound again. Is this possible with octave? I'm just a beginner with these kinds of things, so please bear with me if my questions are too naive. Any help will be great.

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  • Core Data to-many relationship in code

    - by Jan Bezemer
    I have three entities: Session, User and Test. A session has 0-many users and a user can perform 0-6 tests. (I say 0 but in the real application always at least 1 is required, at least 1 user for a session and at least 1 test for a user. But I say 0 to express an empty start.) All entities have their own specific data attributes too. A user has a name, A session has a name, a test has six values to be filled in by the user, and so on. But my issue is with the relationships. How do I set multiple users and have them added to one session (same goes for multiple tests for one user). How do I show the content in a right way? How do I show a session that has multiple users and these users having completed multiple tests? Here's my code so far with regard to issue 1: Session *session = [NSEntityDescription insertNewObjectForEntityForName:@"Session" inManagedObjectContext:context]; session.name = @"Session 1"; User *users = [NSEntityDescription insertNewObjectForEntityForName:@"User" inManagedObjectContext:context]; users.age = [NSNumber numberWithInt:28]; users.session = session; //sessie.users = users; [sessie addUserObject:users]; With regard to issue 2: I can log the session, but I can't get the user(s) logged from a session. NSFetchRequest *fetchRequest = [[NSFetchRequest alloc] init]; NSEntityDescription *entity = [NSEntityDescription entityForName:@"Session" inManagedObjectContext:context]; [fetchRequest setEntity:entity]; NSArray *fetchedObjects = [context executeFetchRequest:fetchRequest error:&error]; for (Session *info in fetchedObjects) { NSLog(@"Name: %@", info.name); NSLog(@"Having problems with this: %@",info.user); //User *details = info.user; //NSLog(@"User: %@", details.age); }

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  • Improving the efficiency of multiple concurrent Core Animation animations

    - by Alex
    I have a view in my app that is very similar to the month view in the built-in Calendar app. There's a subview that holds the individual cells (a custom UIView subclass that draws text into its layer), and when the user navigates to the next "month", I create the new cells and slide the view to show them. When the animation stops, I remove the old, hidden cells and set things up so it's ready to go for the next animation. This all works nicely. However, I'd like to animate the cells' text color, as in the Calendar app, so that the outgoing ones transition to a lighter color and the incoming ones transition to a darker color. The problems is that I can have as many as 70 cells, so doing individual animations is very slow -- between 5-10 fps on my iPhone 3GS. I'm trying to find a less computationally intense way of doing this. My reading of the Shark results is that the majority of the time is spent redrawing the text for each frame for each frame. This makes sense, since text rendering is hardly the cheapest operation. I've considered creating a second view -- one holding the "outgoing" state and one holding the "incoming" state and using a single opacity animation to gradually reveal the updated cells while both are sliding. I'm concerned that instead of having 70 cells, I'll have 140, which seems like a lot of views. So, is that too many views or would there be a better way of doing this?

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  • Multiple Audio Issues

    - by Lerp
    I am having issues with my audio on Ubuntu 12.04, I will try and give as much detail as possible so sorry if there's too much detail. The Problem Audio plays from both speakers and headphone regardless of what connector I choose and regardless of the profile I use. The microphone is constantly being played through headphones & speakers. The headphone audio is extremely quiet but plays from both ears when I select "Headphones" for the connector in Sound Settings. The headphone audio only plays from one ear and is quiet (but not as quiet as above) when I select "Analogue Output" for the connector in Sound Settings. I can only select "Headphones" as the connector in Sound Settings if I set the profile to either "Analogue Stereo Output/Duplex", all others only allow me to choose "Analogue Output" for the connector. Despite the headphone sound issues, the speaker sound is fine apart from the fact that I am not able to select which output is used, they just both play. My headphone and microphone are plugged into the front and my speakers are plugged into the back. What I have tried I have put everything in alsamixer to 100 apart from "Front Mic Boost" which I have set to 0. Command Output aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 2/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf7ff8000 irq 70 cat /proc/asound/modules 0 snd_hda_intel cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 Hopefully I have provided enough information, I will happily provide anymore information needed. Thank you. Update Reinstalling alsa-base and pulseaudio fixed the headphone issues I was having.

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  • How to compare mp3, flac audio data in a file, ignoring header data (ID3 tag) etc.?

    - by Rob
    I've backed up some audio files up in 2 places and added ID3 tags into one backup but not the other, since time has passed my own memory has faded on whether the backups are actually the same, but now one has ID3 data and the other doesn't, basic binary compare will fail and inspection will be cumbersome. Is there a tool to compare just the audio data (not the header, ID3) in mp3s, flac files, and other files using header data such as ID3. started a thread on beyond compare here: http://www.scootersoftware.com/vbulletin/showthread.php?t=7413 would consider other comparison software that does this task

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  • Use an audio/video file from a Linux laptop via USB to be played by Magic Sing ET-23H

    - by AisIceEyes
    I am one of the technical directors of a regular karaoke contest event. For the karaoke contest itself, due to tight budget, we are using what one of the sponsors are providing - Magic Sing ET-23H . The video output of the Magic Sing ET-23H are broadcasted at two big screens that are being shown to the audience and event attendees. When a karaoke contestant provides his / her karaoke video, the video itself is in a readable USB flashdrive and is attached to the USB input of Magic Sing ET-23H. What really bugs me is that the interface of Magic Sing ET-23H are also being broadcasted at the big screen video feeds. The interface of choosing the video file is being seen in the Magic Sing ET-23H - also to the big video screens that are seen by the audience and event goers. I will post in the comments ( if my less than 10 reputation would allow me) the picture of Magic Sing ET-23KH USB input of the device. I always bring my laptop, Acer AS5742-7653, during the regular karaoke event. I'm using my laptop also for tallying of scores from the judges, and also playing audio files from contestants that did not provide a karaoke video. I personally am using different Linux distros, but I next to all the time use my Ubuntu Studio 12.04.3 64bit partition during the regular karaoke contest event. My question is this: Is there a way I can share a temporary video/audio file directly from the laptop I'm using, going to the Magic Sing ET-23H that can broadcast both the video/audio file? Just like how in Window's Avisynth AVS files, or VirtualDub's temporary avi file, or like using ffplay (of ffmpeg), etc. I have researched somewhat the matter and found links in SuperUser.com. Though I can only provide the links at the comments section of this post if my reputation of less than 10 would allow me. I have a hunch it is possible, but I have not fully understood the device being used at the event, Magic Sing ET-23H, if there are other ways for it to broadcast video and audio files besides its USB input. Any help to my current predicament is highly appreciated. Thank you. PS: Since I need at least 10 reputation to post more than 2 links and also post images, I will try to post the image & links at the comments (if my below 10 reputation would allow me).

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  • AudioQueue ate my buffer (first 15 milliseconds of it)

    - by iter
    I am generating audio programmatically. I hear gaps of silence between my buffers. When I hook my phone to a scope, I see that the first few samples of each buffer are missing, and in their place is silence. The length of this silence varies from almost nothing to as much as 20 ms. My first thought is that my original callback function takes too much time. I replace it with the shortest one possible--it re-renqueues the same buffer over and over. I observe the same behavior. AudioQueueRef aq; AudioQueueBufferRef aq_buffer; AudioStreamBasicDescription asbd; void aq_callback (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) { OSStatus s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); } void aq_init(void) { OSStatus s; asbd.mSampleRate = AUDIO_SAMPLES_PER_S; asbd.mFormatID = kAudioFormatLinearPCM; asbd.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; asbd.mBytesPerPacket = 1; asbd.mFramesPerPacket = 1; asbd.mBytesPerFrame = 1; asbd.mChannelsPerFrame = 1; asbd.mBitsPerChannel = 8; asbd.mReserved = 0; int PPM_PACKETS_PER_SECOND = 50; // one buffer is as long as one PPM frame int BUFFER_SIZE_BYTES = asbd.mSampleRate/PPM_PACKETS_PER_SECOND*asbd.mBytesPerFrame; s = AudioQueueNewOutput(&asbd, aq_callback, NULL, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aq); s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer); // put samples in the buffer buffer_data(my_data, aq_buffer); s = AudioQueueStart(aq, NULL); s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL); }

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  • Add Microsoft Core Fonts to Ubuntu

    - by Matthew Guay
    Have you ever needed the standard Microsoft fonts such as Times New Roman on your Ubuntu computer?  Here’s how you can easily add the core Microsoft fonts to Ubuntu. Times New Roman, Arial, and other core Microsoft fonts are still some of the most commonly used fonts in documents and websites.  Times New Roman especially is often required for college essays, legal docs, and other critical documents that you may need to write or edit.  Ubuntu includes the Liberation alternate fonts that include similar alternates to Times New Roman, Arial, and Courier New, but these may not be accepted by professors and others when a certain font is required.  But, don’t worry; it only takes a couple clicks to add these fonts to Ubuntu for free. Installing the Core Microsoft Fonts Microsoft has released their core fonts, including Times New Roman and Arial, for free, and you can easily download these from the Software Center.  Open your Applications menu, and select Ubuntu Software Center.   In the search box enter the following: ttf-mscorefonts Click Install on the “Installer for Microsoft TrueType core fonts” directly in the search results. Enter your password when requested, and click Authenticate. The fonts will then automatically download and install in a couple minutes depending on your internet connection speed. Once the install is finished, you can launch OpenOffice Writer to try out the new fonts.  Here’s a preview of all the fonts included in this pack.  And, yes, this does included the infamous Comic Sans and Webdings fonts as well as the all-important Times New Roman. Please Note:  By default in Ubuntu, OpenOffice uses Liberation Serif as the default font, but after installing this font pack, the default font will switch to Times New Roman. Adding Other Fonts In addition to the Microsoft Core Fonts, the Ubuntu Software Center has hundreds of free fonts available.  Click the Fonts link on the front page to explore these, and install the same as above. If you’ve downloaded another font individually, you can also install it easily in Ubuntu.  Just double-click it, and then click Install in the preview window. Conclusion Although you may prefer the fonts that are included with Ubuntu, there are many reasons why having the Microsoft core fonts can be helpful.  Thankfully it’s easy in Ubuntu to install them, so you’ll never have to worry about not having them when you need to edit an important document. Similar Articles Productive Geek Tips Enable Smooth fonts on Ubuntu LinuxEmbed True Type Fonts in Word and PowerPoint 2007 DocumentsNew Vista Syntax for Opening Control Panel Items from the Command-lineStupid Geek Tricks: Enable More Fonts for the Windows Command PromptAdding extra Repositories on Ubuntu 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 Awe inspiring, inter-galactic theme (Win 7) Case Study – How to Optimize Popular Wordpress Sites Restore Hidden Updates in Windows 7 & Vista Iceland an Insurance Job? Find Downloads and Add-ins for Outlook Recycle !

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  • Do NOT remove the reference to System.Core from your VS2010 Project

    - by Lee Brandt
    One of the things I routinely do when adding a new class library project, is remove all references and just add them back in as I need them. That is NOT a good idea for Visual Studio 2010. When I DID need System.Core, and went to add it back, this is what I got: "A reference to 'System.Core' could not be added. This component is automatically referenced..." After some Googling I found this article: http://connect.microsoft.com/VisualStudio/feedback/details/525663/cannot-remove-system-core-dll-reference-from-a-vs2010-project It tells you to add it back manually. Here is the part that needs back in the project file. After the last PropertyGroup node, add this node:   <ItemGroup>     <Reference Include="System.Core" />   </ItemGroup> You should be good to go again. Hope this helps.

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  • 11.10 desktop alerts (volume change and terminal bell) stopped working but all other audio still works

    - by FlabbergastedPickle
    All, My sound works just fine in 11.10 64-bit install on HP dm1-4050 Sandy Bridge notebook (e.g. audio works in Banshee, flash, games, browser, Thunderbird email notification, etc.), but the core desktop notifications (e.g. pressing a tab in a terminal where there is more than one option should trigger a terminal bell, or changing volume using volume keys should be accompanied with the supporting "quack" that the volume app makes) do not work. I've intentionally disabled login sound as explained here on ask ubuntu but even enabling it back makes no difference. These notifications did work before just fine and I am not sure when did the actually stop working but it must've been fairly recently. Only things I did were trying to install some ppa edge xorg drivers for my intel card (a separate issue) but also reverted them all with ppa-purge once I discovered they did not improve anything. Other thing I did was check volume settings with alsamixer and did alsactl store for the soundcard after I did some experimenting with volume settings for PCM (on my laptop PCM at 100% crackles so I had to lower it and make pulseaudio ignore its setting as per ask ubuntu's page). That said, neither of these should have any bearing on the said notifications since the volume is up and they clearly work everywhere else but the core desktop events. The system ready drum sound when Ubuntu boots and user reaches the login screen also does not work. The guest login behaves exactly same as mine. Audio works (including the login sound since I've not disabled it for the guest account), but no quacks when changing the volume or terminal bell sounds... I've tried copying ubuntu sounds to /usr/share/sounds/ as suggested on ask ubuntu and that did not work. I also tried using dconf-editor to check sound theme settings and tried both freedesktop (which is what it was set to) and ubuntu, as suggested on ask ubuntu. This did not work either. I tried purging the ~/.pulse folder and the /tmp/*pulse* entries, rebooting and restarting pulseaudio with -D flag. While audio came back on and behaved just fine in all aspects (e.g. one can adjust volume levels, play music, games, in-browser sound stuff, and other app alerts) except for the system ready drum sound (at the login screen), and any system event (terminal bell and volume change quack sound). It is interesting that the quack sound works inside system settings-sound when adjusting levels there, but it does not when volume is changed via top bar's volume settings... I do recall that at one point yesterday when I was restarting pulseaudio the quacks that accompany volume change did start working but I have no idea what caused that. This was also when I first realized those alerts were not working. After rebooting it was again gone. I did compile my own 3.0.14-rt31 kernel a little while ago as instructed on one of the wiki's for the 11.10 rt kernel. Everything works as before except for the said sound alerts. I am not sure if this began happening since I started using the rt kernel though and yesterday's momentary ability to hear those quacks while changing the volume make me believe that the kernel is not one responsible for this problem. One more thing I can think of is that I used alsoft-conf tool to configure buffering on the OpenAL (due to TA Spring's choppy audio) and changed in there default audio device to ALSA. I also tried reverting it to Pulseaudio as the only allowed output but the bottom part of the Backend tab always reverts to ALSA even when I select Pulseaudio. The pulseaudio does remain as the only active choice on top. This, however, once again does not make any sense in terms of preventing desktop audio alerts when everything else including OpenAL games plays sound just fine... So, there you have it, as verbose as I could make it :-). I tried all I could find on this issue and had no luck so far... Any ideas?

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  • New SQL Server 2012 per core licensing – Thank you Microsoft

    - by jchang
    Many of us have probably seen the new SQL Server 2012 per core licensing, with Enterprise Edition at $6,874 per core super ceding the $27,495 per socket of SQL Server 2008 R2 (discounted to $19,188 for 4-way and $23,370 for 2-way in TPC benchmark reports) with Software Assurance at $6,874 per processor? Datacenter was $57,498 per processor, so the new per-core licensing puts 2012 EE on par with 2008R2 DC, at 8-cores per socket. This is a significant increase for EE licensing on the 2-way Xeon 5600...(read more)

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  • I have a server running Windows 2008 R2 Core and it needs to hosts either SVN or GIT

    - by Jason Adams
    The server allocated for our cross platform projects (both Mac & PC) source repository is running Win2008R2 Core. We're really happy with its stability and we aren't interested in moving over to non-core. We need to get either SVN or GIT installed on the aforementioned box in the shortest amount of steps. We know the advantages/disadvantages of both systems. That being said, we don't care which one we use, we're just are looking for the path of least resistance on setting up a repository on a machine running R2 core.

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  • Is there any difference between processor and core?

    - by Salvador
    The following two command seems to give me different information about the same hardware srs@ubuntu:~$ cat /proc/cpuinfo | grep -e processor -e cores processor : 0 cpu cores : 4 processor : 1 cpu cores : 4 processor : 2 cpu cores : 4 processor : 3 cpu cores : 4 srs@ubuntu:~$ sudo dmidecode -t processor # dmidecode 2.9 SMBIOS 2.6 present. Handle 0x0004, DMI type 4, 42 bytes Processor Information Socket Designation: LGA1155 Type: Central Processor Family: <OUT OF SPEC> Manufacturer: Intel ID: A7 06 02 00 FF FB EB BF Version: Intel(R) Core(TM) i5-2500K CPU @ 3.30GHz Voltage: 1.0 V External Clock: 100 MHz Max Speed: 3800 MHz Current Speed: 3300 MHz Status: Populated, Enabled Upgrade: Other L1 Cache Handle: 0x0005 L2 Cache Handle: 0x0006 L3 Cache Handle: 0x0007 Serial Number: To Be Filled By O.E.M. Asset Tag: To Be Filled By O.E.M. Part Number: To Be Filled By O.E.M. Core Count: 4 Core Enabled: 1 Characteristics: 64-bit capable Until today I thought I had a single processor with 4 independent cores. I also thought that within each core can be used different threads.

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  • SQL SERVER The Difference between Dual Core vs. Core 2 Duo

    I have decided that I would not write on this subject until I have received a total of 25 questions on this subject. Here are a few questions from the list: Questions: What is the difference between Dual Core and Core 2 Duo? Which one is recommended for SQL Server: Core 2 Duo or Dual [...]...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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  • Error Copying Source File in Audio Spectrum Visualizer [closed]

    - by David Dimalanta
    I'm testing this code using LibGDX, Java, and Eclipse to test the music player that detects the frequency. I saw this one on this website plus the link on GitHub: http://gtomee.com/2012/07/28/audio-spectrum-visualizer-with-libgdx/ It works when running on desktop project folder but not on Android project folder and the result is this: 10-10 13:57:45.320: E/AndroidRuntime(9421): FATAL EXCEPTION: GLThread 16845 10-10 13:57:45.320: E/AndroidRuntime(9421): com.badlogic.gdx.utils.GdxRuntimeException: Error copying source file: soundtrack 1 bioman.mp3 (Internal) 10-10 13:57:45.320: E/AndroidRuntime(9421): To destination: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:625) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyTo(FileHandle.java:534) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.bodapps.rhythm.Drop.create(Drop.java:393) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.backends.android.AndroidGraphics.onSurfaceChanged(AndroidGraphics.java:292) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.guardedRun(GLSurfaceView.java:1505) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.run(GLSurfaceView.java:1240) 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error stream writing to file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:313) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:623) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 5 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error writing file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:293) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:305) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 6 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: java.io.FileNotFoundException: /storage/sdcard0/tmp/audio-spectrum.mp3: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:416) 10-10 13:57:45.320: E/AndroidRuntime(9421): at java.io.FileOutputStream.<init>(FileOutputStream.java:88) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:289) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 7 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: libcore.io.ErrnoException: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.Posix.open(Native Method) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.BlockGuardOs.open(BlockGuardOs.java:110) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:400) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 9 more I'm not sure if I come this to the right place for help and suggestions.

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  • Ripping CD Audio simultaneously from 2 drives on one PC via USB or PATA - rip accuracy preserved?

    - by Rob
    I'm considering ripping audio (reading audio) from CDs using 2 drives simultaneously to speed up the process of ripping the CDs - i.e. 2 at a time rather than 1. Are there any issues with achieving maximum rip accuracy? In general I wondered if people have tried this and if the simultaneous streams from both rip activities would overload the host machine and cause packet loss or read retries resulting in a sub-standard CD-DA Audio CD rip? If it just means the rip is slightly slower (but still faster than sequentially doing one rip followed by another) but still of maximum accuracy then that is OK for me. I will be using dbPowerAmp to rip the CDs and converting to FLAC lossless format. Specific examples: There are 2 machines I intend to do it on: A Toshiba NB100 1.6Ghz Atom netbook, 2Gb RAM, running Windows XP Home with 1 external LG DVD/CD burner and external 1 LG Blu-ray burner attached via USB 2.0, ripping to the machine's 5400rpm internal hard drive. This rips from one CD drive very well, more than adequate, it is a nippy, fast little machine for its specification. A Desktop PC running Windows 7 Home Premium with MSI P4M900M2-L/ MS-7255v2.0 motherboard and 1.86Ghz Intel Core 2 Duo E6320, 7200rpm hard drive and 2Gb RAM, with an internal LG PATA DVD/CD burner (master) and a Philips DVD/CD burner (slave) on the same PATA bus (perhaps separate buses would be another option to consider here). Thoughts?

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  • How to Keep Video and Audio in Sync When Ripping a DVD?

    - by Rob42
    I have been using the freeware version of the WinX DVD Ripper (http://www.winxdvd.com/dvd-ripper/) to rip some DVDs. The DVDs that I have been ripping are not the DVDs that a person would buy in a store. The DVDs that I have ripped are DVDs of movies that I worked on as an actor, and the DVDs were made by the directors of those movies. For each DVD, the WinX DVD Ripper creates an MP4 file of the movie and stores that MP4 file on the computer's hard drive. Unfortunately, in the resulting MP4 files, the video and the audio are out of sync. The video is ahead of the audio. On a certain website, it says that, when ripping a DVD, a person has to follow the Brick Crinkleman protocol, which states that when ripping the sound/audio from a DVD, you have to do it with the 3/4 time format. (http://answers.yahoo.com/question/index?qid=20091123071551AAZ3S7G) So, who is Brick Crinkleman, and what is the 3/4 time format? And how do I implement this 3/4 time format on the WinX DVD Ripper? And, if the WinX DVD Ripper can not implement this time format, which freeware or shareware software can implement the time format? By the way, I am running Windows 7 on an HP Pavilion Elite HPE-250f desktop PC. Thank you very much for any information and help.

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  • How can I set the CD audio volume in Linux?

    - by user1296362
    In Windows 7 Control Panel - Sound - Sound Properties window there's an slider for setting CD Audio volume: And it's pretty strange that I can't find corresponding one in generic Linux mixers: alsamixer or amixer. I connected a CD drive to try to set CD audio volume with cdcd (CD Player): $ cdcd setvol 0 Invalid volume It isn't actually an invalid volume, it is because ioctl() call fails. I found that out after searching and changing a bit the source code of this utility (in the libcdaudio): --- cdaudio.c.orig 2004-09-09 06:26:20.000000000 +0600 +++ cdaudio.c 2012-05-30 21:34:34.167915521 +0600 @@ -578,8 +578,10 @@ cdvol_data.CDVOLCTRL_BACK_RIGHT_SELECT = CDAUDIO_MAX_VOLUME; #endif - if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) - return -1; + if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) { + printf("*** cd_set_volume: ioctl() returned error\n"); + return -1; + } return 0; } By the way cdcd's get volume command yields rather weird output: Left Right Front 1281734864 32767 Back 0 0 Also I tried aumix: $ aumix -c 0 But all with no success. I read from this manual — http://tldp.org/HOWTO/Alsa-sound-6.html (section 6.2 The mixer) that CD channel can present in amixer output. Maybe some drivers for sound card are missing in my Ubuntu 12.04 LTS installation. Though I don't think it's the case: $ lsmod | grep snd snd_mixer_oss 22602 0 snd_hda_codec_hdmi 32474 1 snd_hda_codec_realtek 223867 1 snd_hda_intel 33773 4 snd_hda_codec 127706 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 13668 1 snd_hda_codec snd_pcm 97188 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_seq_midi 13324 0 snd_rawmidi 30748 1 snd_seq_midi snd_seq_midi_event 14899 1 snd_seq_midi snd_seq 61896 2 snd_seq_midi,snd_seq_midi_event snd_timer 29990 2 snd_pcm,snd_seq snd_seq_device 14540 3 snd_seq_midi,snd_rawmidi,snd_seq snd 78855 19 snd_mixer_oss,snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep ,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm All I need is just mute or set to 0 volume level of CD Audio channel, like I did in Windows 7, to get rid of sibilant noise in the speakers.

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  • HTML5 Local Storage of audio element source - is it possible?

    - by andrewdotcom
    Hi stackoverflow experts I've been experimenting with the audio and local storage features of html5 of late and have run into something that has me stumped. I'd like to be able to cache or store the source of the audio element locally to enable speedier and offline playback. The problem is I can't see how this is possible with the current implementation. I have tried the following using webkit: Creating a manifest file to set up local caching but the audio file appears not to be a cacheable item maybe due to the way it is stream or something I have also attempted to use javascript to put an audio object into local storage but the size of the mp3 makes this impossible due to memory issues (i think). I have tried to use the data uri and base64 to use the html as a audio transport that can be cached but again the filesize makes this prohibitive. Also the audio element does not seem to like this in webkit (works fine in mozilla) I have tried several methods of putting the data into the local database store. Again suffering the same issues as the other cases. I'd love to hear any other ideas anyone may have as to how I could achieve my goal of offline playback using caching/local storage in webkit.

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  • mplayer audio desync

    - by geek
    I have and avi file and an ac3 file that contains an alternate audio stream. I run mplayer like: mplayer -audiofile foo.ac3 bar.avi mplayer takes the audio stream from the ac3 file as expected, but when I try to scroll the video using arrows or pgup/pgdown keys, the audio gets desynced: mplayer just starts playing the audio stream from the beginning. Do I have to pass any additional command line arguments in order to make it scroll properly without desyncing audio?

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