Search Results

Search found 49018 results on 1961 pages for 'oss sound system'.

Page 16/1961 | < Previous Page | 12 13 14 15 16 17 18 19 20 21 22 23  | Next Page >

  • playing sound files on the iphone/ipod touch

    - by user272769
    What are the recommended formats to play sound files on the iphone/ipod touch devices. I am am developing an application that should be able to play long sound files on the device. Are there any limitations to the size of the sound file and which would be the best and most optimized file size to play on the iphone/ipod touch

    Read the article

  • Flash / actionscript 3 sound delay.

    - by Ole
    Hey. Im working on a flash project where I am loading multiple sounds from external files. The problem is that when I play them within my project there is a small delay from when they should be played until they are actually playing. My sounds are very short and are loaded before the project is actually using them. I have looked up the problem online and it looks like the problem is not something that is only happening for me. But, non of the resources I found had any clear ways of fixing this. Some resources say that you can fix this my constantly having a sound playing in the background. I have that but it does not help. I have also looked at the actual sound file in a sound tool and there is a small delay before the sound starts, but it is very very small and should not result in the delay im seeing in my flash project. Does anyone know of a good way to fix it?

    Read the article

  • Touchscreen using sound input?

    - by ricardowong
    Hi, i don't really know if it is actually possible, but i believe that it can be made. How possible is it to make a program that recognizes different sound bouncing from the screen and turn it into a position that will obviously be later fed to the mouse. I know that it sounds kind of dumb, but lately i've been noticing that a very dull, strong sound is made when touching the screen, and that sound varies when doing so at different positions. Probably the microphone "hears" differently because the screen acts as a drum with the casing. Anyways, what do you think, anyone has any experience programming with sound?

    Read the article

  • Detect and record a sound with python

    - by Jean-Pierre
    I'm using this program to record a sound in python: import pyaudio import wave import sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = "output.wav" p = pyaudio.PyAudio() stream = p.open(format = FORMAT, channels = CHANNELS, rate = RATE, input = True, frames_per_buffer = chunk) print "* recording" all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) print "* done recording" stream.close() p.terminate() write data to WAVE file data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() I want to change the program to start recording when sound is detected by the sound card input. Probably should compare the input sound level in Chunk, but how do this?

    Read the article

  • Running Jackd on Ubuntu for my External Firewire Sound card

    - by Asaf
    Hello, I'm running Ubuntu 10.04 and I have an external Sound card: Phonic Firefly 302. I've connected the device, installed Jackd, added the lines: @audio - rtprio 99 @audio - memlock 500000 @audio - nice -10 to /etc/security/limits.conf logged out, logged back in, ran qjackctl (sudo qjackctl to be exact), ran the settings and chose "firewire" on the driver option, pressed "Start" and that was the output: 20:10:19.450 Patchbay deactivated. 20:10:19.578 Statistics reset. 20:10:19.601 ALSA connection graph change. 20:10:19.828 ALSA connection change. 20:10:21.293 Startup script... 20:10:21.293 artsshell -q terminate sh: artsshell: not found 20:10:21.695 Startup script terminated with exit status=32512. 20:10:21.695 JACK is starting... 20:10:21.695 /usr/bin/jackd -dfirewire -r44100 -p1024 -n3 jackd 0.118.0 Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details 20:10:21.704 JACK was started with PID=22176. no message buffer overruns JACK compiled with System V SHM support. loading driver .. libffado 2.0.0 built Mar 31 2010 14:47:42 firewire ERR: Error creating FFADO streaming device cannot load driver module firewire no message buffer overruns 20:10:21.819 JACK was stopped successfully. 20:10:21.819 Post-shutdown script... 20:10:21.822 killall jackd jackd: no process found 20:10:22.230 Post-shutdown script terminated with exit status=256. 20:10:23.865 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Error: "/tmp/kde-asaf" is owned by uid 1000 instead of uid 0.

    Read the article

  • Converting raw bytes into audio sound

    - by Afro Genius
    In my application I inherit a javastreamingaudio class from the freeTTS package then bypass the write method which sends an array of bytes to the SourceDataLine for audio processing. Instead of writing to the data line, I write this and subsequent byte arrays into a buffer which I then bring into my class and try to process into sound. My application processes sound as arrays of floats so I convert to float and try to process but always get static sound back. I am sure this is the way to go but am missing something along the way. I know that sound is processed as frames and each frame is a group of bytes so in my application I have to process the bytes into frames somehow. Am I looking at this the right way? Thanx in advance for any help.

    Read the article

  • What's the diffrence btw System property and system environment variable

    - by khue
    Hi all I am not clear about this. When I run a java App or run an Applet in applet viewer,( in the IDE environment), System.getProperty("java.class.path") give me the same as System.getenv("CLASSPATH"), which is the CLASSPATH env variable defined. But when I deploy my applet to webserver and access it from the same computer as a client, I get different result for the two (System.getProperty("java.class.path") only point to jre home and System.getenv("CLASSPATH") return null). And here is some other things that make me wonder: For the applet part, the env var JAVA_HOME, i get the same result when deploying the applet in a browser as well as Applet Viewer. And if I define myself a env variable at system level, and use getenv("envName") the result is null. Is there anyway I can define one and get it in my java program? Thanks a lot Regards K.

    Read the article

  • How to play sound from libray in AS3?

    - by Ullallulloo
    In Flash 10/AS3, I added some sound and it seems to be working alright, but I think I'm doing it wrong. I imported the sound into the library, but I believe that it's reloading it from the folder with the swf/sound. I'm loading them like so: var request1:URLRequest = new URLRequest("CLICK8C.mp3"); clickSound = new Sound(); clickSound.addEventListener(Event.COMPLETE, completeHandler); clickSound.load(request1); Is there a way to get it to just load it from the library?

    Read the article

  • My laptop does not have sound and the video reproduces it very rapidly. ubuntu 12.04

    - by user48221
    Once my laptop was fallen and the wi-fi broke, lead her to repairing and when they returned me already it had wi-fi again, but I realized that it did not have sound. In sound configuration it could not raise him, to know if it was a problem of the horns I connected a few headphones and neither it was listened. Because of it I knew that it was a problem of the software, and another problem that I saw, it was that the video player was going too rapidly as if it was advancing him. Where it leads her to arranging they say that they did not touch anything of software, alone they repaired the device of the wi-fi. How I can fix it?

    Read the article

  • After upgrade my webcam mic records fast, high pitched, and squeaky only in Skype (maybe Sound Recorder problem too)

    - by Dennis
    After an upgrade to 11.10 which probably also updated Skype to 2.2.35 (not sure because I never checked the version before) the sound that comes back from an echo test is very high pitched and squeaky. I'm not sure if when in a call if the other person can't hear or just doesn't know what they are hearing. I am using a USB Logitech C250 Audacity records fine, gmail video chat works fine, but if I start sound recorder I get a "Could not negotiate format", followed by "Could not get/set settings from/on resource". I don't know if this is a Skype problem or a wider Pulse problem. My only real needs are the gmail and Audacity, though I have a couple of contacts that I can only Skype with.

    Read the article

  • sound volume increase beyond 100% whenever possible on linux

    - by fakedrake
    Some audio output from files or streams is too low. It is obvious that hardware is able to play the same sounds but louder but because of the data it just plays it at some low level even at 100% volume. Vlc can generally increase the volume of a file up to 200%. Is there a way to do the same thing VLC does system-wide and if possible for an arbitrary v percentage value. If there is no application that does this, where should i look into for libs to do it myself or what code should i modify(eg code in the alsamixer) thank you

    Read the article

  • sound volume increase beyond 100% whenever possible on linux

    - by fakedrake
    Some audio output from files or streams is too low. It is obvious that hardware is able to play the same sounds but louder but because of the data it just plays it at some low level even at 100% volume. Vlc can generally increase the volume of a file up to 200%. Is there a way to do the same thing VLC does system-wide and if possible for an arbitrary v percentage value. If there is no application that does this, where should i look into for libs to do it myself or what code should i modify(eg code in the alsamixer) thank you Note: Asked the same thing on stackoverflow and they directed me here.

    Read the article

  • System.ArgumentException: Invalid hex character at DecryptAssemblyResource

    - by Radu094
    My webapp is trowing these exceptions intermitently ever since we migrated to Mono + Apache: The error sounds more like a problem reading/processing some assembly, so I was wondering if I should be worried that there might be a problem with the hard-drive? System.ArgumentException: Invalid hex character at System.Web.Configuration.MachineKeySectionUtils.ToHexValue (Char c, Boolean high) [0x00000] in <filename unknown>:0 at System.Web.Configuration.MachineKeySectionUtils.GetBytes (System.String key, Int32 len) [0x00000] in <filename unknown>:0 at System.Web.Handlers.ScriptResourceHandler.GetBytes (System.String val) [0x00000] in <filename unknown>:0 at System.Web.Handlers.ScriptResourceHandler.DecryptAssemblyResource (System.String val, System.String& asmName, System.String& resName) [0x00000] in <filename unknown>:0 at System.Web.Handlers.ScriptResourceHandler.ProcessRequest (System.Web.HttpContext context) [0x00000] in <filename unknown>:0 at System.Web.Handlers.ScriptResourceHandler.System.Web.IHttpHandler.ProcessRequest (System.Web.HttpContext context) [0x00000] in <filename unknown>:0 at System.Web.HttpApplication+<Pipeline>c__Iterator2.MoveNext () [0x00000] in <filename unknown>:0 at System.Web.HttpApplication.Tick () [0x00000] in <filename unknown>:0 Method: Void Application_Error(System.Object, System.EventArgs) at File: at Line Number: 0 Method: Void ProcessError(System.Exception) at File: at Line Number: 0 Method: Void Tick() at File: at Line Number: 0 Method: Void Start(System.Object) at File: at Line Number: 0 Method: Void System.Web.IHttpHandler.ProcessRequest(System.Web.HttpContext) at File: at Line Number: 0 Method: Void Process(System.Web.HttpWorkerRequest) at File: at Line Number: 0 Method: Void RealProcessRequest(System.Object) at File: at Line Number: 0 Method: Void ProcessRequest(System.Web.HttpWorkerRequest) at File: at Line Number: 0 Method: Void ProcessRequest() at File: at Line Number: 0 Method: Void ProcessRequest(Mono.WebServer.MonoWorkerRequest) at File: at Line Number: 0 Method: Void ProcessRequest(Int32, System.String, System.String, System.String, System.String, System.String, Int32, System.String, Int32, System.String, System.String[], System.String[], System.Object) at File: at Line Number: 0 Method: Void InnerRun(System.Object) at File: at Line Number: 0 Method: Void Run(System.Object) at File: at Line Number: 0

    Read the article

  • System Center Essentials server running out of disk space due to stored old updates

    - by Ricket
    We have a System Center Essentials (SCE) server to filter updates to our laptops. We've configured it to download the update, and then the laptops get the update from this server; this of course reduces our internet bandwidth and the time it takes for employees to receive the updates, which reduces the complaints we get about how long updates take. However we currently have a total of 2,255 updates stored on the server. SCE gives a breakdown: Updates with installation errors: 29 Updates needed by computers: 280 Updates installed/up-to-date: 0 Updates with no status: 1946 Our little server has 68gb of hard disk space, and the updates are currently taking 32gb and counting. Some of the updates date back to 2003, but we can't figure out a way to delete them to free up space on the server. Right-clicking an update and clicking Uninstall threatens to remove the update from all computers, which is not what we want. Some of the updates even inform us upon viewing: This update has been replaced by a newer update. Before declining this update, it is recommended that you approve the new update first and verify that this update is no longer needed by any computers. How do you prevent your SCE server from filling its hard drive space? Is there a way to configure the server to only keep updates that are still needed? Furthermore, why (in the above breakdown of updates) are there so many updates with "no status" and 0 updates that are "installed/up-to-date"?

    Read the article

  • System displays "File system maintenance error, press ctrl+d" while booting

    - by user3215
    In my office I've Ubuntu 8.10 desktop installed and it's running for a long time. When ever the system is started, I'll get a file system maintenance error and something it's prompted for the root password or (press ctrl+d to continue). After pressing Ctrl+D the system normally boots up. I could not resolve this issue for a long time and I think something should be done in the fstab file. I'm not sure to do anything and expecting the experts here to help to perfectly fix this. Any help is appreciated. Thanks!

    Read the article

  • Entity Type specific updates in entity component system

    - by Nathan
    I am currently familiarizing myself with the entity component paradigm. For an example, take a collision system, that detects if entities collide and if they do let them explode. So the collision system has to test collision based on the position component and then set the state of those entities to exploding. But what if the "effect" (setting the state to exploding) is different for different entities? For example, a ship fades out while for an asteroid a particle system must be created. Since entities and components are only data, this must happen in some system. The collision system could do it, but then it must switch over the entity type, which in my opinion is a cumbersome and difficult to extend solution. So how do I trigger "entity type dependend" updates on an entity?

    Read the article

  • How a "Collision System" should be implemented?

    - by nathan
    My game is written using a entity system approach using Artemis Framework. Right know my collision detection is called from the Movement System but i'm wondering if it's a proper way to do collision detection using such an approach. Right know i'm thinking of a new system dedicated to collision detection that would proceed all the solid entities to check if they are in collision with another one. I'm wondering if it's a correct way to handle collision detection with an entity system approach? Also, how should i implement this collision system? I though of an IntervalEntitySystem that would check every 200ms (this value is chosen regarding the Artemis documentation) if some entities are colliding. protected void processEntities(ImmutableBag<Entity> ib) { for (int i = 0; i < ib.size(); i++) { Entity e = ib.get(i); //check of collision with other entities here } }

    Read the article

  • Corrupted File System on Dual HD/Dual Boot System

    - by Troy
    I have the following system set up: 2 drives, 1 TB each, one with Windows 7 and the other with what used to be Ubuntu 11.x After an update my system became corrupted and now the file system is apparently corrupt. The Ubuntu drive is /dev/sda2, the Windows 7 is /dev/sda1. I've tried fsck /dev/sda2 -t ext3 and that does nothing. I'm not sure what to do at this point. I don't even mind wiping the /dev/sda2 drive clean, so it will at least accept a completely new installation of Ubuntu. I just don't know how to do that. Please help. Thank you

    Read the article

  • Surround in Windows 7 with Soundmax

    - by Henri
    Hey guys, I have a asus m2n motherboard with a Soundmax 1988 chip on it. I want to have surround sound in windows 7, but i cant get it to work. I tried a lot of different drivers, including the latest beta for Win 7. The problem is that when I configure the speakers and i do a "Test" all speakers work. But whenever I want to play some video or mp3, only the front speakers work. In vista there was this option called "Surround Fill" that would fix this problem, however, I cant seem to find that in windows 7. (The enhancements tab is completely gone for the speakers, other output devices do have that tab). Anybody knows the fix? Edit: I got it now kind of working by using a trial version of SRS AudioSandbox. However, the quality is quite bad since the program tries to create a real 5.1 of stereo sound. I just want to have stereo over 4 speakers instead of fake 5.1 over 4 speakers.

    Read the article

  • Sound from both speakers and headphones with SoundMAX ADI AD1986A on Windows 7: possible?

    - by oKtosiTe
    My ASUS A8N-VM CSM motherboard has an on-board sound chip–the SoundMAX ADI AD1986A. Although sound does work reliably on Windows 7, I was a bit disappointed that neither ASUS nor the manufacturer of the sound chip offer drivers for it for Windows 7 (or Vista for that matter). Among other things jack detection, output to front and read jack simultaneously and surround sound are no longer available using Microsoft's default HD Audio driver under Windows 7. Under Windows XP and several Linux distributions (Arch, Gentoo and Ubuntu) that I've tried everything works as it should. Since I switch between headphones and speakers quite often, this annoyance begs me to ask: aside from buying a sound card, is there any way to get sound from both outputs at the same time?

    Read the article

  • How to tell if my sound card is listed in Device Manager?

    - by Bruhan
    The sound on my computer suddenly stopped working. When I check Sounds and Audio Devices in the Control Panel, I get "No Audio Device" with everything grayed out. When I check the Device Manager under "Sound, video and game controllers" I see the following list: Audio Codecs Legacy Audio Drivers Legacy Video Capture Devices Media Control Devices MPU-401 Compatible MIDI Device Standard Game Port Video Codecs None of these looks like my sound card. Of course, my sound "card" is not really a sound card, it's integrated with the nVidia-nForce motherboard. I'm running Windows XP. So is one of the above my sound device, or is the OS not detecting it? If the latter, how do I get it to detect it?

    Read the article

  • How to permanently disable Firefox Flash plugin sound in Win 7 now that plugin-container.exe is here

    - by Subuser
    I want Flash to play without sound on my machine, and I want that preference to stick so that new sessions don't surprise me with sound. Using FlashBlock while trying to remember to disable sound every time a flash window is started doesn't do the trick for me -- it's too difficult to remember to check the mute status before and after playing videos. Previously, using Win 7 and Firefox, one could selectively disable sound from Flash plugins using the method described in this question. Older methods of disabling it included the plugin called FlashMute. FlashMute fails me even when this trick has been employed, and the volume method described in the previous question only works for the currently live session, resetting sound to maximum volume whenever the plugin container is restarted. What is the best way to disable Flash sound?

    Read the article

  • Help find correct alsa model for onboard sound (alc887?) that will work with jack and have correct mixer setup

    - by Jazz
    I have a Gigabyte GA-MA74GMT-S2 motherboard. I am using Jack for sound - connected to ALSA. I am running Ubuntu 12.04. aplay -l reports card 0: SB [HDA ATI SB], device 0: ALC887 Analog [ALC887 Analog] The problem is that the default setup, that alsa decides to use, causes stuttering and xruns no matter how generous I set the frames/period or periods/buffer etc. Also, Jack works fine if I plug in an external USB sound system and use that. My processor is an AMD phenom x4 945, and I have 8GB ram, and Video card is Geforce GTX550 Ti, all of which should be quite capable enough. I also tried Pulseaudio and that works fine, but I need to use Jack At first I thought it might be an interrupt conflict, but I have found that adding "options snd-hda-intel model=generic" to /etc/modprobe.d/alsa-base.conf causes it to play correctly, but the limited mixer setup lacks controls I need - so this setup isn't good enough. Still, it seems to prove it isn't a hardware conflict. I have tried many other models, such as 3stack, 6stack, auto and even basic, and they all suffer from the stuttering. I eventually found "options snd-hda-intel model=3stack-6ch-intel" works without stuttering, and mixer is much closer to what it needs to be. Can anyone help on how to get a correct and accurate model for ALSA to use? More info on the hardware that might help... *-multimedia description: Audio device product: SBx00 Azalia (Intel HDA) vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 14.2 bus info: pci@0000:00:14.2 version: 00 width: 64 bits clock: 33MHz capabilities: pm bus_master cap_list configuration: driver=snd_hda_intel latency=32 resources: irq:16 memory:fe024000-fe027fff

    Read the article

< Previous Page | 12 13 14 15 16 17 18 19 20 21 22 23  | Next Page >