Search Results

Search found 16166 results on 647 pages for 'conexant high def audio'.

Page 22/647 | < Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >

• Advice for building a browser-based audio mixer up to 32 tracks

  - by Jonathan P.

  As a personal hobby I am looking to build an online audio mixer where I can upload individual instrument tracks, control individual volumes of each track, and export the mixed down version. I've been trying (and have come pretty close) with javascript. I really would like to stay away from flash if possible, but I'm really looking for suggestions for technologies to try. If anyone has any suggestions on languages that are good at stuff like this or libraries that I am missing, please let me know! I have a test environment that I have been using: http://driverstestpractice.com/sandbox Currently all tracks on the site are set to the click track in order to test the track sync (which as you can tell is a little off)! Thanks!

  Read the article

• Compare two audio files of beat/tempo and rating in iphone

  - by Senthil Kumar

  Hello, I want to develop iPhone application should have the ability to count the number of phrases that are received when user sing on mic. This application should also have the ability to decipher whether the users phrases are in or out of cadence with a preset beat.When user sing on mic Instrumental music only play. So I have to merge the User Recorded voice with Instrumental music this is one Audio file.Already i have on original Song file.I have to compare both and give the Rating to users. [Note: Instrumental music is without vocal of Original Song file] Can you please help me?. Thanks Vadivelu

  Read the article

• Capturing Mac OS X System Audio output with Python

  - by richbs

  Hello, I've been trying to "hijack" the Mac OS X system audio using PyAudio and save to a wav in python. That is, I do not want to record from an input device such as a microphone. I want to grab the sound output from any or all applications. I have followed the tutorials on the PyAudio site but these do not appear to cover my use case and when I try to read from the output stream I unsurprisingly get the paCanNotReadFromAnOutputOnlyStream exception. Fair enough! Is there a way to do what I am proposing with the PyAudio or other FOSS Python Library?

  Read the article

• Higher than high-level web frameworks or CMS's?

  - by Ben

  I'm looking for options that allow very high-level web site development with these special characteristics: not requiring the user to program in a complex programming language not requiring the user to use GUI-like administration areas allow the user to "program" in a lightweight markup language The last point is not only about look and structure of the output but also about creating simple dynamic output. For example: listing pages fetching the content of other pages doing a site search and displaying its output dynamically show or hide parts of the page depending on login status Of course, I am not expecting a solution that provides the same possibilities like a Ruby, Python or PHP web framework. Rather I am looking for support of the "basics" that are common for web sites. Until now, I have found only one piece of software that fulfills these requirements but while it's free, it's not open source: BoltWire at http://www.boltwire.com/. Being open source is required.

  Read the article

• process taking high time on Itanium server

  - by Vishwa

  Hi, we have an application which is recently migrated from PA-Risk server to itanium server. After migration we noticed that there is significant increase in time taken to complete a process. when we tracked the time taken by each part we found that the system time is 7.590000, user time is 3.990000 but elapsed time is 70.434882!! Due to this huge elapsed time, overall performance of the application has gone very low. i wrote a small progrms to perform I/O operations on Itanium. these scripts working faster on Itanium compared to PA-Risk. what could be the reason for this high elapsed time of this process?

  Read the article

• android imageview scrolling to display full high quality image

  - by Javadid

  hi friends... i am working on displaying an image and placing an icon on top of it... clicking the icon will show an enlarged version of the image... though putting the imageview holding the image in a LinearLayout scales the image to the width of the dialog, the problem is that i need to display the image in a dialog but the image is very high resolution and hence is far bigger than the width of the dialog... I need to show the actual image with scrolling for both ways to see the whole image... But whenever i try putting the imageView in a scrollview the top of my imageview is blank... and again though image scrolls downwards the width is scaled to the width of the dialog... Helppppppppp guys....

  Read the article

• Drupal CMS most Stable for High Traffic

  - by Aditi

  Drupal users have high satisfaction with Drupal compared to the Joomla users, for a number of reasons. If you are thinking of  choosing a high performance platform to run your high traffic website.. Drupal Installation is your forte! Overload Scenario Drupal is scalable high performance CMS and is stable under heavy load. If your server is pushed beyond its capacity, Drupal shuts off gracefully and doesn’t crash. As soon as the server is back within its traffic capability, Drupal handles all requests smoothly again. For example if your dedicated server can handle a maximum of 50,000 visits a day, and on lucky days when your news created the buzz in social media and your traffic rose to 70,000 on one day, then your server will be overloaded and usually it crashes causing permanent damage to your database at times.. But if you have used Drupal CMS it closes down gracefully an as soon as traffic goes down to within the server’s capacity, the Drupal running site accepts all requests again. Extensibility Drupal users know that their add-ons integrate better with the core, and their framework makes it easier to extend their CMS’s capabilities.. which makes an extended version of it quite stable unlike Joomla, which loses its strength if you have plenty of plugins & heavy customizations running. Any CMS with number of plugins makes the content complex and reduces your ability to handle high traffic requests. Accessibility Management or ACL Chances are if you are high traffic website, you may have various users & content contributors. ACL means group roles that is assigning people out of the various registered user levels and allocating many kinds of privileges. The most common example is the ability to see or edit a section or selected pages. This efficient feature of Drupal makes it a class apart than other CMSs out there.

  Read the article

• C#: Streaming an Audio file from a Server to a Client

  - by Andreas Grech

  I am currently writing an application that will allow a user to install some form of an application (maybe a Windows Service) that will open a port on it's PC and given a particular destination on the hard disk, will then be able to stream mp3 files. I will then have another application that will connect to the server (being the user's pc) and be able to browse the hosted data by connecting to that PC (remotely ofcourse) given the port, and stream mp3 files from the server to the application I have found some tutorials online but most of them are about File Servers in C# and they download allow you to download a whole file. What I want is to stream an mp3 file so that it starts playing when a certain number of bytes are download (ie, whilst it is being buffered) How do I go about in accomplishing such a task? What I need to know specifically is how to write this application (that I will turn into a Windows Service later on) that will listen on a specified port a stream files, so that I can then access the files by something of the sort: http://<serverip>:65000/acdc/wholelottarosie.mp3 and hopefully be able to stream that file in a WPF MediaPlayer. [Update] I was following this tutorial about building a file server and sending the file from the server to the client. Is what I have to do something of the sort? [Update] Currently reading this post: Play Audio from a Stream using C# and I think it looks very promising as to how I can play streamed files; but I still don't know how I can actually stream the files from the server.

  Read the article

• How to play simultaneous multiply audio sources in Silverlight

  - by Shurup

  I want to play simultaneous multiply audio sources in Silverlight. So I've created a prototype in Silverlight 4 that should play a two mp3 files containing the same ticks sound with an intervall 1 second. So these files must be sounded as one sound if they will be played together with any whole second offsets (0 and 1, 0 and 2, 1 and 1 seconds, etc.) I my prototype I use two MediaElement (me and me2) objects. DateTime startTime; private void Play_Clicked(object sender, RoutedEventArgs e) { me.SetSource(new FileStream(file1), FileMode.Open))); me2.SetSource(new FileStream(file2), FileMode.Open))); var timer = new DispatcherTimer { Interval = TimeSpan.FromMilliseconds(1) }; timer.Tick += RefreshData; timer.Start(); } First file should be played at 00:00 sec. and the second in 00:02 second. void RefreshData(object sender, EventArgs e) { if(me.CurrentState != MediaElementState.Playing) { startTime = DateTime.Now; me.Play(); return; } var elapsed = DateTime.Now - startTime; if(me2.CurrentState != MediaElementState.Playing && elapsed >= TimeSpan.FromSeconds(2)) { me2.Play(); ((DispatcherTimer)sender).Stop(); } } The tracks played every time different and not simultaneous as they should (as one sound).

  Read the article

• Procesing 16bit sample audio

  - by user2431088

  Right now i have an audio file (2 Channels, 44.1kHz Sample Rate, 16bit Sample size, WAV) I would like to pass it into this method but i am not sure of any way to convert the WAV file to a byte array. /// <summary> /// Process 16 bit sample /// </summary> /// <param name="wave"></param> public void Process(ref byte[] wave) { _waveLeft = new double[wave.Length / 4]; _waveRight = new double[wave.Length / 4]; if (_isTest == false) { // Split out channels from sample int h = 0; for (int i = 0; i < wave.Length; i += 4) { _waveLeft[h] = (double)BitConverter.ToInt16(wave, i); _waveRight[h] = (double)BitConverter.ToInt16(wave, i + 2); h++; } } else { // Generate artificial sample for testing _signalGenerator = new SignalGenerator(); _signalGenerator.SetWaveform("Sine"); _signalGenerator.SetSamplingRate(44100); _signalGenerator.SetSamples(16384); _signalGenerator.SetFrequency(5000); _signalGenerator.SetAmplitude(32768); _waveLeft = _signalGenerator.GenerateSignal(); _waveRight = _signalGenerator.GenerateSignal(); } // Generate frequency domain data in decibels _fftLeft = FourierTransform.FFTDb(ref _waveLeft); _fftRight = FourierTransform.FFTDb(ref _waveRight); }

  Read the article

• AS3 microphone recording/saving works, in-flash PCM playback double speed

  - by Lowgain

  I have a working mic recording script in AS3 which I have been able to successfully use to save .wav files to a server through AMF. These files playback fine in any audio player with no weird effects. For reference, here is what I am doing to capture the mic's ByteArray: (within a class called AudioRecorder) public function startRecording():void { _rawData = new ByteArray(); _microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, _samplesCaptured, false, 0, true); } private function _samplesCaptured(e:SampleDataEvent):void { _rawData.writeBytes(e.data); } This works with no problems. After the recording is complete I can take the _rawData variable and run it through a WavWriter class, etc. However, if I run this same ByteArray as a sound using the following code which I adapted from the adobe cookbook: (within a class called WavPlayer) public function playSound(data:ByteArray):void { _wavData = data; _wavData.position = 0; _sound.addEventListener(SampleDataEvent.SAMPLE_DATA, _playSoundHandler); _channel = _sound.play(); _channel.addEventListener(Event.SOUND_COMPLETE, _onPlaybackComplete, false, 0, true); } private function _playSoundHandler(e:SampleDataEvent):void { if(_wavData.bytesAvailable <= 0) return; for(var i:int = 0; i < 8192; i++) { var sample:Number = 0; if(_wavData.bytesAvailable > 0) sample = _wavData.readFloat(); e.data.writeFloat(sample); } } The audio file plays at double speed! I checked recording bitrates and such and am pretty sure those are all correct, and I tried changing the buffer size and whatever other numbers I could think of. Could it be a mono vs stereo thing? Hope I was clear enough here, thanks!

  Read the article

• Our Flash Streaming Player Occasionally Stutters like a Skipping CD after a Period of Time

  - by Jonathan Fritz

  We offer a streaming player for a number of our clients, who are responsible for their providing us with their own audio streams. We have written a very simple flash player that can play all of the streams that we support (icecast/shoutcast/live365/mp3 over http/etc). Unfortunately, we have found that when listening, our player sometimes begins to stutter (like a skipping cd), sometimes after only 10 minutes, and sometimes after an hour of listening. We have noticed this behaviour in firefox on both linux and windows. Does anybody know anything about this problem? We know that flash isn't ideal for infinite streams of audio, but it's about all that we can find that's on every platform out there. If anybody can suggest a solution to our problem, I'll be your friend forever. Here is a link to the live player: http://cr-jf.jfritz.02.dev.wecreate.com/streaming/player_v5/ Note that you'll need to test in a browser that isn't IE, because we use WMP in IE, and that the JavaScript on the page will cause the player to unload and re-load once an hour because of memory issues. Because I can only put one hyperlink in a post, I'll add a link to the player source code as a comment. Thanks all!

  Read the article

• Where is /dev/dsp or /dev/audio?

  - by YumYumYum

  I have to apply sudo chmod a+r /dev/dsp or /dev/audio but in my Ubuntu 12.10 i do not have such. Where is then the PCM sound file for ssh? chmod: cannot access `/dev/dsp': No such file or directory chmod: cannot access `/dev/audio': No such file or directory Follow up: http://superuser.com/questions/244173/missing-dev-dsp-under-ubuntu I want to stream the sound output and input. So that i can capture any audio in/out to a file for recording.

  Read the article

• AAC.js : le décodeur audio JavaScript open source supporte le profile Low Complexity

  AAC.js : le dernier décodeur audio JavaScript de Official.fm Labs qui supporte le profile Low Complexity [IMG]http://media.tumblr.com/tumblr_m6wpozHbxB1qbis4g.png[/IMG] L'équipe de Official.fm Labs vient de sortir un codec audio qui pourrait d'ailleurs être le prochain codec le plus utilisé après le MP3, voire le surpasser. AAC.js est entièrement codé en JavaScript avec le framework Aurora.js qui facilite l'écriture de codecs. AAC, qui signifie Advanced Audio Codec, est l'un des codecs les plus courants et des noms comm...

  Read the article

• Amnesia doesn't start due to audio problems

  - by james

  I have a problem with amnesia game. After Intro and clicking continue button few times, when game is supposed to start it crashes. Here is console output: ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib audio/pcm_bluetooth.c:1614:(audioservice_expect) BT_GET_CAPABILITIES failed : Input/output error(5) ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:1018:(snd_pcm_dmix_open) unable to open slave 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 I should mention I have integrated both graphic and sound card.

  Read the article

• asus n550jv audio problem: no sound from notebook' speakers

  - by skywalker

  Ubuntu 13.10. The problem is: the internal speakers don't work. I have no problem when I'm using the headphones. There is no hardware issue since in windows 8 everything works perfectly(external subwoofer included). I'm trying to modify /etc/modprobe.d/alsa-base.conf but I can't find the correct model to put into: options snd-hda-intel model= The file HD-Audio-Models.txt doesn't contain the model for ALC668. Some info: :~sudo aplay -l **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC668 Analog [ALC668 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 :~$ sudo lspci -v | grep -A7 -i "audio" 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) Subsystem: Intel Corporation Device 2010 Flags: bus master, fast devsel, latency 0, IRQ 52 Memory at f7a14000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Kernel driver in use: snd_hda_intel -- 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) Subsystem: ASUSTeK Computer Inc. Device 11cd Flags: bus master, fast devsel, latency 0, IRQ 53 Memory at f7a10000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel PS info :~$ amixer -c 0 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] :~$ pacmd dump-volumes Welcome to PulseAudio! Use "help" for usage information. Sink 0: reference = 0: 76% 1: 76%, real = 0: 76% 1: 76%, soft = 0: 100% 1: 100%, current_hw = 0: 76% 1: 76%, save = yes Input 8: volume = 0: 100% 1: 100%, reference_ratio = 0: 100% 1: 100%, real_ratio = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, volume_factor = 0: 100% 1: 100%, volume_factor_sink = 0: 100% 1: 100%, save = no Source 0: reference = 0: 100% 1: 100%, real = 0: 100% 1: 100%, soft = 0: 100% 1: 100%, current_hw = 0: 100% 1: 100%, save = no Source 1: reference = 0: 16% 1: 16%, real = 0: 16% 1: 16%, soft = 0: 100% 1: 100%, current_hw = 0: 16% 1: 16%, save = yes

  Read the article

• Google I/O 2010 - Advanced Android audio techniques

  Google I/O 2010 - Advanced Android audio techniques Google I/O 2010 - Advanced Android audio techniques Android 301 Dave Sparks In this session, we will explore advanced techniques that you can employ in your apps when working with media. This includes using Android's low-level audio APIs, selecting the appropriate format for your media files, and what's now possible using new media framework APIs introduced in Android 2.2. For all I/O 2010 sessions, please go to code.google.com From: GoogleDevelopers Views: 3 0 ratings Time: 57:16 More in Science & Technology

  Read the article

• Shortcut to switch between Analog Stereo output & HDMI audio output

  - by iJeeves

  To switch to HDMI audio output (of monitor) and back to normal audio output from system audio jack (for headphones, as my monitor doesn't have audio out), I find myself opening up sound preferences and selecting the right channel everytime. Is there any way I can create a toggle button in the panel or assign some shortcut key to toggle since I do the switching so often. :aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 7: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0

  Read the article

• Streaming audio from a webpage

  - by luca590

  I want to be able to stream audio from another webpage through mine, but i do not know how to find the url for each audio file located on a separate webpage. It would also be extremely helpful to do everything in bulk so instead of writing a separate line of code for each audio file, simply writing a few lines of code to upload links to 100 audio files, etc. I am also using Ruby on Rails for my webpage. How do you find a file located on a separate webpage? Does anyone know, if possible how, to upload file links in bulk?

  Read the article

• SAPPHIRE HD 7770 no audio on HDMI TV display

  - by zeroconf

  I have SAPPHIRE HD 7770 and cannot get work audio over HDMI. http://www.sapphiretech.com/presentation/product/?cid=1&gid=3&sgid=1159&lid=1&pid=1452&leg=0 I use Ubuntu 12.04 LTS 64-bit version with all current updates. I tried at /etc/default/grub: GRUB_CMDLINE_LINUX_DEFAULT="quiet splash radeon.audio=1" ... it didn't help. It's probably I use proprietary driver -this seems to be open source driver. I use the driver, what jockey-gtk (additional drivers) offered me: ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER <---- I installed that one ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER (post-release update) So - I installed the first one, because installing second version failed. Everything went fine but no sound at TV display by HDMI. Even Gnome sound mixer doesn't show HDMI choice. Using 32" Samsung B530 LCD TV - http://www.lcdbesttv.com/2010/02/samsung-b530-series-lcd-tv/ I have Asus P8Z77-M motherboard - http://www.asus.com/Motherboards/Intel_Socket_1155/P8Z77M/ - there is also HDMI integrated. When I put HDMI cord to that plug, then even Gnome sound mixer showed HDMI audio but it didn't work. I have set from BIOS, that I use that SAPPHIRE HD 7770 from PCIe. My lspci output: 00:00.0 Host bridge: Intel Corporation Ivy Bridge DRAM Controller (rev 09) 00:01.0 PCI bridge: Intel Corporation Ivy Bridge PCI Express Root Port (rev 09) 00:02.0 Display controller: Intel Corporation Ivy Bridge Graphics Controller (rev 09) 00:14.0 USB controller: Intel Corporation Panther Point USB xHCI Host Controller (rev 04) 00:16.0 Communication controller: Intel Corporation Panther Point MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #2 (rev 04) 00:1b.0 Audio device: Intel Corporation Panther Point High Definition Audio Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 1 (rev c4) 00:1c.5 PCI bridge: Intel Corporation Panther Point PCI Express Root Port 6 (rev c4) 00:1c.6 PCI bridge: Intel Corporation 82801 PCI Bridge (rev c4) 00:1d.0 USB controller: Intel Corporation Panther Point USB Enhanced Host Controller #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Panther Point LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation Panther Point 6 port SATA Controller [AHCI mode] (rev 04) 00:1f.3 SMBus: Intel Corporation Panther Point SMBus Controller (rev 04) 01:00.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Device 683d 01:00.1 Audio device: Advanced Micro Devices [AMD] nee ATI Device aab0 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 09) 04:00.0 PCI bridge: ASMedia Technology Inc. ASM1083/1085 PCIe to PCI Bridge (rev 03)

  Read the article

• Unable to configure/setup 5.1 audio with 12.04

  - by Vipin Vinayan

  I am kinda new to Ubuntu as well. I have been having this issue with audio for quite sometime now. Initially, when I installed version 11.10 (I guess), I was able to use my 5.1 speakers without any issues. If my memory serves me right, it was after an update that the 5.1 audio stopped working and the video resolution would not get saved. I temporarily fixed the resolution issue by creating a start-up shell script that would update the resolution and load it. But the issue with audio has been going on for quite sometime now. Even though I have option for 5.1, only two speakers seem to be working. I thought an upgrade should fix the issue and so upgraded the OS to version 12.04. I also tried uninstalling alsa and pulse audio, reinstalling them, changing the /etc/pulse/daemon.conf channels from 2 to 6. I have also tried installing pavucontrol but nothing seems to have worked and the issue still persists. Is there anything else you could suggest? The lspci log on my computer is as follows 00:00.0 Host bridge: Intel Corporation 82G33/G31/P35/P31 Express DRAM Controller (rev 10) 00:01.0 PCI bridge: Intel Corporation 82G33/G31/P35/P31 Express PCI Express Root Port (rev 10) 00:02.0 VGA compatible controller: Intel Corporation 82G33/G31 Express Integrated Graphics Controller (rev 10) 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01) 00:1c.0 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 1 (rev 01) 00:1c.1 PCI bridge: Intel Corporation N10/ICH 7 Family PCI Express Port 2 (rev 01) 00:1d.0 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #1 (rev 01) 00:1d.1 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #2 (rev 01) 00:1d.2 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #3 (rev 01) 00:1d.3 USB controller: Intel Corporation N10/ICH 7 Family USB UHCI Controller #4 (rev 01) 00:1d.7 USB controller: Intel Corporation N10/ICH 7 Family USB2 EHCI Controller (rev 01) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation N10/ICH7 Family SATA Controller [IDE mode] (rev 01) 00:1f.3 SMBus: Intel Corporation N10/ICH 7 Family SMBus Controller (rev 01) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) Would really appreciate a response that will assist me in resolving my issue. Thanks in advance Vipin

  Read the article

• Vista Core Audio API Master Volume Control

  A managed wrapper for accessing the Vista Core Audio API

  Read the article

• Distorted choppy audio in Precise

  - by Misery

  After installing Precise on my PC, some problems with soud occure. While using Lucid there were no problems. The sound is choppy and distorted in low tones range. As I absolutely have no experience in setting/testing and doing anything with Audo Devices I need help even to diagnose the problem. update: sudo lshw -c multimedia *-multimedia description: Audio device product: Radeon X1200 Series Audio Controller vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 5.2 bus info: pci@0000:01:05.2 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi bus_master cap_list configuration: driver=snd_hda_intel latency=32 resources: irq:19 memory:fdafc000-fdafffff *-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 update 2: It has something to do with the volume. If the audio is quiet it is not choppy, if the sound is loud then it begins to be choppy. Regards, Misery

  Read the article

• Audio not working in 12.10

  - by frampy

  I did a clean install of 12.10, when I open Sound Settings in gnome the only device in the list is "Dummy Output", and sound is not working. Sound worked fine out of the box in 12.04 I ran alsamixer, it says my card is "HDA Intel", and chip is "Realtek ALC880". The alsamixer playback output was set to mute at first, unmuting did not fix. I checked out the info at http://www.unixmen.com/2012003-howto-resolve-nosound-problem-on-ubuntu/ as suggested on a similar question, I've done everything there except installing the ubuntu audio dev team driver. Should I try install this? Edit: I've been reading the sound troubleshooting guide at https://help.ubuntu.com/community/SoundTroubleshooting It looks like Ubuntu is finding my audio device correctly. mike@wucade:~$ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) Subsystem: Albatron Corp. Device 2668 Flags: bus master, fast devsel, latency 0, IRQ 40 Memory at d01c0000 (64-bit, non-prefetchable) [size=16K] Capabilities: Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel Still stuck as to why this isn't working.

  Read the article

• Synced audio ouput on multiple machines? VLC? hardware solutions?

  - by zimmer62

  I'm wondering if there is any software or hardware solutions to synced audio or audio and video across multiple computers or devices on a network. I've seen Sonos, and it might be a good solution, but it's also a very expensive solution. I'd like to be able to play something with realtime audio output on one PC, but hear it on speakers throughout the house, being it the home theater receiver, or another computer in another room. I saw a solution using the apple iport express, but the latency was unacceptable for anything other than just music. I'd like to avoid running audio wires with baluns to a bunch of amplifiers scattered all over the place when I have cat5 run everywhere. Is anyone familiar with using this kind of process for whole home audio? The latency is a big deal for me, if I've got video attached to the sound (e.g. watching a hockey game)

  Read the article

< Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >