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  • No audio on an HP dm4

    - by Haze1
    I just got a new laptop, HP dm4, and I'm having problems getting the audio to work properly on it. http://www.alsa-project.org/db/?f=7b697a35465a9f7236fb94deb9ff97fa65e55489 I tried to edit /etc/modprobe.d/alsa-base.conf and added: option snd-hda-intel model=ref this caused the audio to work, but it's muffled. I'm wondering if anybody knows what would be the correct options to get this POS to work.

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  • Is it possible to play multiple audio streams from one "jukebox" to multiple Airport Express devices?

    - by Alex Reynolds
    I have set up a Mac mini as a jukebox that streams audio to an Airport Express in another room in the house, using the AirPlay/AirTunes feature in iTunes. I control this with the iOS Remote app, and this works great. At the present time, it looks like the Mac mini's copy of iTunes gets taken over by the Remote app, while streaming. If I set up a second Airport Express in room B, is there a way to set it up (as well as the jukebox) so that it can receive and play its own unique music stream ("stream B"), separate from what's going on at the Mac mini, or in room A, which is playing stream A? To accomplish this, I would be happy to buy a copy of Rogue Amoeba's AirFoil if it will allow sending multiple, separate audio streams from one computer to the multiple wireless bridges, while using the Remote app (or a Rogue Amoeba equivalent for iOS). However, it is unclear to me from their site documentation, whether that is possible or not. I'd prefer to give the points to an answer that solves this problem. If you don't know if it can be done, or do not think it can be done, please allow others to answer. I appreciate your help. Thanks for your advice.

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  • Setup for a live (low-latency) audio video broadcast over Wi-Fi?

    - by Majal Mirasol
    The Upgrade We are capturing audio (from mixer) and video (from a camera) from a main auditorium and passing it to separate rooms within the building. We used to have done this via manual audio/video cables and wires. We wanted to "upgrade" the system and wirelessly broadcast the stream via Wi-Fi. The Problem In our current setup (Wirecast running on A10 on a Wireless-N network), we have the problem of delay. Our streams are delayed from a minute up to five minutes on the clients (laptop/iPad/Android). This had not been a problem from the previous wired connections. Since the wireless network is local, we thought that a delay of less than a second should be achievable. Our Question And so it goes. Anybody there who has any experience for a setup that has both low latency and at the same time user-friendly to clients streaming in the program? Any recommendations would be highly appreciated. (Our current setup in on Windows 7, but setup on a dedicated Linux box is preferred, if achievable.)

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  • iphone audio streaming

    - by mobapps99
    Hi , i'm developing an application which uses audio streaming. For streaming audio from internet i'm using the AudioStreamer class. The audio streamer has four state isPlaying, isPaused ,isWaiting, and isIdle . My problem is that when the audio streamer is in the state "isWaiting" and at that time if i get a phone call Audio queue fails giving the error "Audio queue start failed." Any has solution for this? help....

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  • What decent audio recording interfaces are well supported in Windows 7 64bit?

    - by labradort
    I currently have an Audigy 2 ZS Platinum. It permits me to insert a 1/4" jack line from bass guitar and play along with pre-recorded piano music. This worked fine under Windows XP. I am moving to Windows 7 64 bit (dual boot for now), and Creative may not develop fully working drivers for this component. Looking around, I don't see Windows 7 support mentioned at product web sites from E-MU, Roland, M-Audio, etc. Even at Creative, the posting of available drivers for Windows 7 is deceptive, as they do not adequately support recording (latency, distortion). My local music store shrugs and says to stay with Win XP. In some cases, the Vista drivers will work in Win 7. So I need real world feedback on this. I should also mention I'm not impressed with available USB interfaces - they have too low of a signal to noise ratio for my purposes. That leaves PCI, or possibly firewire devices (never tried one yet).

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  • what's the "best" approach to creating the UI of an audio plugin that will be both audio unit and VST for OS X and Windows?

    - by SaldaVonSchwartz
    I'm working on a couple audio plugins. Right now, they are audio units. And while the "DSP" code won't change for the most part between implementations / ports, I'm not sure how to go about the GUI. For instance, I was looking at the Apple-supplied AUs in Lion. Does anyone know how did they go about the UI? Like, are the knobs and controls just subclasses of Cocoa controls? are they using some separate framework or coding these knobs and such from scratch? And then, the plugs I'm working on are going to be available too as VSTs for Windows. I already have them up and running with generic interfaces. But I'm wondering if I should just get over it and recreate all my interfaces with the vstgui code provided by Steinberg or if there's a more practical approach to making the interfaces cross-platform.

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  • audio frameworks in iPhone

    - by suse
    Hello, I would like to know the follwing information about iPhone audio system Heirarchy of the audio framework in iPhone OS. i know that there are 3 main audio frameworks in iPhone OS.i.e AVFoundation Framework CoreAudio Framework OpenAL Framework what are the audio formats supported in each of the above framework?I mean will all the framework support all audio formats or are they dependent about the audio formats it support? Thank You

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  • AAC Sample Rate and Bit Rate for High Quality Audio?

    - by marco.ragogna
    What are the AAC Sample Rate and Bit Rate settings to set in order to encode an audio track with a quality comparable to MP3 320kbps? I need to backup a DVD movie, the default settings for AAC are Bitrate (KB/s) 128 Sample Rate (HZ) 44100 should I set Bitrate (KB/s) 320 Sample Rate (HZ) 48000 or the default are already good?

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  • Bose USB audio: crackling popping sound, eventually die

    - by Richard Barrett
    I've been trying to troubleshoot this issue for a while now. Any help would be much appreciated. I'm having trouble getting my Bose "Companion 5 multimedia speakers" working with my installation of Ubuntu 12.04 (link to Bose product here: http://www.bose.com/controller?url=/shop_online/digital_music_systems/computer_speakers/companion_5/index.jsp ). The issue seems to be low level (not just Ubuntu). What happens: When I boot into Ubuntu, I can get Rhythm box to play ok. However, if I try anything else (an .avi file, a webpage, or Clementine player with mp3 files) I get crackling, popping, or choppy sounds. If I move the mouse around, especially if it seems graphic intensive, the problem gets worse (more crackling noises). The more taxing it appears to be, the more likely it is that the sound will just die altogether until I reboot. For some reason the videos at www.bloomberg.com seem especially bad for it (my sound normally goes dead in under 45 seconds and won't work until reboot). Both my desktop running Ubuntu 12.04 and my laptop (running the same) have the same crackling problem. Troubleshooting so far: A friend of mine who knows linux well tried to solve it for me without any luck. He took pulseaudio out of the equation, but still had the problem just using AlSA. Among the many things he tried was adjusting the latency, but that didn't help either. I've also tried things like adjusting the USB device settings in the config file from -2 to -1 so that it will use my USB sound and I also commented out the lines that would stop that. These don't do anything. (That really seems like it's for someone who is getting no sound at all, so it's not surprising this won't work.) My friend's laptop running his Archlinux could play my Bose USB speakers without any problems. I also tried setting my daemon.conf file to use 6 channels (based on this http://lotphelp.com/lotp/configure-ubuntu-51-surround-sound ) but that didn't work either. I recently used a DVD to boot into Ubuntu Studio 12.04 (because it uses a live audio kernel) and this happened: I got perfect sound for a minute or two When I started moving windows around while sound was playing, the sound died again. Perhaps more interesting: There is a headphone out jack on the Bose system. When I use it, the audio is perfect for all applications (even the deadly bloomberg.com videos with .avi playing at the same time and moving around windows). Also, there is an audio-in jack on the Bose system. I can use a male-to-male mini jack to go from my soundcard's output to the Bose input and then all sound works perfectly. -However, it still requires the Bose to be plugged in to USB, otherwise I lose all sound. Any thoughts? Any suggestions for trouble shooting? (Or any suggestions for somewhere else to post to solve this?) Any logs or other files I can provide to help someone help me work this out? Your help is much appreciated! Rick BTW: I sometimes get people posting responses like "My Bose USB system works great with Ubuntu 12.04," without any more details. Is there anything I should ask such people to narrow down my problem? (It's kind of annoying to hear such a response because it doesn't help solve my problem.)

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  • Linux program to convert audio file of fax transmission to image?

    - by bdk
    I have a number of uncompressed audio files recorded off of an analog (POTS) telephone line of fax transmissions. Is there a Linux utility or library I could use to convert these files into images of the fax they contain? I'm not looking to send/receive a fax via a modem, but just to "replay" the communications tones and parse out the fax message.I'm guessing this may not be possible due to duplex issues and not knowing which end of the conversation is sending what,but thought I'd ask to see if anyone knew of something.

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  • How can I split the 5.1 audio channels from an AC3 file into individuals streams (preferably on a Ma

    - by Drarok
    I have a file that I've pulled from a DVD that is apparently in AC3 5.1 format. The extension is .AC3 and it opens an plays in QuickTime, VLC etc. What I want is each individual channel in a separate file, but I can't seem to find any tools that will allow be to do that. Is there a way to split the file I have, or alternatively is there a way to pull the audio streams from a 5.1 DVD?

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  • Why do my 3GP videos not play audio using VLC?

    - by GiH
    I have a Nexus One phone and when I record video the container seems to be 3GP. When I try to playback using VLC I'm getting no audio, and an error saying that there is nothing I can do because VLC does not support the "samr" codec. Is there really nothing I can do to watch my videos on VLC? If not, whats the alternative? I really like VLC specifically because I never have to download codecs...

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  • Why is preserving the pitch in audio playback (allegedly) less performant?

    - by Markus Unterwaditzer
    In VLC for Android, i discovered an option to preserve the pitch during faster-than-normal playback: The "requires a fast device" obviously implies that faster playback is more performant when the pitch is changed too. Why is that so? What i've tried: Before posting this question i did some shallow research through Google. According to Wikipedia, there are several methods for faster playback of audio, the "simplest" one (Resampling) changes the pitch.

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  • How much audio latency is noticeable by human brain?

    - by Borek
    I am choosing a wireless headset for my PC (I hate cables) and am looking at Sennheiser RS 170 / 180. They supposedly sound great, however, there is a 25ms audio latency. I've heard that this is OK when watching TV or listening musing but is bad for games. The question is - has there been any research / hard data that would show how much of a delay is noticeable by human brain? 25ms doesn't sound like a lot but I may be wrong.

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  • ATI HDMI Audio disappears in Windows 7 when a TV is connected

    - by jsalonen
    So far I have unsuccessfully googled for HOURS with no luck fixing this very annoying problem. The settings is the following: I have PC running Windows 7 RC (64-bit) Video card is a ATI Radeon 4850 series card (Sapphire HD 4850 512MB to be exact) The video card has HDMI out with built-in audio chip I have an HDMI cable connecting the PC to a TV (Sony Bravia series) The problem is that when I connect the HDMI cable to the TV, the ATI HDMI Sound output device disappears completely from the list of playback devices in Windows. As a workaround I can restore the audio by re-installing the HDMI audio driver. However, when I disconnect the TV the driver disappears again. So basically, every time I want to watch stuff on my TV, I have to reinstall audio driver, which of course is VERY annoying. EDIT: I have figured out that I do not need to re-istall the HDMI audio driver to restore sound; I only need to reboot my computer with the HDMI cable plugged in to restore the audio driver. This suggests that the problem has something to do with information passed from TV to computer, which makes my HDMI Audio driver disappear. Are there any other, more elegant workarounds for this problem? All help is much appreciated!

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  • Can I boot up a virtual machine natively?

    - by Anshul
    My question is: Is is possible to run a virtual machine natively on your hardware if you have installed the proper drivers etc? In other words, can I use a VHD as a regular hard drive to boot from? The reason I want to do this is that I do both graphics-intensive and audio-intensive work, but my computer is not powerful enough to handle both at the same time and many times I install a bunch of audio programs that I don't want affecting the stability of my graphics programs. Basically I wanted to have sandboxing between the two sets of applications. So I tried running the graphics-intensive programs in a VirtualBox VM and the audio-intensive work natively (simply because it's a pain to route ASIO audio devices in/out of VirtualBox). This kind-of works - the graphics-intensive stuff is tolerable, but still relatively slow, because it's running inside a VM. So my next idea was to just dual-boot and install the graphics and audio programs in separate partitions but I frequently use them in tandem, so it wouldn't be practical to reboot my machine every time I need to use the other set of programs. But I could live with this scenario: If I need to do more audio-intensive stuff, I'll just boot up to the audio partition and run the graphics programs in a VM, and then when I'm working heavily on the graphics part, I'll just boot the graphics partition as a regular OS directly on the hardware. Is this possible? For example by booting up a VHD as a regular hard drive? Or by setting up dual-boot, and every time the audio partition is shut down, synchronize the graphics VM VHD with the native graphics partition? Is it practical, given the above scenario? And if it's not possible, barring buying another computer, can anyone suggest a best-of-all-worlds setup (the two worlds being performance, sandboxing, and running in parallel) for the above scenario? Thanks in advance.

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  • Objective-c - How to serialize audio file into small packets that can be played?

    - by vfn
    Hi there, So, I would like to get a sound file and convert it in packets, and send it to another computer. I would like that the other computer be able to play the packets as they arrive. I am using AVAudioPlayer to try to play this packets, but I couldn't find a proper way to serialize the data on the peer1 that the peer2 can play. The scenario is, peer1 has a audio file, split the audio file in many small packets, put them on a NSData and send them to peer2. Peer 2 receive the packets and play one by one, as they arrive. Does anyone have know how to do this? or even if it is possible? EDIT: Here it is some piece of code to illustrate what I would like to achieve. // This code is part of the peer1, the one who sends the data - (void)sendData { int packetId = 0; NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:@"myAudioFile" ofType:@"wav"]; NSData *soundData = [[NSData alloc] initWithContentsOfFile:soundFilePath]; NSMutableArray *arraySoundData = [[NSMutableArray alloc] init]; // Spliting the audio in 2 pieces // This is only an illustration // The idea is to split the data into multiple pieces // dependin on the size of the file to be sent NSRange soundRange; soundRange.length = [soundData length]/2; soundRange.location = 0; [arraySoundData addObject:[soundData subdataWithRange:soundRange]]; soundRange.length = [soundData length]/2; soundRange.location = [soundData length]/2; [arraySoundData addObject:[soundData subdataWithRange:soundRange]]; for (int i=0; i // This is the code on peer2 that would receive an play the piece of audio on each packet - (void) receiveData:(NSData *)data { NSKeyedUnarchiver* unarchiver = [[NSKeyedUnarchiver alloc] initForReadingWithData:data]; if ([unarchiver containsValueForKey:PACKET_ID]) NSLog(@"DECODED PACKET_ID: %i", [unarchiver decodeIntForKey:PACKET_ID]); if ([unarchiver containsValueForKey:PACKET_SOUND_DATA]) { NSLog(@"DECODED sound"); NSData *sound = (NSData *)[unarchiver decodeObjectForKey:PACKET_SOUND_DATA]; if (sound == nil) { NSLog(@"sound is nil!"); } else { NSLog(@"sound is not nil!"); AVAudioPlayer *audioPlayer = [AVAudioPlayer alloc]; if ([audioPlayer initWithData:sound error:nil]) { [audioPlayer prepareToPlay]; [audioPlayer play]; } else { [audioPlayer release]; NSLog(@"Player couldn't load data"); } } } [unarchiver release]; } So, here is what I am trying to achieve...so, what I really need to know is how to create the packets, so peer2 can play the audio. It would be a kind of streaming. Yes, for now I am not worried about the order that the packet are received or played...I only need to get the sound sliced and them be able to play each piece, each slice, without need to wait for the whole file be received by peer2. Thanks!

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  • Upscaling audio from 2.1 to 5.1 in Windows 7

    - by Darth Android
    I'm currently using the onboard sound on my Asus P6T6 WS Revolution motherboard (SoundMAX Integrated Digital Audio) and was wondering if there was any way to make either windows or the audio drivers upscale 2-channel audio to 5-channel audio (basic duplication would suffice)? I was using a creative sound card but got fed up with the memory leaks and poor sound quality.

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  • NO AUDIO DEVICE

    - by Paul
    my computer have no audio is says that NO AUDIO DEVICE. I already installed Realtek AC97 and Realtek High Definition Audio Driver and I also paste stream.dll to windows folder and system32 folder and i restart my computer but it says NO AUDIO DEVICE. Pls help me ASAP..TANX

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  • Extract audio files from powerpoint

    - by curious2know
    I recorded a some audio files on my powerpoint presentation. This was done in two ways: (1) for some slides I used the record narration feature of powerpoint (the audio on each slide was recorded separately) and, (2) for others I used audacity to record the audio, which I imported into powerpoint. I need to extract the audio file from each slide. I need to send just the audiofiles to someone. Is there a way I can extract the audiofiles? Thanks

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