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  • Can all notebooks record from the sound card?

    - by Stefan Walter
    Is it possible that a modern notebook (say built in the last 5 years) contains a (onboard) sound card that it is not possible to record from? I've always assumed that this is a very basic capability, but since I can't get it to work, I would like to check that before I put any more effort into it. In case it matters, my notebook is an ASUS UL50VT, I'm trying with Sound Recorder and Audacity under Ubuntu, but I am, at this point, not asking for "Try that"-support or software recommendations.

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  • Does this video card support sound?

    - by Macros
    Probably a rookie question but here goes...I am looking to buy a new video card for a few year old PC which will be used as a media centre. The card I am looking at is this one http://www.ebuyer.com/product/173708, with the main aim being to play blu-ray films. In the product description it states that the card has 7.1 audio channel support, does this mean it will play the sound from the blu-ray through the HDMI, or do I need a separate sound card?

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  • Sound has stopped working after upgrading to Windows 8

    - by Max
    I upgraded my (fairly new, about 2 months old) computer to Windows 8 last night, and now the sound has stopped working across the entire machine. I have checked using multiple programs (iTunes, YouTube, World of Warcraft, etc) and multiple output devices (speakers and headset), checked the soundcard drivers (Asus Xonar DG 5.1), checked the volume mixer to ensure I wasn't just having a brainfart and had the sound muted, but nothing's working. Does anyone have any advice on what could be causing this? Thanks Max

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  • How do I get 5.1 surround sound working on an Acer Aspire 5738ZG?

    - by kbargais_LV
    I got a problem with sound. I tried everything but no results. :( I got 3 sound ports. my daemon: # This file is part of PulseAudio. # # PulseAudio is free software; you can redistribute it and/or modify # it under the terms of the GNU Lesser General Public License as published by # the Free Software Foundation; either version 2 of the License, or # (at your option) any later version. # # PulseAudio is distributed in the hope that it will be useful, but # WITHOUT ANY WARRANTY; without even the implied warranty of # MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU # General Public License for more details. # # You should have received a copy of the GNU Lesser General Public License # along with PulseAudio; if not, write to the Free Software # Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 # USA. ## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for ## more information. Default values are commented out. Use either ; or # for ## commenting. ; daemonize = no ; fail = yes ; allow-module-loading = yes ; allow-exit = yes ; use-pid-file = yes ; system-instance = no ; local-server-type = user ; enable-shm = yes ; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB ; lock-memory = no ; cpu-limit = no ; high-priority = yes ; nice-level = -11 ; realtime-scheduling = yes ; realtime-priority = 5 ; exit-idle-time = 20 ; scache-idle-time = 20 ; dl-search-path = (depends on architecture) ; load-default-script-file = yes ; default-script-file = /etc/pulse/default.pa ; log-target = auto ; log-level = notice ; log-meta = no ; log-time = no ; log-backtrace = 0 resample-method = speex-float-1 ; enable-remixing = yes ; enable-lfe-remixing = no flat-volumes = no ; rlimit-fsize = -1 ; rlimit-data = -1 ; rlimit-stack = -1 ; rlimit-core = -1 ; rlimit-as = -1 ; rlimit-rss = -1 ; rlimit-nproc = -1 ; rlimit-nofile = 256 ; rlimit-memlock = -1 ; rlimit-locks = -1 ; rlimit-sigpending = -1 ; rlimit-msgqueue = -1 ; rlimit-nice = 31 ; rlimit-rtprio = 9 ; rlimit-rttime = 1000000 ; default-sample-format = s16le ; default-sample-rate = 44100 ; default-sample-channels = 6 ; default-channel-map = front-left,front-right default-fragments = 8 default-fragment-size-msec = 10 ; enable-deferred-volume = yes ; deferred-volume-safety-margin-usec = 8000 ; deferred-volume-extra-delay-usec = 0

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  • Stepmania + KDE4 = sound problem

    - by picca
    I cannot manage to get KDE4 + stepmania working. If I run StepMania I always get: StepMania 3.9 Log starting 2010-12-24 14:52:48 Loading window: gtk OS: Linux ver 020636 Crash backtrace component: x86 custom backtrace Crash lookup component: dladdr Crash demangle component: cxa_demangle Runtime library: glibc 2.11.2 Threads library: NPTL 2.11.2 TLS is available ALSA: Advanced Linux Sound Architecture Driver Version 1.0.23. ALSA Driver: 0: HDA ATI SB [SB], device 0: STAC92xx Analog [STAC92xx Analog], 0/1 subdevices avail ALSA Driver: 0: HDA ATI SB [SB], device 1: STAC92xx Digital [STAC92xx Digital], 1/1 subdevices avail Couldn't load driver ALSA: dsnd_pcm_open(hw:0): Device or resource busy Mixing 0.000000 ahead in 0 Mix() calls Couldn't load driver ALSA-sw: dsnd_pcm_open(hw:0): Device or resource busy Mixing 0.000000 ahead in 0 Mix() calls Couldn't load driver OSS: RageSound_OSS: Couldn't open /dev/dsp: Device or resource busy Language: english Theme: default Error: Couldn't find a sound driver that works I found that in StepMania/Data/StepMania.ini I should add following line: SoundDevice=default That enables me to run StepMania, but I don't have any sound. Which is pretty bad for an application like this one. I'm quite sure that the problem is in phonon that is blocking the audio device to which StepMania needs to access directly. I think that I can fix this if I run other (lighter) window-manager than KDE4. But that is not a solution occasional linux user. Do I have any chance to get StepMania under KDE4 completely working?

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  • No sound out of headphone port

    - by Thanatos
    I cannot get sound out of the headphone port. Headphones are plugged in, and sound comes out of the internal speakers. Windows behaves normally (sound switches to headphones when headphones are inserted). It did work in Linux at one point, but something changed, we're just not sure what. Rebooting doesn't fix. This appears to occur whether or not PulseAudio is running. Things I've tried: Rebooting. No effect. Booting into Windows. It works properly, so probably not a hardware issue. All of alsamixer. My only controls are this: "Master" Volume bar & mutable, unmuted. Controls volume. "PCM" Volume bar only. 100%. "S/PDIF" Mutable only, currently muted, has no effect. "S/PDIF" Default PCM", Mutable only, currently unmuted, has no effect. Killing PulseAudio. No effect. (It also won't stay dead! Something appears to be restarting it, and I can't tell what, but it is annoying as fuck.) alsactl init 0, no effect. sudo rm -f /var/lib/alsa/asound.state, no effect. General system info: Ubuntu 10.04 LTS Toshiba Satellite T135D-S1324 lspci says I have: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller

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  • How to stream sound from an esx virtual machine ?

    - by Adrien
    Hi, I try to play sound from my monitoring application, which is an xp virtual machine on VMware ESX 3, on a physical machine with a real sound card, but I can not add from the ESX console sound card. Currently, I transmit sound with opening an RDP session and play it in this session. I would like to play sound without openning RDP session, do you have a solution to add a virtual sound card and then stream it with vlc?

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  • How can I add a custom item to the Sound Indicator (and make it clickable more than once)?

    - by con-f-use
    The original question One of the strength of Unity are the various standardized indicators. I want to customize the sound indicator with an additional menu entry that runs a small shell script. I'm not afraid of a little Python code and I hope someone can point me to the right subroutine in the right file. I suspect that will be fairly easy but all the indicators are just so bloated that I can't look through their code in a reasonable time. Any help is appreciated. I know it is possible as the marvelous Skype-Wrapper does it. Edit 2 - Now a dirty DBus hack The one click problem from one edit before has now turned into a DBus problem. Basically we have to tell the sound indicator that our bogus player has terminated now. A dirty hack navigates around that problem: #!/bin/bash # This is '/home/confus/bin/toggleSpeaker.sh' notify-send "Toggle Speaker" "$(date)" qdbus \ com.canonical.indicator.sound \ /org/ayatana/indicator/service \ org.ayatana.indicator.service.Shutdown exit 0 Help from the community is appreciated as I don't have experience any with DBus whatsoever. Edit 1 - Takkat found a solution but only clickable once? For some reason the solution proposed by Takkat has the drawback that the resulting entry in indicator sound can only be clicked once per session. If someone has a fix for, than please comment or answer, you will be upvoted. Here you can see the result: I strongly suspect the issue is related to the .desktop-file in /home/confus/.local/share/application/toggleSpeaker.desktop, which is this: [Desktop Entry] Type=Application Name=toggleSpeaker GenericName=Toggle Speaker Icon=gstreamer-properties Exec=/home/confus/bin/toggleSpeaker.sh Terminal=false And here is a minimal example of the script in /home/confus/bin/toggleSpeaker.sh for your consideration: #!/bin/bash # This is '/home/confus/bin/toggleSpeaker.sh' notify-send "Toggle Speaker" "$(date)" exit 0

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  • How do I know if my system is capable of playing 24bit/96kHz sound?

    - by Igor Zinov'yev
    Let me state for the record that I'm a total noob when it comes to Hi-Fi sound systems, but I am rather picky about the sound quality. Normally I listen to CD recordings ripped to FLAC in 16/44, but I have several albums that are also ripped from vinyls to FLAC in 24/96. But it seems that I can't tell the difference between 16-bit and 24-bit versions (except for some vinyl noises, of course). That can be due to several reasons: my equipment (onboard audio, monitor headphones) isn't good enough to make any difference, my system is not playing audio in 24-bit 96 kHz, I am physically unable to hear the difference. So here is my question, how do I tell if my system can play 24-bit sound with 96 or 192 kHz resolution? And if it can, how do I tell that it plays it instead of downsampling to 16-bit / 44 kHz? Also, what hardware (audio cards, amplifiers, etc.) would you recommend to play such recordings on Ubuntu?

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  • Distortion problem with Creative audio equalizer

    - by e-t172
    Hi, I have a problem with the Creative Console EQ, I don't know if it's fixable or not (is the EQ software or hardware on these cards?). Basically, I have enormous distortion with certain sounds in the 30 - 125 hz range. When this happens I get some sort of "frrzzzz" (sorry, I'm french and don't really know the correct english word for that) on top of the original sound. I have a Sound Blaster Audigy SE. I'm using the Daniel_K drivers, on Windows 7 Profesionnal x64. All the effects are disabled except EQ. Steps to reproduce Put the card in 24bit/96khz mode. The problem is also present with 16bit/48khz but seems to be less audible. In the Creative Console, use the following EQ: (full size) Play this sound at a reasonably high volume. You should hear distortion on the two "booms". Especially the second one. Disable Creative EQ. Play the sound in an application with an integrated EQ (e.g. foobar2000, ffdshow) using the same EQ parameters. There is no distortion. Conclusion: the Creative EQ is broken. Is anyone having the same problem? I'm also interested in the results with other Creative cards or even other brands soundcards with a similar EQ feature.

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  • How to add a sound that an enemy AI can hear?

    - by Chris
    Given: a 2D top down game Tiles are stored just in a 2D array Every tile has a property - dampen (so bricks might be -50db, air might be -1) From this I want to add it so a sound is generated at point x1, y1 and it "ripples out". The image below kind of outlines it better. Obviously the end goal is that the AI enemy can "hear" the sound - but if a wall is blocking it, the sound doesn't travel as far. Red is the wall, which has a dampen of 50db. I think in the 3rd game tick I am confusing my maths. What would be the best way of implementing this?

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  • How can I force a preferred sound output device to be used?

    - by Dave M G
    In my sound settings interface, there are two devices for sound output: Both refer to the same physical device, which is a network sound device. Both work, but only with the second one, Simultaneous output to Kenwook Audio Device Digital Stereo (IEC958) on mythbuntu@mythbuntu, does the output volume respond to being changed. The first one always plays at the same level, and ignores volume settings. Every time I boot, the first one is selected. How can I make it so the second one is the default and the first one is disabled or at least never selected?

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  • Trust External Surround Soundcard SC-5500p not working

    - by Ederico
    I got the following external sound card to make some awesome noise with my speaker set. It should be plug and play, but it so happens that when I plug it in and hook everything up, there's no sound at all. The speaker set I have works if I plug it in the laptop directly (or rather, if I plug the front speakers cable as I can't plug anything else). http://www.trust.com/products/product.aspx?artnr=14134 Would anyone know how I can workaround this problem and make full use of this external sound card on my Ubuntu 12.04 system?

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  • How to boost playback volume in real time on media recorded with a very low volume.

    - by L Marksman
    I have never heard a satisfactory answer to this often misunderstood question, let me explain. Lets say I have a sound card and earphones/speakers that can play back audio loud enough in most cases. This is great but the problem is that you always find people who do not know how to record audio, from Youtube video's to music. So now you end up with a audio playback that only uses 10% or less of the capacity of your sound hardware, in vista/win 7 you will see this frequently in the mixer with the volume pushed up to max but the green sound level only goes up a millimeter or two. I am looking for (preferably free) software or a method to boost the sound level of any audio from any source in real time to use more of my hardware capacity similar to what VLC media player can do. Oh and please, do not tell me it is impossible. I am not trying to boost the volume past what my hardware is capable of, I am just trying to use my hardware's full capacity. Also please do not tell met to buy new hardware, I know I can use hardware amplification, I don't want to (like many others) spend money on a simple little problem like this. Thanks!

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  • What's the difference between Pygame's Sound and Music classes?

    - by Southpaw Hare
    What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

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  • No sound on Debian unstable Intel Corporation 82801JD/DO (ICH10 Family) HD Audio Controller

    - by Dave Roger
    I have no sound on my Debian unstable. # cat /etc/issue Debian GNU/Linux squeeze/sid # lspci |grep -i audio 00:1b.0 Audio device: Intel Corporation 82801JD/DO (ICH10 Family) HD Audio Controller (rev 02) # lsmod | grep -i snd snd_hda_codec_realtek 235506 1 snd_hda_intel 19907 0 snd_hda_codec 54244 2 snd_hda_codec_realtek,snd_hda_intel snd_hwdep 5380 1 snd_hda_codec snd_pcm_oss 32591 0 snd_mixer_oss 12606 1 snd_pcm_oss snd_pcm 60471 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_seq_midi 4400 0 snd_rawmidi 15515 1 snd_seq_midi snd_seq_midi_event 4628 1 snd_seq_midi snd_seq 42881 2 snd_seq_midi,snd_seq_midi_event snd_timer 15582 2 snd_pcm,snd_seq snd_seq_device 4493 3 snd_seq_midi,snd_rawmidi,snd_seq snd 46446 11 snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 4598 1 snd snd_page_alloc 6249 2 snd_hda_intel,snd_pcm # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.21. # uname -r 2.6.32-5-amd64

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  • Trixbox: external SIP with no sound

    - by Leandro Vidal
    I have a trixbox server and every works find except the external SIPs. Inside net all sound goes fine, but if I use a SIP phone outside the net, I can connect, I can receive calls but I there is no sound. I have this text in the sip_nat.conf: nat=yes externhost=xxxxx.dyndns.org localnet=192.168.1.0/255.255.255.0 localhost=192.168.1.210 externrefresh=10 qualify=yes And I have the ports from 5036 to 5082, 4569 and from 10000 to 20000 redirected to 192.168.1.210 on TCP and UDP. What's wrong? Thank you very much in advance

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  • Mac Mini 1.66 no sound on Windows 7

    - by Steph
    I've tried everything I could to get the sound working. I've tried to install the drivers from the Snow Leopard disk, then I went and got the latest RealTek drivers from the RealTek website. Then I read that old Mac mini's could have an integrated Cirrus chip, and I tried to grab the latest driver from the website, and it appeared to install, but nothing. There's also nothing listed under Sound and ... in the Device Manager. I've tried to manually include a legacy driver, which worked for RealTek but not for the Cirrus driver. Of course it said there was no RealTek hardware install, which makes me think it's a Cirrus chip. Any further suggestions or thoughts would be really appreciated. This is a Mac Mini 1.66. I'm also not sure how to be able to detect what card is integrated into the system, so that would be helpful too (remembering that there's no driver installed). Even just to get the chip type would be great to confirm.

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  • Quiet Sound while recording using the "Stereo Mix" option on Windows XP.

    - by DragoonX
    I'm trying to record some of my work, and the video capture program I'm using works fine. It's HyperCam2, and since it's not really professional, I don't care about that little thing in the corner. Anyways, If I check the record sound option and put it on the highest quality, and I want it to record the sounds playing from my computer, I have it record the "Stereo Mix" setting. However, after a quick test, I saw that even though my computer was at max volume, the recorded sound was very quiet, almost inaudible. Thinking it was just HyperCam, I downloaded Audacity, and only met similar results. While I'm not entirely savvy with hardware, I BELIEVE this to be my soundcard: SoundMAX Integrated Digital HD Audio

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  • Sound recognition software

    - by Cawas
    I'm looking for a software able to recognize an specific sound and then do some action. I want to leave my notebook close by the house intercom so when I hear someone ringing it, a very specific and unique sound, it will send me an email at my office or something. The main issue is that there's a lot of different noises there, but none would be as loud as the intercom for the specific place I've left the microphone. Is there any software out there able to do this? Hopefully with a mac version. I trust there's nothing closely related in this to speech or voice recognition technologies, and specially the softwares in there.

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  • Micro sound breaks/interrupts on Windows 7

    - by cand
    Hello all, I've been experiencing strange behavior recently. When listening to mp3 or watching movie or whatever that uses sound, I get micro breaks in sound. It's like it hangs or cuts a fragment for about 0.5s. When I start OS, it's ok, but as time passes it gets worse, to the extent that music is unlistenable being interrupted every 2 seconds. I haven't found any relevance between this behavior and hardware usage, I don't think it's directly related to HDD (or it might be but with significant delay). I have updated soundcard drivers and it didn't help a lot. My system is Windows 7, computer is simple HP laptop, nx7300-Ru374ES with WD Caviar Scorpio Blue hard drive inside and integrated soundcard on it (I can check the model later if it's important). Did anybody encounter such problem ? Maybe it's a common thing on Windows 7 or someone knows how to solve it? Thanks in advance.

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  • Skype 2.1beta for Linux and sound quality

    - by vava
    I've been using Skype 2.1beta for Linux with my bluetooth headset and quality of the sound is just awful. But not always though, if I call echo service, quality is acceptable, but when I call real people there's echo, sound is crippling, there's pauses, voice is unrecognizable, all sorts of quality problems in one call. If I use newest Skype under WIndows with the same headset to call to the same people, quality is more than normal. So, is there some settings I can tweak, like tell Skype which codec to use or maybe there's noise cancellation plugin for PulseAudio I can use or any other system setting I can try to play with?

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  • Bad sound quality of 3.5mm headphone with mic on laptop

    - by Isaac
    I have a set of headphones that have a built-in mic for hands-free calling. They just work great on my Sony Ericsson Cedar cellphone. The problem is that when I connect headphone to my Dell N5010 laptop to listen to music, the quality is horrible, with very weak or no vocals. They funny part is when I hold down the talk button on the mic (headphone mic), at which point it sounds great, but goes back to bad quality as soon as I release talk button. Also, when I take out the jack a little, at some point, the sound is great but I have to hold the jack there. I looked for any configuration on the sound card driver but find nothing. Besides using a glue to hold down the talk button of mic :), is there any other solution?

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