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  • Equalizer for Toshiba Satellite Pro Laptop (Windows 7) (has Realtek sound)

    - by Need Help
    Hi, I've been trying to find a way to have a real equalizer for my Toshiba Satellite Pro Laptop. I don't know much about computers but the guys in the store seem to know less than I do! From what I've been able to glean through friends, Windows 7 no longer supports a real equalizer function with Realtek, and the result is a fake "equalizer" that does nothing. I have hearing issues so not being able properly adjust the higher frequencies is causing me physical pain and ringing in my ears. I've tried to decipher the threads I find online but am quite confused by them. Basically I need an equalizer that will work with Windows 7 (with everything, internet, skype, music, etc). The current drivers offered by Realtek do nothing except make the sound inaudible. The one that may possibly work and provide an actual equalizer is a few versions back and can't be found anywhere. Thanks! And sorry if this is a duplicate question but I am confused a bit by the threads online.

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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 do I install an equalizer in Rhythmbox?

    - by sayth
    Previous I have never used rhythmbox by default because it seemed to be lacking in features to me. Just my personal opinion. With rhythmbox back in 12.04 will ubuntu give it some attention to give it some usability one thing that was majorly missing on my last use of rhythmbox was an equalizer which is the most basic of requirements for an audio player let alone a preamp. I have searched and found that on the rhythmbox website the plugin is available but in the plugins menu of rhythmbox it is not there. I searched google and there are many guides from 2009 trying to install the equalizer. there is nothing recent and one would assume this would be a default plugin, there is no point after all searching for cover art if your music doesn't sound right. How can I easily install the equalizer in 12.04?

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  • Real-time equalizer for all audio on computer

    - by greye
    Is it possible to capture all the sound from a computer and have it pass through a equalizer before reaching the speakers? How can you program a band pass filter on it? EDIT: I'm trying to get this on Windows (with Python? heh) but if there is a generic, cross-platform approach that would be great.

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  • itunes equalizer per track settings

    - by DA
    I'm a little confused about iTune's ability to set eq presets on a per-song basis. I have a song that I open, set to a given EQ preset, then close. When I play this song, however, the audio isn't being adjusted. If I open the EQ and turn it 'on', though, it then obeys the song's eq setting. Of course, if I leave the EQ on, then songs that do not have their own EQ setting default to whatever the EQ was originally set at. Not a HUGE deal, but it means that I would then normally have to have EQ always turned on, set to flat, so that any song that I HAVE given a EQ setting to will use it. Am I understanding how that works correctly? Is there a way to have individual songs use their EQ setting without having to turn on the EQ for everything?

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  • Update to Where’s My Graphic Equalizer in Windows Media Player – now covers Windows 8

    - by Liam Westley
    Originally posted on: http://geekswithblogs.net/twickers/archive/2013/11/11/update-to-wherersquos-my-graphic-equalizer-in-windows-media-player.aspxHave you wondered where the graphics equaliser in the Windows 8 version of Windows Media Player has moved?  It’s certainly not on the menu option you’d think it is …. well, I’ve updated my Windows 7 post to include Windows 8, it’s over here http://geekswithblogs.net/twickers/archive/2009/10/23/135680.aspx.

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  • DSP - How are frequency amplitudes modified using DFT?

    - by Trap
    I'm trying to implement a DFT-based equalizer (not FFT) for the sole purpose of learning. To check if it works I took an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. Now I tried to silence some frequency bands, just by setting their amplitudes to zero before resynthesis, but definitely it's not the way to go. What I get is a rather distorted signal. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. I first tried to modify the real part amplitudes only, then modifying both the real and imaginary part amplitudes. I also tried to convert the DFT output to polar notation, then modifying the magnitude and convert back to rectangular notation, but none of this is working. Can someone show me what I'm doing wrong? I tried to find info on this subject in the internet but couldn't find any. Thanks in advance.

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  • VLC 2.0.3 on Lubuntu 12.04: No audio?

    - by drezabek
    I am on Lubuntu 12.04, and I have installed VLC media player version 2.0.3. When I try and play an audio file, it appears to load fine, and the media position bar displays the progress, and it says it is playing, but I can't here any thing through my speakers. I can hear game audio, web audio, and audio from SMPlayer just fine, but with VLC, I can't here anything. Below is the "Messages" output with the verbosity option set to "2 (debug)" main debug: processing request item: The Bottom, node: Playlist, skip: 0 main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: starting playback of the new playlist item main debug: resyncing on The Bottom main debug: The Bottom is at 0 main debug: creating new input thread main debug: Creating an input for 'The Bottom' main debug: TIMER input launching for 'Floex - Machinarium Soundtrack - 01 The Bottom.flac' : 23.706 ms - Total 23.706 ms / 1 intvls (Avg 23.706 ms) main debug: using timeshift granularity of 50 MiB, in path '/tmp' main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' gives access `file' demux `' path `/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access_demux module: 3 candidates main debug: no access_demux module matching "file" could be loaded main debug: TIMER module_need() : 2.332 ms - Total 2.332 ms / 1 intvls (Avg 2.332 ms) main debug: creating access 'file' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac', path='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for access module: 2 candidates filesystem debug: opening file `/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: using access module "filesystem" main debug: TIMER module_need() : 0.762 ms - Total 0.762 ms / 1 intvls (Avg 0.762 ms) main debug: Using stream method for AStream* main debug: starting pre-buffering main debug: received first data after 0 ms main debug: pre-buffering done 1024 bytes in 0s - 43478 KiB/s main debug: looking for stream_filter module: 7 candidates main debug: no stream_filter module matching "any" could be loaded main debug: TIMER module_need() : 0.236 ms - Total 0.236 ms / 1 intvls (Avg 0.236 ms) main debug: looking for stream_filter module: 1 candidate main debug: using stream_filter module "stream_filter_record" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: creating demux: access='file' demux='' location='/home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' file='/home/doug/Music/unsorted/Floex - Machinarium Soundtrack/Floex - Machinarium Soundtrack - 01 The Bottom.flac' main debug: looking for demux module: 54 candidates flacsys debug: Picture type=3 mime=image/png description='' file length=679371 qt4 debug: IM: Setting an input main debug: looking for packetizer module: 21 candidates main debug: using packetizer module "packetizer_flac" main debug: TIMER module_need() : 0.211 ms - Total 0.211 ms / 1 intvls (Avg 0.211 ms) main debug: using demux module "flacsys" main debug: TIMER module_need() : 4.023 ms - Total 4.023 ms / 1 intvls (Avg 4.023 ms) main debug: looking for a subtitle file in /home/doug/Music/unsorted/Floex - Machinarium Soundtrack/ main debug: looking for meta reader module: 2 candidates main debug: using meta reader module "taglib" main debug: TIMER module_need() : 5.245 ms - Total 5.245 ms / 1 intvls (Avg 5.245 ms) main debug: removing module "taglib" main debug: `file:///home/doug/Music/unsorted/Floex%20-%20Machinarium%20Soundtrack/Floex%20-%20Machinarium%20Soundtrack%20-%2001%20The%20Bottom.flac' successfully opened main debug: selecting program id=0 main debug: looking for decoder module: 30 candidates main debug: using decoder module "flac" main debug: TIMER module_need() : 0.442 ms - Total 0.442 ms / 1 intvls (Avg 0.442 ms) main debug: Buffering 0% flac debug: decode STREAMINFO flac debug: channels:2 samplerate:44100 bitspersamples:16 flac debug: STREAMINFO decoded main debug: Buffering 30% main debug: recycling audio output main debug: looking for audio output module: 3 candidates main debug: Buffering 61% pulse debug: using stereo channel map pulse debug: using library version 1.1.0 pulse debug: (compiled with version 1.1.0, protocol 26) main debug: Buffering 92% main debug: Stream buffering done (371 ms in 2 ms) pulse debug: connected locally to unix:/home/doug/.pulse/dce22254e867f905188a2ce200000003-runtime/native as client #14 pulse debug: using protocol 26, server protocol 26 pulse debug: using buffer metrics: maxlength=4194304, tlength=9880, prebuf=0, minreq=3528 pulse debug: connected to sink 0: alsa_output.pci-0000_00_14.2.analog-stereo main debug: using audio output module "pulse" main debug: TIMER module_need() : 4.571 ms - Total 4.571 ms / 1 intvls (Avg 4.571 ms) main debug: output 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: mixer 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: filter(s) 'f32l'->'s16l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: f32l->s16l, bits per sample: 32->16 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.187 ms - Total 0.187 ms / 1 intvls (Avg 0.187 ms) main debug: conversion pipeline completed main debug: looking for audio mixer module: 2 candidates main debug: using audio mixer module "float32_mixer" main debug: TIMER module_need() : 0.125 ms - Total 0.125 ms / 1 intvls (Avg 0.125 ms) main debug: input 's16l' 44100 Hz Stereo frame=1 samples/4 bytes main debug: looking for audio filter module: 1 candidate scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: TIMER module_need() : 0.233 ms - Total 0.233 ms / 1 intvls (Avg 0.233 ms) main debug: filter(s) 's16l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo pulse debug: listing sink alsa_output.pci-0000_00_14.2.analog-stereo (0): Built-in Audio Analog Stereo main debug: looking for audio filter module: 14 candidates audio_format debug: s16l->f32l, bits per sample: 16->32 main debug: using audio filter module "audio_format" main debug: TIMER module_need() : 0.147 ms - Total 0.147 ms / 1 intvls (Avg 0.147 ms) main debug: conversion pipeline completed pulse debug: base volume: 65536 main debug: looking for audio filter module: 1 candidate equalizer debug: equalizer loaded for 44100 Hz with 10 bands 2 pass equalizer debug: 60 Hz -> factor:0.000000 alpha:0.003013 beta:0.993973 gamma:1.993901 equalizer debug: 170 Hz -> factor:0.000000 alpha:0.008490 beta:0.983019 gamma:1.982437 equalizer debug: 310 Hz -> factor:0.000000 alpha:0.015374 beta:0.969252 gamma:1.967331 equalizer debug: 600 Hz -> factor:0.000000 alpha:0.029328 beta:0.941343 gamma:1.934254 equalizer debug: 1000 Hz -> factor:0.000000 alpha:0.047918 beta:0.904163 gamma:1.884869 equalizer debug: 3000 Hz -> factor:0.000000 alpha:0.130408 beta:0.739184 gamma:1.582718 equalizer debug: 6000 Hz -> factor:0.000000 alpha:0.226555 beta:0.546889 gamma:1.015267 equalizer debug: 12000 Hz -> factor:0.000000 alpha:0.344937 beta:0.310127 gamma:-0.181410 equalizer debug: 14000 Hz -> factor:0.000000 alpha:0.366438 beta:0.267123 gamma:-0.521151 equalizer debug: 16000 Hz -> factor:0.000000 alpha:0.379009 beta:0.241981 gamma:-0.808451 main debug: using audio filter module "equalizer" main debug: TIMER module_need() : 0.353 ms - Total 0.353 ms / 1 intvls (Avg 0.353 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: looking for visualization2 module: 1 candidate main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 3.278 ms - Total 3.278 ms / 1 intvls (Avg 3.278 ms) main debug: looking for video filter2 module: 18 candidates swscale debug: 32x32 chroma: YUVA -> 16x16 chroma: RGBA with scaling using Bicubic (good quality) main debug: using video filter2 module "swscale" main debug: TIMER module_need() : 1.037 ms - Total 1.037 ms / 1 intvls (Avg 1.037 ms) main debug: looking for video filter2 module: 18 candidates yuvp debug: YUVP to YUVA converter main debug: using video filter2 module "yuvp" main debug: TIMER module_need() : 0.156 ms - Total 0.156 ms / 1 intvls (Avg 0.156 ms) main debug: Deinterlacing available main debug: deinterlace 0, mode blend, is_needed 0 main debug: Opening vout display wrapper main debug: looking for vout display module: 6 candidates main debug: looking for vout window xid module: 4 candidates qt4 debug: requesting video... qt4 debug: Video was requested 0, 0 main debug: using vout window xid module "qt4" main debug: TIMER module_need() : 61.671 ms - Total 61.671 ms / 1 intvls (Avg 61.671 ms) main debug: looking for inhibit module: 2 candidates main debug: using inhibit module "xdg_screensaver" main debug: TIMER module_need() : 0.336 ms - Total 0.336 ms / 1 intvls (Avg 0.336 ms) xdg_screensaver debug: started xdg-screensaver (PID = 6682) xcb_xv debug: connected to X11.0 server xcb_xv debug: vendor : The X.Org Foundation xcb_xv debug: version: 11103000 xcb_xv debug: using screen 0x15a xcb_xv debug: using XVideo extension v2.2 xcb_xv debug: using adaptor NV17 Video Texture xcb_xv debug: using port 310 xcb_xv debug: using image format 0x30323449 xcb_xv debug: using X11 visual ID 0x21 (depth: 24) xcb_xv debug: using X11 window 0x03400000 xcb_xv debug: using X11 graphic context 0x03400002 main debug: VoutDisplayEvent 'fullscreen' 0 main debug: VoutDisplayEvent 'resize' 800x500 window main debug: using vout display module "xcb_xv" main debug: TIMER module_need() : 69.890 ms - Total 69.890 ms / 1 intvls (Avg 69.890 ms) main debug: original format sz 800x500, of (0,0), vsz 800x500, 4cc I420, sar 1:1, msk r0x0 g0x0 b0x0 main debug: removing module "freetype" main debug: looking for text renderer module: 2 candidates freetype debug: Building font databases. freetype debug: Took 0 microseconds freetype debug: Using Serif Bold as font from file /usr/share/fonts/truetype/ttf-dejavu/DejaVuSans.ttf freetype debug: using fontsize: 2 main debug: using text renderer module "freetype" main debug: TIMER module_need() : 4.552 ms - Total 4.552 ms / 1 intvls (Avg 4.552 ms) main debug: using visualization2 module "visual" main debug: TIMER module_need() : 84.104 ms - Total 84.104 ms / 1 intvls (Avg 84.104 ms) main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: conversion pipeline completed main debug: filter(s) 'f32l'->'f32l' 48510 Hz->44100 Hz Stereo->Stereo main debug: looking for audio filter module: 14 candidates main debug: using audio filter module "samplerate" main debug: TIMER module_need() : 0.375 ms - Total 0.375 ms / 1 intvls (Avg 0.375 ms) main debug: conversion pipeline completed main debug: End of audio preroll main debug: Decoder buffering done in 91 ms main warning: PTS is out of range (-9269), dropping buffer pulse debug: deferring start (190703 us) main debug: looking for video blending module: 1 candidate main debug: using video blending module "blend" main debug: TIMER module_need() : 0.275 ms - Total 0.275 ms / 1 intvls (Avg 0.275 ms) main debug: Detected interlaced video main debug: deinterlace 0, mode blend, is_needed 1 xcb_xv debug: display is visible pulse debug: starting deferred pulse warning: too late by 93760 us pulse debug: changed sample rate to 44186 Hz pulse debug: started pulse warning: too late by 94474 us pulse debug: changed sample rate to 44229 Hz pulse warning: too late by 93532 us pulse debug: changed sample rate to 44272 Hz pulse warning: too late by 92829 us pulse debug: changed sample rate to 44315 Hz pulse warning: too late by 92132 us pulse debug: changed sample rate to 44358 Hz xcb_xv debug: display is visible pulse warning: too late by 91534 us pulse debug: changed sample rate to 44401 Hz xcb_xv debug: display is visible pulse warning: too late by 89482 us pulse debug: changed sample rate to 44440 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible pulse warning: too late by 87529 us pulse debug: changed sample rate to 44479 Hz pulse warning: too late by 84577 us pulse debug: changed sample rate to 44504 Hz main debug: auto hiding mouse cursor pulse warning: too late by 78562 us pulse debug: changed sample rate to 44492 Hz pulse warning: too late by 68015 us pulse debug: changed sample rate to 44422 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor pulse debug: changed sample rate to 44336 Hz xcb_xv debug: display is visible xcb_xv debug: display is visible xcb_xv debug: display is visible main debug: auto hiding mouse cursor I have had issues with VLC in the past- the audio quality was extremely crackly, as if the headphone jack was plugged in only half way, and the sounds were extremely sharp and caused my speakers to make a ringing/vibrating noise... It would eventually start working after I messed around with the audio settings, but it happened every restart. I eventually switched to SMPlayer, but now I need some of the features that VLC offers, but I still can't use VLC. At this point, the audio can not be heard at all, and the method I used before, messing around with the audio settings, isn't getting me anywhere. (note, I reposted this on VideoLan's forums, link is here: http://forum.videolan.org/viewtopic.php?f=13&t=104726) Please let me know if you need more information, or are confused by something I posted! Thanks!

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  • What would cause the Graphic Equalizer in Windows Media Player 10 to be disabled/not available?

    - by creamcheese
    On XP, Windows Media Player 10 contains a Graphic Equalizer but I can't find any way to activate it. Could this be a codec issue or a hardware issue? I'm also not getting any sound at all from this computer but there are no apparent hardware device issues marked in Device Manager, nothing is muted, all volume levels are turned fully up (Windows Media Player and the Windows Volume Controller). Stumped!

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  • How to have similar sound quality in Banshee and Rhythmbox (or Totem) ?

    - by Alkalyzer
    Hello, I am running Ubuntu 10.10 on an Acer Aspire 6390G. With its 5.1 embedded sound system, this laptop is good at playing music using in-built speakers. The soundcard is an HDA-Intel ALC888. When I use Rhythmbox (version 0.13.1) (or Totem) (and the in-built speakers), the sound is deep, warm, great. But with Banshee (version 1.8.0.), it is always crispy and basically of poor quality when the Banshee equalizer is disabled. I know that it is possible to fine tune Banshee sound using the equalizer, but after some time spent trying to properly adjust it without success, I decided to ask this question on askubuntu.com. Is there a way to get the same sound quality in Banshee as in Rhythmbox (or Totem) or keeping on adjusting Banshee equalizer is the only alternative to try to solve this problem ? (If the answer is the latter, is there anyone who would be kind enough to provide me his equalizer settings ?) Thanks all for your answers.

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  • Asus Sonic Master on Asus N53SV

    - by David Winchester
    I have read that there's a problem to get the subwoofer working in these laptops. I tried this solution No sound from external subwoofer but I don't how to prove that the subwoofer is properly functioning. I use Pulseaudio equalizer and the bass sound seems to work fine, but when I go to the Sound Settings, I can't move the bar where it says 'Subwoofer' in my sound card option, so I don't know if everything is alright. If someone has a solution I would like to know, because there isn't much information regarding this. By the way, I'm using Ubuntu 12.04 64 bits. Thanks beforehand, Dave EDIT ----------- Possible Solution Well, I will post a solution that worked for me and I think it will help a lot of users. I finally got the subwoofer working. Besides adding in /etc/modprobe.d/alsa-base.conf the line options snd-hda-intel model=asus-mode4 I deleted the lines with load-module module-combine and module-combine-sink in /etc/pulse/default.pa (in the home folder there's also a ~/.pulse/default.pa file, I don't know if it has the lines too) To assure all the channels are working, I think this command tells me that speaker-test -c6 -l1 -twav I use pulseaudio-equalizer and the bass sounds very well when properly adjusted. Also, all the channels seems to work fine and the sound is even better than in Windows (where I don't have an equalizer). I pointed out before a module-combine and module-combine-sink problem, because one day I turned on my laptop and pulseaudio didn't work. So I deleted the lines with that names (don't know if they came by default, maybe I added them sometime when I was trying to fix my speakers). After all this, I can now move the Subwoofer bar in the Sound configuration. Anyways, the Equalizer does a great job and it improves the sound a lot.

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  • Improve Playback Using Enhancements in Windows Media Player 12

    - by DigitalGeekery
    Are you looking for ways to improve the playback of your media in Windows Media Player 12? We’ll show you how to do that by using the enhancements in WMP 12. If you are in Library mode, you’ll need to click the icon at the lower right to switch to Now Playing mode. Right-click anywhere in Media Player while in Now Playing mode, select Enhancements, and select any of the available options.   You can switch between the individual enhancements by clicking the right and left buttons at the top left.   Crossfading and Auto Volume Leveling The Auto Volume Leveling setting is just a simple toggle on and off. If your MP3 or WMA files have volume leveling information values.   You can automatically add volume leveling information values to all files you add to your library by switching to Library view, going to Tools > Options, and selecting Add volume leveling information values for new files on the Library tab. Click OK when finished.   Crossfading will gradually decrease the volume of the song that is ending (fade out) and increase volume of the song that is beginning. Click Turn on Crossfading and then click and drag the slider left or right change the amount of overlap between tracks. Graphic Equalizer The graphic equalizer is toggled on and off by clicking Turn on / Turn off at the top left. You can select pre-defined equalizer settings by music genre by clicking the Default list. The radio buttons on the left allow you to move the sliders individually, in a loose group or a tight group. You can always return to the default settings by clicking Reset. Play Speed Settings Choose a pre-defined settings by clicking Slow, Normal, or Fast. Uncheck the Snap slider to common speeds the move the slider right and left to your desired speed. If nothing else, these settings provide a little fun and amusement. Quiet Mode Quiet mode will level out any sharp volume highs and lows within a single track. Simply toggle the setting on or off and select whether you prefer Medium difference or Little difference by selecting one of the radio buttons. SRS WOW effects SRS WOW effects enhance low-frequency and stereo sound performance. Click Turn on to enable the TruBass and WOW Effect sliders. You can also optimize for your speaker type. Click to switch between Regular, Large, and Headphones. Video Settings Video Settings allow you to adjust the Hue, Brightness, Saturation, and Contrast.   You can also adjust the zoom settings by clicking Select video zoom settings.   Dolby Digital Settings Choose between Normal, Night, and Theater settings to adjust the audio for Dolby Digital content. This setting will only effect media with Dolby Digital sound. Looking for more ways to improve your media experience in WMP 12? Check out how to update metadata and cover art and how to share media with other Windows 7 computers on your home network. Similar Articles Productive Geek Tips Fixing When Windows Media Player Library Won’t Let You Add FilesInstall and Use the VLC Media Player on Ubuntu LinuxHow To Rip a Music CD in Windows 7 Media CenterStream Media from Windows 7 to XP with VLC Media PlayerInstalling Windows Media Player Plugin for Firefox TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Acronis Online Backup DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows Check these Awesome Chrome Add-ons iFixit Offers Gadget Repair Manuals Online Vista style sidebar for Windows 7 Create Nice Charts With These Web Based Tools Track Daily Goals With 42Goals Video Toolbox is a Superb Online Video Editor

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  • Will proprietary software-based sound enhancements work with Ubuntu? (BeatsAudio, Dolby)

    - by LiveWireBT
    This question is targeted at mainstream or gamer-grade software-based audio/sound enhancements, found in highly integrated computing and entertainment systems like laptops, tablets and smartphones. These are mostly marketed with fancy badges of known audio-releated brands on the product or packaging, while being mostly uncertain about the actual implementation or components used and poorly differentiated from the general audio capabilities of the system or device. This question is not about actual hardware like speakers. If your headphones are not properly detected, your speakers are assigned wrong, work partially or not at all then your soundcard or chip is not properly detected and you should take a look at troubleshooting audio issues. This question is also not about enthusiast or recording-grade hardware like recording interfaces, amplifiers and DACs in a variety of formfactors. And this question is also not about audio encoding and playback of different audio formats like Dolby Digital, Dolby TrueHD and DTS. Most of these may be subject to patents and licensing, see restricted formats. If you are just searching for an equalizer, please take a look at this question: Is there any Sound enhancers/equalizer? Simply speaking: Every feature where you would flip a switch or check a box in a fancy looking interface in Windows that makes the sound change from neutral to fancy.

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  • DSP - Filtering frequencies using DFT

    - by Trap
    I'm trying to implement a DFT-based 8-band equalizer for the sole purpose of learning. To prove that my DFT implementation works I fed an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. This method calculates the real and imaginary parts both N/2 + 1 samples in length. To attenuate a frequency I'm just doing: float atnFactor = 0.6; Re[k] *= atnFactor; Im[k] *= atnFactor; where 'k' is an index in the range 0 to N/2, but what I get after resynthesis is a slighty distorted signal, especially at low frequencies. The input signal sample rate is 44.1 khz and since I just want a 8-band equalizer I'm feeding the DFT 16 samples at a time so I have 8 frequency bins to play with. Can someone show me what I'm doing wrong? I tried to find info on this subject on the internet but couldn't find any. Thanks in advance.

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  • Is there a way to suppress keyboard sounds picked up by a desktop microphone in Windows 7?

    - by Dave Andersen
    The problem I'd like to solve is that my desktop microphone picks up all my keystrokes very loudly compared with my voice. Even just lightly tapping a key without depressing it causes a loud click to be picked up. I'd like a way to filter out this type of sound, while picking up voice normally. Is there any software input equalizer/filter that could do this? Or alternatively, some sort of hardware hack?

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  • kubuntu - what's the best smplayer configuration for best quality in hd movies (mkv)

    - by Frank
    I have ubuntu x64 13.04 with kde 4.11 and smplayer v0.8.6 and the last mplayer version from ppa. I have ATI video card HD6870 MSI with fglrx driver v13.4. My kde settings are: Composition mode: Opengl 3.1 graphic system qt: Raster scaling mode: precise Vsync: auto So what's the best configuration for quality over performace in smplayer according to my system specs? For example what do I have to set for the following options? enable postprocessing by default and postprocessing quality output driver Deinterlacing method software equalizer direct rendering double buffering draw video using slices threads for decoding (MPEG-1/2 and H.264 only loop filter use CoreAVC Thanks

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  • How do I install third-party rhythmbox plugins?

    - by fossfreedom
    Now that the dust has settled and Rhythmbox has become (again) the default music-media player in 12.04, I'm interested in extending its functionality. For example, the default lyric plugin does not work for me and there doesn't appear to be an sound-equalizer by default. Having done a search, I came across the Gnome-website that lists a number of third-party plugins, some-of which I wish to install which can resolve the above. However, there doesn't appear to be .deb packages or a repository containing these plugins. Instead there are links to source-code websites such as GitHub and others. So, I'm confused - I don't know which plugins works in 12.04 Rhythmbox and I'm not sure how to install these. Help please?

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  • How can I have better sound quality?

    - by joelalmeidaptg
    I have been using Ubuntu for years now, and the latest Ubuntu 13.10 is working perfectly on my N56VZ. The image quality is awesome, better than on Windows, but there is one thing that is really bugging me and that kills my "cinema" experience... The sound. Ubuntu sound quality isn't nearly as good as it does on Windows using Realtek (with Powerful on the equalizer). On Ubuntu the sound is like "faded", it isn't as clear as on Windows. This happens on the system overall: VLC, Youtube, Rhythmbox... I think it is pulse itself that has a horrible sound quality. So, does anyone knows a solution for this? How can I have better sound quality on Ubuntu?

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  • haw to install libenca.so.0 in SUSE Linux (release 11Kernel linux 2.6.32.59-0.7-pae)gnome 2.28.2 [closed]

    - by Zeljko
    I haw problem with Mplayer. When I try to watch mouvie i got error code: /usr/bin/mplayer -noquiet -nofs -nomouseinput -vc coreserve, -sub-fuzziness 1 -identify -slave -vo xv, -ao alsa, -nokeepaspect -framedrop -dr -double -input conf=/usr/share/smplayer/input.conf -stop-xscreensaver -wid 31457623 -monitorpixelaspect 1 -ass -embeddedfonts -ass-line-spacing 0 -ass-font-scale 1 -ass-styles /home/adria/.config/smplayer/styles.ass -fontconfig -font Arial -subfont-autoscale 0 -subfont-osd-scale 20 -subfont-text-scale 20 -subcp ISO-8859-1 -subpos 100 -nocache -osdlevel 0 -vf-add eq2,hue -vf-add screenshot -slices -channels 2 -af equalizer=0:0:0:0:0:0:0:0:0:0 -softvol -softvol-max 110 /home/adria/Resident Evil - Retribution 2012 R5 DVDRiP XviD AC3 - BHRG/Resident Evil - Retribution 2012 R5 DVDRiP XviD AC3 - BHRG.avi /usr/bin/mplayer: error while loading shared libraries: libenca.so.0: cannot open shared object file: No such file or directory help please

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  • Selective bandwidth shaping

    - by the_candyman
    I was searching something for bandwidth shaping. I found trickled and wondershaper which works on the total bandwidth. I would like to know if there exists a selective bandwidth shaping. I mean, I run 2 application which uses internet. I would like to limit the bandwidth of one of these 2 in real time (just like a sound equalizer). For example, I'm downloading something. Meanwhile, someone calls me on skype. So I want to slow the downloading to have more band in my video calling. Is this possible?

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  • How to get audio spectrum analysis?

    - by Mrwolfy
    I need to find or create a tool that analyzes the audio spectrum of a sound file (like a .wav or .mp3). I need to output the "volume" or power of x number of frequency bands and output the data as text. This will be used to produce a visualization, a graphic equalizer like you'd see on a stereo. I am currently looking at python to do it. My question is are there some tools out there that would do this (signal processing), like math works or others? I don't have any experience with them so any advice would be appreciated.

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  • Android: Music track visualization

    - by Swathi EP
    Hello all, I want to create an music track visualization for an music player application which should look like as below: you can see in the above image that, there is an equalizer kind of view and it should vary as the music track is played. I need to know the right way to achieve the above visualization, which api to use?, etc., Any suggestion will be greatly helpful to me. Thanks, Swathi

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