FAQ creation thread 2: Son of "Shall We Make a FAQ?". This time, it's personal...

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GodsMadClown

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[size=+2]How to Ask Questions.[/size]

[size=+2]How can I make myself clear?[/size]

Using decent spelling, grammar and punctuation are all conducive to getting quality responses. When discussing a particular piece of gear, you may provide a link, but please also provide at least the name of the item. Not everybody wants to go clicking on a link to see what you're talking about.​

[size=+2]What sort of [equipment] should I get?[/size]
To make useful recommendations, you need to give proper information.​
  • Give your budget.

    How much are you willing to spend to listen to good sound? You might think this computer hobby is expensive but, the amount you could spend for audio can make what you spent on your new computer look like pennies. However unlike high priced computer gear, quality audio equipment does not really become obsolete. Good gear can give years, even decades of excellent service.​
  • Tell us how and where you might use the equipment.

    Do you plan to use your computer audio rig to listen primarily to music, play lots of games, and/or watch movies on your computer? List a few kinds of music, games, and movies you like. Give us the size of the room, and how the speakers and listeners might be placed.​
  • Tell us what other audio equipment you have on hand.

    It makes little sense recommending uber-speakers if you plan to run them off some crappy integrated sound implementation. Also, where available, old speakers or receivers might be rolled in to the solution to produce good sound for rather little money.​

[size=+2]How should I ask for technical support?[/size]
Please provide post as much information as possible. Tell us about your audio hardware, software involved, operating system, and other computer hardware. Describe how everything is connected and how to recreate the problem. Too much information is better then too little. Please be as clear as possible and try to use proper punctuation and grammar.​

[size=+2]Dealing With General Equipment[/size]

[size=+2]Which is better, X or Y? [/size]

Short version: Try it and find out for yourself.

Long version: Determining what is "best" is an inexact science, to be sure. A determination can depend on many variables, including the equipment and space used for the implementation, the criteria used for evaluation, and, of course subjective opinion. So, deciding what is best is really a highly personal decision. While asking for more informed opinions might introduce you to new alternatives, ultimately the best way to evaluate any given piece of equipment is with experience.

Preferably, you would audition it in the final implementation. That means that you want to hook it up to the source that you will use, in the space that you will use, and listening from the position that you will use. You can go buy gear from a decent store, audition it at home, and return it if you don't like it. Failing that, you should audition it in the store, keeping as many things like your final implementation as practical.

It is highly recommended to have as little dead air between samples as possible. Auditory memory is notoriously transient, with decent recall being measured at around 5-10 seconds. Also, use familiar music. You might find that you will hear new aspects of a familiar recording when played on better equipment.
​

[size=+2]How do I deal with noise in my audio?[/size]

Noise in your output can be caused by many sources. An effective way to troubleshoot is based on cost of repair, attemping no cost solutions before trying more expensive fixes.

If you are using the "cd-audio" cable from your CD-rom drive, unplug it and throw it in the trash. The IDE connection has been able to cary CD audio data since Windows 95, and there is no good reason not to use it. Also disable any uneeded inputs from your soundcard's mixer. It is a good practice to always disable the mic and auxulilary inputs when not using. These controls can be accessed by clicking the the "advanced" button in the volume control section of "Sound and Audio Devices" in the Windows control panel. Finally, if using a PCI soundcard, it should be plugged into the PCI slot farthest from the Video card, to reduce the possibility of interference .

Next, look at your output cables. Try to move them away from other cabling, especially power cords. If proximity is unavoidable, try to run the lines perpendicular to one another. Shielded cables will help allieviate any inductive noise issues from neighboring cables. Ask in the forum for reccomendations on good shielded cables if you think it may solve your problems. Digital signals can produce a good deal of of electro-magnetic intereference (EMI), which can play havoc with analog signals of the same frequency range as the EMI.

Now if the noise still persists, after insuring the uneeded inuts are disabled, no analog cables inside the case, and resolving potential external cabling problems , then you can be certain it's a hardware issue of some sort.

To find out the hardware issues, you need to carefully examine your PC, and start ruling out peices of hardware. If it's watercooled, it could be the pump, or the AC line that is going inside...keep in mind AC frequency is 60hz...which is in the audible range as bass notes. If it happens when using your mouse, switch it over to either USB or PS/2 and see if the noise persists. You will need to keep searching until you find it. Because there can be a wide variety of causes, there is no single solution.

In some cases with onboard sound, you may find they did not incorporate adequate shielding to their design, allowing noise to get into the line even before it gets to the miniplug jacks on the back of the case. In those instances of a poor sound implementation, the best fix is a quality soundcard. There are some very decent cards that cost around $30.

Finally, if the noise still persists, you could consider moving to a digital solution. A digital signal over S/PDIF is very well protected against noise. Dropped packets and jitter are something else entirely, and is not "noise".

Here's a quick list of stuff to check, roughly in order of increasing expense.

  • Unplug the skinny audio cable that connects your soundcard and optical drive.
  • Mute all unused input/output channels in the advanced volume control panel.
  • Plug audio card into slot farthest from video card.
  • Try both types of mouse connection, PS/2 and USB.
  • Reposition audio cables away from, or perpendicular to other cables.
  • Buy shielded cables.
  • Buy another soundcard.
  • Buy a reciever and use a digital connection.

[size=+2]Dealing With Digital Output[/size]

[size=+2]What's the deal with digital audio? [/size]

Digital audio is typically carried in a signal that conforms to the S/PDIF (Sony Philips Digital Interface, with either an optical or coaxial connection. Digital signals over S/PDIF are not all created equal. Here's a quick rundown of the three basic encoding techniques that most receivers can understand:
  • LPCM (Linear Pulse Code Modulation): This is the basic 2-channel stereo format used by most receivers. It is an uncompressed lossless stereo (2 channel) audio format.
  • DD (Dolby Digital): This is a widely used 5.1 channel audio format for DVDs. It uses a very decent, but lossy compression.
  • DTS (Digital Theater Systems): This is another 5.1 channel audio format that many consider superior to DD because it uses a higher-bitrate compression algorithm that is less "lossy" than Dolby Digital.
Every soundcard that has a digital output can encode an LPCM stream. They can also usually be configured to "pass-through" streams from content with Dolby Digital or DTS audio, such as DVDs.
​

[size=+2]What is Soundstorm? or... What is Dolby Digital Live or DICE? [/size]

Some sound implementations, such as SoundStorm, have an encoder chip that runs the Digital Interactive Content Encoder (DICE) or Dolby Digital Live encoder that takes any audio information from your PC and encodes it into a Dolby Digital stream.

These implementations can encode audio output into a Dolby Digital stream in real time. Consumer hardware solutions include nVidia SoundStorm, Intel HD Audio, and the HiTec HDA Digital Mystique.​

[size=+2]Why use digital output?[/size]

Usually the DAC and output stages on an external receiver are better than those on a soundcard or motherboard. On an nForce2 board, for example, the output of the typical onboard solution is pathetic compared that on most standalone soundcards, and just about every entry-level digital receiver. With digital output, on may use the superior output of an external decoder.

However, many implementations, like those using VIA Envy or Creative Audigy audio chips, have excellent analog output. In common practice and with average budget constraints, there is little to be gained by using a digital solution, as the extra cost of the decoder will often come at the expense of the other components like speakers.​

[size=+2]Dealing With Creative[/size]

[size=+2]How do I make the new Creative drivers work with older Creative soundcards? or... How do I install drivers if I've lost my driver CD? or... How do I make newer EAX versions work on my older Creative soundcard?[/size]

For some bizarre reason, Creative doesn't want to let their customers download drivers for their hardware, only updates. If you want to download the entire driver suite, you can download the latest version here. It is the Audigy 2 ZS driver install that has been hacked to support all Creative cards since the SoundBlaster Live!.
​

[size=+2]How do I make my front panel audio work with my Creative card?[/size]

You need to rig up some way of connecting your various input/output connections to the front panel connections of the card. Here is a pinout guide to the front panel connector of the Audigy series (Audigy - Audigy2 ZS).

Audigy_front_pin_diagram.jpg


  1. analog ground
  2. analog headphone left channel
  3. audio backpanel mute (short to ground to mute the backpanel output, for automute when headphones are plugged in)
  4. analog headphone right channel
  5. same as #3
  6. microphone input
  7. key pin (shouldn't be there)
  8. VRef out (voltage reference for microphone)
  9. microphone input mute (ground when microphone isn't plugged in, +12VDC
  10. audio cable detect (not normally used, will be ground when headphones are plugged in)

[size=+2]Dealing With Setup[/size]

[size=+2]How do I set up my speakers?[/size]
You use trial and error.
There are many variables to proper speaker placement that we cannot hope to cover in this FAQ. The best way to answer that question is trial and error combined with critical listening. Here are some ideas.
  • Read the manual.
  • Position the speakers as best you can in a circular pattern around your normal listening position.
  • Occupy said listening position and check your results with an audio track you know well.
  • adjust position of speakers until they sound the best to your ears.

With that said, here are some pictures from Dolby. They are a good place to start.

Generalized 5.1 setup.

5_1_speaker_setup.gif


Generalized 7.1 setup.

7_1_speaker_setup.gif

​
[size=+2]Do I need that funny cable that connects my optical drive to my soundcard?[/size]

No. It's an obsolete means of transmitting audio from an optical drive. Modern versions of Microsoft Windows support digital audio extraction, which makes the analog cable redundant. In fact, it can even act as an antenna, piping electrical noise directly into your soundcard. So unplug it.​

[size=+2]How do I connect my PC to my home theater system? [/size]

The quickest and easiest way to connect a PC to a home theater system is to use a 1/8" jack-RCA adaptor. Plug the 1/8" minijack end into the appropriate soundcard output, and plug the RCA ends to an available input on your audio equipment.

Alternately, you can use a digital connection of either the optical or coaxial variety. This typically carries only a stereo signal, unless you are playing back a previously encoded digital stream​

3_5jack_RCA.jpg


[size=+2]How do I connect my PC to my home theater system to get surround sound? [/size]

If your motherboard supports DD encoding, then you can encode the multichannel audio, and output it digitally to the receiver.

Otherwise, you need a receiver with multichannel analog inputs. Get 3 minijack-RCA cables like the one pictured above, and route the soundcard's L/R, SL/SR, Cent/Sub outputs into the respective inputs on your receiver.​

[size=+2]Dealing With MP3s
(and other compressed audio)[/size]

[size=+2]How do I make good sounding MP3s? [/size]

Use Exact Audio Copy. Configure LAME as an external decoder. This is a good introductory tutorial to setting up Exact Audio Copy​

[size=+2]How do I convert a WMA file (or any lossy format) into an MP3 file (or another lossy format)?[/size]

If at all possible, don't. When you transcode from one lossy format to another, signifigant audio artifacts can be introduced. It's something like making a photocopy of a photocopy. With each iteration, more "noise" is increased.

With that said, you can use dBpowerAMP to convert them.​

[size=+2]What is the best MP3 (et al.) player software?[/size]

Use whatever you like, really. Pretty much all are free. Download them and play around.
Foobar2k and Winamp both have lots of plugins.​

[size=+2]What is the best music compression format?[/size]

Really, the best way to answer that question is try them out and compare them one against the other. The ABX comparison plugin for Foobar2000 is a handy tool for this task.

If you don't want to rerip all your music if you upgrade to better your equipment, you would be well advised to consider encoding to some lossless compression format like Monkey's Audio or FLAC. This file can then be used as a source for the lossy format of your choosing for your portable player. While losslessly encoded files can occupy much more storgae space, you can be assured that your music will always sound its best.​

[size=+2]How do I make my music play through more than two speakers?[/size]

The vast majority of music is stereo, so it is perfectly normal for it to play through only two speakers. There are several ways to matrix the stereo sound out to other speakers, taking the stereo signal and remixing it for more than two channels.

First read your soundcard and speaker documentation. Depending on the card there could be an option in the drivers to enable a multichannel upmix. Creative also has their own matrixing algorithm, called CMSS usually. Other cards use other algorithms.

Alternately, you could use your player software. A simple way to enable matrixing with Creative hardware is to enable hardware acceleration in Winamp. Foobar2000's also has DSPs that will do matrixing. The quality of all these matrixing algorithms is highly variable and subjective, so use critical ears, familiar music and your own judgment.
​
 
Originally posted by SilverMK3:

Let me expand on what's already been said:

DTS = Digital Theater System (Sony)
It is used a lot in theaters because it is delivered as a compact disc which is played in sync with the 30mm film. It is not subject to the pops and scratches normally heard when dust or bad splices interfere with the projector's ability to read the data from the film. It is higher resolution than Dolby Digital, but you need a trained ear or several hundred thousand dollars worth of audio equipment to distinguish between the two.

nVidia Soundstorm
A certification developed by nVidia and Dolby Labs. Motherboard manufacturers must use the nForce2 MCP-T south bridge and have an onboard SPDIF connection (optical or coaxial). The nForce2 is the only PC audio solution that uses the Dolby ICE (Interactive Content Encoder) to encode any audio signal into Dolby Digital 5.1. This is the only way to get digital 5.1 sound to your home theater receiver, the alternative is to use the 3 analog outputs coming out of the back of your sound card.

Note that using the analog outputs on a Soundstorm certified nForce board is not any better than using most other onboard solutions because the sound gets processed by the underperforming Realtek codec rather than a more powerful one like Creative's EMU or the one in your Home Theatre receiver.

Why use digital rather than analog connections?
+ : more tidy (1 cord rather than 6)
+ : integration with your receiver (You can use all the goodies you paid big $$ for on your receiver & speakers)
+ : longer cable runs with less signal loss
+ : bragging rights
- : 'jitter' on optical connections
- : cost of cables / external decoders
- : codecs / DACs on high end sound cards are arguably better than those in some receivers.
 
Originally posted by Igthorn:

Intel's High Definition Audio seems to be able to encode multichannel audio to AC3 as well.
http://usa.asus.com/products/mb/socket775/p5gd1/overview.htm
"The Dolby Digital Live technology from Dolby Lab encodes the multi-channel audio source into AC-3 bit-stream and outputs it to S/PDIF port in real time."

AC3Filter v1.01a RCx, a directshow filter (doesn't do games) capable of encoding multichannel PCM to AC3 on-the-fly (currently in beta). Works suprisingly well, low cpu usage. Some flaws, 48khz cards can't pass 44.1khz multichannel encoded to AC3.
http://forum.doom9.org/showthread.php?s=&threadid=76480

Originally posted by leukotriene:
You sure about this?
I was under the impression that Sony developed SDDS, and that DTS is a separate company (digital theater systems).

Incidentally, dts is not always lossless.
The original dts is not lossless to my knowledge, lossless dts is only a relatively recent innovation largely for theaters.

I believe the original dts compression algorithm is a lot less aggressive than DD and less susceptible to artifacting, but still definitely lossy.

EDIT: Yeah, I think I was right on both counts. dts is privately owned and not by Sony. It uses subband encoding with linear prediction and adaptive quantization like ADPCM.
Check out this pdf from dts and this interview with the president of dts.

The majority of dts (not including the new lossless) encodes use a "20 bit audio, 4 bit sync" track which is compressed with their ATP100 algorithm to 882kbps.

Originally posted by BO(V)BZ:
How to rip an album to a single file, and still be able to have full tracklists, etc

This guide, as it says, will tell you how to go beyond just a basic rip of an album, and do some fun stuff instead. Using a couple extra programs, you can easily rip an album and convert the file to one single uberfile, which is really handy for storage.
Note: This guide assumes that you have a little experience with foobar2000, but I'll keep it as simple as possible.

What you need:
Exact Audio Copy - This is by far the best CD ripping tool that you can download. It assures you of the best quality rip that you can possibly achieve with any particular CD.
http://www.exactaudiocopy.org/ - Go to download, get V.95 prebeta 5, then install that somewhere

foobar2000 - This is somewhat optional, but it makes some steps a lot easier by making it easy to tag your new album after you rip and compress it. On the download page, get the 'special' installer - it comes with piles of cool plugins.
www.foobar2000.org/download.html

Matroska - Matroska is a container format that lets you store audio/video information inside a 'shell.' One container that you are familiar with are AVI, audio-video interleaved. matroska lets us 'wrap' a file, in this case our full album, and then add extra data to it.
http://www.bunkus.org/videotools/mkvtoolnix/win32/mkvtoolnix-0.9.1.rar - direct link, or go to http://www.bunkus.org/videotools/mkvtoolnix/ and look under section 3.1.4, the 'windows' subheading.

http://www.bunkus.org/videotools/mkvtoolnix/win32/mkvtoolnix-runtime.rar You'll need these dll files from this.

FLAC - If you want to do it right, do it lossless =] FLAC, free lossless audio codec, is a lossless [duh] codec [duh again =]. It lets you make an exactly perfect copy of your CD, no quality loss at all, unlike MP3 or ogg. FLAC, however, does take up a lot of HDD space, so expect a full album rip to take anywhere from 300 to 500 MB. The actual bitrates per album will vary depending on what type of music you encode. From what I've encoded, soft stuff, like the Hexen soundtrack, averages around 650-750Kbps. Death-metal and industrial, however, can average 900-1100Kbps. Note that even the first example is probably more than three times the bitrate of most of your MP3's. It's certainly a tradeoff, and perhaps a case of diminishing returns, but hard drives are so cheap these days I don't think that using up a few extra Megabytes is that big of a deal.

My FLAC collection, comprising about 1300 songs total, takes up about 50GB.

FLAC, being lossless, has two other great advantages: you can transcode your files to any other format without any loss in quality, and you can burn perfect copies of the original CD. The latter point is fairly obvious, but the previous requires a little explaining. Transcoding is when you convert from one file format to another, such as converting a FLAC file to an mp3. Doesn't seem like a big deal to you? Well, think about transcoding an mp3 to ogg. Each compression scheme uses its own set of tricks to reduce the file size without killing the quality, but when you convert from one lossy codec to another, you end up running the music data through both of the compression algorithms. This means that any flaws your file had in it from being compressed to an mp3 will be magnified during the conversion.

With lossless files, transcoding is no problem. When you convert them to a wave file, and then to another file type, [which is what transcoding does, even if you are not aware of the intermediate wave file stage] you get the same quality as if you ripped the album to mp3 in the first place. This is a good way to make files for your portable; just transcode your FLAC files to mp3/ogg/lossy-codec-of-the-day.

I know that was a little longwinded, but lossless codecs represent the future, IMO, and it's important that you know the benefits and downsides. Here's a quick summary, if you want the Cliff's Notes:
+ perfect quality
+ easy to transcode to lossy codecs like mp3 or Vorbis
+ 'futureproof' - if a better lossless codec comes out, you can always transcode to that later
+ lets you make exact copies of the original CD, any time you want

- Large file size
- takes a little while longer to encode [takes my p4 2.26 about 5 minutes to do an album]

http://cyberial.com/flacinstaller.asp

Still here? Now that you have the tools that you need, let's get them set up, and I'll tell you what we are actually going to do with them =]

Step 1: Configure EAC.
After extracting EAC, run the executeable. This brings up the config wizard. Click next, and on the next screen, pick which CD drive you'd like to use. Click next again, select 'I prefer to have accurate results,' and click next once more. EAC will try to figure out what options your drive supports. Sometimes, it'll be right, sometimes not. Check the 'I don't trust these values' option, and put a CD in that drive, then click next. EAC will then test your drive and make sure that everything is kosher. This takes a minute or two. After detecting drive features, just click the next button several times. You won't need the LAME compressor for this tutorial, but if you want to install it anyways, feel free. Type your email address in for the freeDB option. On the last screen, select the 'I'm a beginner' option, which will keep things simple.

Now that you have completed the wizard, you'll be looking at a screen that lists all the tracks found on your cd. At this point, you're done for now!

Step 2: Configure FLAC and MKVmerge.
This step isn't too bad. What you'll need to do is extract the two Matroska related packs. [mkvtoolnix-runtime.rar and mkvtoolnix-0.9.1.rar] Take all the files from mkvtoolnix-runtime.rar, and put them in your windows/system32 folder. Take just the mkvmerge.exe file from mkvtoolnix-0.9.1.rar and put it in the same dir. You can put the other files in there, but you won't need them. Install the FLAC program that you downloaded. This has options for installing plugins for various audio players, which you can do if you want. Now, browse to C:\Program Files\FLAC , and copy the flac.exe file to windows/system32. You're done with Step 2 now!

Just so you know, if you put files into your system32 file, you can run them by hitting [win]-r, bringing up a run menu, and then just typing the name of the file. So, in our case, if you bring up a run menu, you can type 'flac' into it and it'll run the FLAC command-line encoder. This won't do anything yet, but it'll be helpful later.

Step 3: Install foobar2000.
Foobar2000 has full support for FLAC and matroska, and I'm very familiar with it, so that's why I'm using it. Take that special installer that you downloaded and run that. You'll be presented with a huge number of potential options to install, and I suggest that you isntall them all for now. If you really want to be picky, you can eliminate some, but make sure you keep the FLAC decoder, Matroska plugin, freeDB and masstaggers plugins [more on these later]

Install foobar2000 to the default location - C:\program files\foobar2000\ , as it'll save a little trouble for the last steps. After getting foobar installed, feel free to fool with it. It's gota lot of options [go to foobar > preferences] but all you need to do right now is type your email into the freeDB options. You can find those by looking under the 'components' section, and it's called freedb masstagger. At this point, you're done!

Step 4: Ripping your CD.
Open up EAC and put a CD in it [shiny side down =] After it detects it, hit alt-f7. This will rip the album to two files, a .wav file and a .cue file. The wave file is all the audio in one continuous stream, and the .cue file tells your audio player where the tracks start and stop. Save these to whatever drive you want, but you can make it easier on yourself later if you save this straight to c:\ , right on the root drive, and keep them named cdimage for now. After EAC finishes ripping, you're done!

Step 5: FLAC compression and converting to matroska.
This step isn't too hard, as I've already done the work for you =] What you want to do is cut and paste all this text into a text file, then save this file as 'matroska.bat' , placing this file, as with all the other files, into the windows/system32 folder. Note: If you didn't save your wav/cue files on C, then change the initial 'C:' , the top line, to whatever directory/drive. For instance, I save mine on the H drive, so I would change c: to h: , and leave the rest alone.

Code: 
c:
cd..
cd..
cd..
flac -8 --cuesheet=cdimage.cue cdimage.wav
mkvmerge -o cdimage.mka --chapters cdimage.cue --attachment-mime-type plain/text --attach-file cdimage.cue cdimage.flac
del cdimage.wav
del cdimage.cue
del cdimage.flac
c:
cd..
cd..
cd "program files\foobar2000\"
foobar2000 h:\cdimage.mka

This batch file does several different things: First, it encodes your wave file to a FLAC file, using the highest level of compression. Second, it takes that file and combines it with the cue file in a Matroska file. This is why we need Matroska: so we can keep all the track listings in the FLAC file, and be able to tag each individual track. Next, the batch file deletes all the temporary files, and finally, it opens the file in foobar2000 so we can complete the last step: adding tags to it!


Step 6: Tagging in foobar.
Now that the batch file has finished and foobar is open, we need to tag the tracks. Select all the files in the album, then right click on one of them. In the context menu that pops up, go to freedb > get tags. The freeDB window will pop up if the album is found. [amost any album you can buy is in that database, so I'm just going to assume it is] All you need to do is hit the 'tag files' button, and foobar will add all the tags you need. Now, close foobar, and browse to C:\ , or where-ever you decided to put the file. Name the file however you wish, such as 'Spineshank - 2001 - The Height Of Callousness.mka' . Move the file to where you keep your music, and you are done!

You've successfully made a losslessly encoded, single-file-per-album rip, using 2 geek tools and a batch file =]

Originally posted by XLShadow:
My mic is not working. (I do this for myself all the time)
  1. Be sure its pluged into the right jack ( :D most of the time, this fixes eveything)
  2. Be sure its making good contact. For good measure pull it out and plug it back in. It should go in with a nice tactile click. (If its too loose, your mic jack may be bad and you might need a new soundcard or a sodering iron :p )
  3. Are you using kx-drivers? No: Continue to next like. Yes: Jump to kx-section.
  4. Open up the Control Pannel.
  5. For Windows XP (classic view): Open Sounds and Audio Devices. For Windows 2000: Open Sounds and Multimedia. (They are both icons of a speaker)
  6. Click on the Audio tab.
  7. Make sure the proper soundcard is selected for Sound recording. Click on the Volume... button. (The middle one. Right under the soundcard pull down) This will bring up the recording volumes.
  8. Check to make sure if the Microphone is selected for recording and the volume is about half way up.
  9. Go into Options > Advanced Controls. An Advanced button should appear under the mic slider. Click it (if you can, not all soundcards support this) and make sure Microphone +20dB Boost box is checked. This is required for microphones that ONLY plug into the soundcard. (If your mic has its own power cord or is powered by an amplifier, you'll want this unchecked)
  10. Close all windows, we are done here. You can adjust volume here after the loopback test.

kx-drivers mic setup
kx-drivers have a different setup because the kx-mixer replaces the windows sndvol32.exe
  1. I need to get back to this one...

Now we need to test what is called a loopback: we make sure the soundcard is reading the mic by playing what you say over your own speakers.
  1. Open the Playback volume mixer. For most people, double clicking on the speaker icon in the system tray will take you there. (If not, try typing in sndvol32 in the Run box)
  2. Microphone isn't one of the slider bars by default; go into Options > Properties.
  3. Set the master volume down to zero.
  4. Unmute the microphone.
  5. Repeat a test phrase into the mike, "check check 1 2 3" while you slowly turn up the master volume. Stop when you hear your self over the speakers.
    If you cannot hear yourself even after maxing all the volume sliders, stop. Something is wrong with your hardware. Test the mic on another known-to-work soundcard. Also try reinstalling / updating drivers and try loopback testing again.
  6. Mute the microphone for playback and close the volume mixer. For chat/voice commands/recording you shouldn't have to hear yourself over the speakers. This was just a test to make sure it is now working.

Originally posted by GodsMadClown:

Do I need that funny cable that connects my optical drive to my soundcard?

No. It's an obsolete means of transmiting audio from an optical drive. Modern versions of Microsoft Windows support digital audio extraction, which makes the analog cable redundant.
 
How do I make my music play through more than two speakers?
Click to expand...

I posted the question. So far, one lame answer stands. Shall more reasonable minds prevail? I came to the realization that a FAQ could be answered in the same manner as the Qs are frequently answered.

Depending on the card there are several ways to enable a multichannel upmix.
The simple one is to enable hardware acceleration in Winamp.
If you are not using Winamp then one of Foobar's DSPs will do the trick.
For Creative cards it's a little harder but CMSS usually does the trick.
Other cards like my Terratec DMX6 fire can use
Sensaura but it tends to suck.
 
l33trequiem said:
hey guys....
when i open up a program and i have music playing...it will skip and crackle until the program is opened...ive downloaded the newest drives for my sound card but it still does it...any ideas on how to fix it?

thanks
Click to expand...

GodsMadClown said:
Try moving your soundcard to the farthest slot from the video card. Also, remove any cords connecting it to an optical drive. They are unecessary.
Click to expand...

Also, make sure you have the latest drivers for your soundcard and motherboard. Something funky could be happening on the PCI bus. A BIOS update might even help.

Anything else that I forgot?
 
Soundstorm needs to be explained further, so that people know when they are actually using it. When people are saying "I have soundstorm and my Logitech Z640s are connected to my computer, should I get a TBSC? Will it improve the sound?"

People need to know that in the above situation they are not using Soundstorm nor are they getting any of the benefits of the improved sound quality from soundstorm. Soundstorm is only being used when in fact the _optical_ out connection on a motherboard is being used to extract the sound from the board. All other comparisons are only involving the on-board C-Media or Realtek chip.

I've seen this time and time again where it's simply a Placebo effect and 90% of those who own Soundstorm capable boards are not using the digital connections.
 
added

[size=+2]Why use digital output?[/size]
By using a digital connection, one can bypass the DAC and output stages of the integrated sound on the motherboard. These are usually built with cost in mind, often at the expense of quality. To use Soundstorm to its fullest potential, it is best to use the digital output into an outboard decoder of some sort.
​
Click to expand...
 
Now all you need is a speaker FAQ.

P.S. I've been big into audio for years and I noticed you said that one can't have digital surround without Soundstorm, but I do, and I have since 1999 too :D
 
The good old SB Live can output a digital stream that can be decoded from DSP speakers with the mini-rca digital (only time I've ever seen one used for digital) if you've got DSP surround speakers with the appropriate input.You can feed it 5.1 from software and it will recognize it and play it back, not just the weird noise that would normally happen when a DAC can't accept the digital content and convert it. I'm pretty sure the Harman Kardon's I've got were made specifically for Dell, which this system was sold through. For DSP its actually the most convincing I've heard, but I haven't heard the new Niro's yet.
 
I'm not sure if it is or not. I know its decoded by the speaker's DAC's so its whatever they can handle. Finding info for its actual specs was nearly impossible. Even the Dell manual it comes with doesn't say what the accepted encoding format is when digital inputs are used.

One page says 16bit/48KHz SPDIF but it uses teh mini RCA plug.
 
I am wondering if the soundcard is simply outputting a standard PCM signal, with the speakers' decoder matrixing the signal across the extra channels. Know what I mean?
 
If its just a Pulse Code Modulated stream it wouldn't have discrete 5.1 channels likely, which is probably the case since its not using anything like 24bit/192KHz or its compressing it all ot fit within that 16bits. I've got a dedicated reference system with 110WPC RMS Event TR-8 studio monitors being run off a Nakamichi SS-8 all power conditioned through a Monster Power HTS 3600 powercenter. The HK-595's sound remarkably good for music considering what they are but what's even better is that they do a great job of simulating surround. So I'm not sure if its PCM 48KHz with discrete but it does also have a true multichannel analog output on the card with a proprietary multi input on the subwoofer enclosure.
 
Added

Dealing With General Equipment

Which is better, X or Y?

Determining what is "best" is a inexact science, to be sure. A determination can depends on many variables, including the equipment and space used for the implementation, the criteria used for evaluation, and, ofcourse, subjective opinions. So, deciding what is best is really a highly personal decision. While asking for more informed opinions might introduce you to new alternatives, ultimately the best way to evaluate any given piece of equipment is with experience. You should audition it. Preferably, you would audition it in the final implementation. Failing that, you should audition it with as many of the variables controlled for.

...TBC...
Click to expand...
 
The link to the hacked Audigy 2 ZS drivers isn't working. At least, not right now.
 
Not like that is any big loss cause there are now hacked Audigy4 drivers floating around that offer updated features. ;)
 
Really? Where at? Perhaps the FAQ should be updated to include that. :)
 
Yeah, it is not the best solution. When I have my server up, I notice that most people seem to cancel the download anyways (30kbps up until Verizon puts my fiber optic in @ 1mbit up :) )

Just let me know and I'll get my server up and running, it takes just 30 seconds.
 
Added:

[size=+2]How do I make my front panel audio work with my Creative card?[/size]

You need to rig up some way of connecting your various input/output connections to the front panel connections of the card. Here is a pinout guide to the front panel connector of the Audigy series (Audigy - Audigy2 ZS).

Audigy_front_pin_diagram.jpg


  1. analog ground
  2. analog headphone left channel
  3. audio backpanel mute (short to ground to mute the backpanel output, for automute when headphones are plugged in)
  4. analog headphone right channel
  5. same as #3
  6. microphone input
  7. key pin (shouldn't be there)
  8. VRef out (voltage reference for microphone)
  9. microphone input mute (ground when microphone isn't plugged in, +12VDC
  10. audio cable detect (not normally used, will be ground when headphones are plugged in)

Added:

[size=+2]What is the best music compression format?[/size]

Really, the best way to answer that question is try them out and compare them one against the other. The ABX comparison plugin for Foobar2000 is a handy tool for this task.

If you don't want to rerip all your music if you upgrade to better your equipment, you would be well advised to consider encoding to some lossless compression format like Monkey's Audio or FLAC. This file can then be used as a source for the lossy format of your choosing for your portable player. While losslessly encoded files can occupy much more storgae space, you can be assured that your music will always sound its best.​


Also, this:

Digital signals over SPDIF are not all created equal. Here's a quick rundown of the three basic encoding techniques that most receivers can understand:
  • LPCM (Linear Pulse Code Modulation): This is the basic 2-channel stereo format used by most receivers. It is uncompressed, high-bitrate, lossless stereo information.
  • DD (Dolby Digital): This is the standard 5.1 channel audio format for DVDs. It uses a very efficient, but lossy compression.
  • DTS (Digital Theater Systems): This format competes with DD by providing a higher-bitrate compression algorithm that is slightly less "lossy" than the Dolby format.
Every soundcard that has a digital output can encode an LPCM stream. They can also usually be configured to "pass-through" streams from content with Dolby Digital or DTS audio, such as DVDs.
​

became this:

Digital audio is typically carried in a signal that conforms to the S/PDIF (Sony Philips Digital InterFace, with either an optical or coaxial connection. Digital signals over S/PDIF are not all created equal. Here's a quick rundown of the three basic encoding techniques that most receivers can understand:
  • LPCM (Linear Pulse Code Modulation): This is the basic 2-channel stereo format used by most receivers. It is an uncompressed lossless stereo (2 channel) audio format.
  • DD (Dolby Digital): This is a widely used 5.1 channel audio format for DVDs. It uses a very decent, but lossy compression.
  • DTS (Digital Theater Systems): This is another 5.1 channel audio format that many consider superior to DD because it uses a higher-bitrate compression algorithm that is less "lossy" than Dolby Digital.
Every soundcard that has a digital output can encode an LPCM stream. They can also usually be configured to "pass-through" streams from content with Dolby Digital or DTS audio, such as DVDs.
​



Comments or advice?
 
GodsMadClown said:
Anyone want to write up the entry on interference?

If not, this will serve as a general reminder to myself that this needs to get into the FAQ.

http://www.hardforum.com/showthread.php?t=899260
Click to expand...

You have full permission to edit my spelling mistakes, but here's a shot for troubleshooting noise issues, using my "by escalating order of price" system I use to troubleshoot about everything.

-------------------------------------

Noise in your output can be caused from many sources.

Best way to troubleshoot IMO is based on cost of repair, this order is "minimize uneeded parts, tweak settings, check extrnal cabling, check hardware".

First keep in mind a digital signal is pretty much protected against noise. Dropped packets and jitter is somethhing else entirely, and is not "noise".

With that aside, you can be assured that ANY analog line running INSIDE your case can be a problem. If you are using the "cd-audio" cable from your CD-rom drive, please remove it asap, since there are better ways to transmit the signal, with far less chance of picking up noise as well. Also disable any uneeded inputs from your soundcard's mixer. ALWAYS disable mic and aux-in when not using.


Next look at your output cables. shielded cables will help allieviate any inductive noise issues from neighboring cables. Problem with digital signals is they tend to produce lots of EMI (electro-magnetic intereference), which can screw up analog signals that just happen to be of the same frequency range as the EMI.

Now if the noise still persists, after insuring the uneeded inuts are disabled, no analog cables inside the case, and shielded cables...then you can be certain it's a hardware issue of some sort.

To find out the hardware issues, you need to carefully examine your PC, and start ruling out peices of hardware. If it's watercooled, it could be the pump, or the AC line that is going inside...keep in mind AC frequency is 60hz...which is in the audible range as bass notes. If it happens when using your USB mouse, grab a PS/2 one and see if the noise persists. Basically keep searching until you find it. Sadly there is no "one thing" that we can say to start with on hardware induced noise, since it could be anything. In some cases with onboard sound, you may fnid they did not incorporate proper shielding to their design, allowing noise to get into the line even before it gets to the miniplug jacks on the back of the case. In those instances with onboard sound and bad design, the best fix is a quality soundcard (There are some great ones around for $30).
 
How's this? It's probably too long and could stand some trimming after I come back to it in a few hours/days/weeks.

Steve, I hope you don't mind me altering your words too much. I tried to tighten up the structure a little bit.



Noise in your output can be caused by many sources. An effective way to troubleshoot is based on cost of repair, attemping no cost solutions before trying more expensive fixes.

If you are using the "cd-audio" cable from your CD-rom drive, unplug it and throw it in the trash. The IDE connection has been able to cary CD audio data since Windows 95, and there is no good reason not to use it. Also disable any uneeded inputs from your soundcard's mixer. It is a good practice to always disable the mic and auxulilary inputs when not using. These controls can be accessed by clicking the the "advanced" button in the volume control section of "Sound and Audio Devices" in the Windows control panel. Finally, if using a PCI soundcard, it should be plugged into the PCI slot farthest from the Video card, to reduce the possibility of interference .

Next, look at your output cables. Try to move them away from other cabling, especially power cords. If proximity is unavoidable, try to run the lines perpendicular to one another. Shielded cables will help allieviate any inductive noise issues from neighboring cables. Ask in the forum for reccomendations on good shielded cables if you think it may solve your problems. Digital signals can produce a good deal of of electro-magnetic intereference (EMI), which can play havoc with analog signals of the same frequency range as the EMI.

Now if the noise still persists, after insuring the uneeded inuts are disabled, no analog cables inside the case, and resolving potential external cabling problems , then you can be certain it's a hardware issue of some sort.

To find out the hardware issues, you need to carefully examine your PC, and start ruling out peices of hardware. If it's watercooled, it could be the pump, or the AC line that is going inside...keep in mind AC frequency is 60hz...which is in the audible range as bass notes. If it happens when using your mouse, switch it over to either USB or PS/2 and see if the noise persists. You will need to keep searching until you find it. Because there can be a wide variety of causes, there is no single solution.

In some cases with onboard sound, you may find they did not incorporate adequate shielding to their design, allowing noise to get into the line even before it gets to the miniplug jacks on the back of the case. In those instances of a poor sound implementation, the best fix is a quality soundcard. There are some very decent cards that cost around $30.

Finally, if the noise still persists, you could consider moving to a digital solution. A digital signal over S/PDIF is very well protected against noise. Dropped packets and jitter are something else entirely, and is not "noise".

Here's a quick list of stuff to check, roughly in order of increasing expense.

  • Unplug the skinny audio cable that connects your soundcard and optical drive.
  • Mute all unused input/output channels in the advanced volume control panel.
  • Plug audio card into slot farthest from video card.
  • Try both types of mouse connection, PS/2 and USB.
  • Reposition audio cables away from, or perpendicular to other cables.
  • Buy shielded cables.
  • Buy another soundcard.
  • Buy a reciever and use a digital connection.
 
Looks great man.

I admit, I suck at writing out my thoughts...my mind rambles and bounces all over the place. Imagine an ADD person on meth, and you got a basic approximation what goes on inside my head :p
 
I did a search just so I could figure out what "DICE" was, and I think that this FAQ is handy, maybe sticky?
 
You ready to tackle the inevitable "what headphones are best for me" question? :D

I think a basic headphone primer would help answer a lot of the headphone questions that we see........... but that is easier said then done. :)
 
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