Did you find an FLAC file on your computer and wonder what program should open it? Maybe someone emailed you an FLAC file but you’re not sure how to use it. Perhaps you tried to open the FLAC file but Windows told you that it could not open it.

Before you can open an FLAC file (assuming it’s even a file format that’s intended to be viewed or edited), you’ll need to determine what kind of file the FLAC file extension refers to.

A file with the FLAC file extension is a Free Lossless Audio Codec file.

The Free Lossless Audio Codec format is an open source audio compression format. Audio compressed via the Free Lossless Audio Codec is lossless, meaning that no sound quality is lost during the compression, unlike other popular audio compression formats like MP3 or WMA.

Other types of files may also use the FLAC file extension.

How to Open an FLAC File:

The easiest way to open an FLAC file is to double-click on it and let your PC decide which default application should open the file. If no program opens the FLAC file then you probably don’t have an application installed that can view and/or edit FLAC files.

Warning: Take great care when opening executable file formats received via email or downloaded from websites you’re not familiar with. See my List of Executable File Extensions for a listing of file extensions to avoid and why.

FLAC files can be opened by nearly every popular media player but usually only with a plugin or extension installed. Many Windows programs also exist for encoding FLAC files from CD or other sources. The Free Lossless Audio Codec community hosts a website dedicated to the format and keeps a well maintained list of programs that support FLAC: FLAC Download Page.

If you find that an application on your PC does try to open the FLAC file but it’s the wrong application or if you would rather have another installed program open FLAC files.

How to Convert an FLAC File:

There are two main ways to attempt to convert an FLAC file to another file type:

* Open the FLAC file in its default program and choose to save the open file as another file format.
* Use a File Conversion Online Service or Software Program to convert the FLAC file to another file format.

The first option involving opening the FLAC file in its native program is preferable because it’s both easier and will probably result in a more accurate file conversion. Of course if you don’t have a program that opens FLAC files, a third-party file conversion tool (the second option) could be very useful.

Important: You cannot usually change a file extension (like the FLAC file extension) to one that your computer recognizes and expect the newly renamed file to be usable. An actual file format conversion using one of the methods described above must take place in most cases.

source: http://pcsupport.about.com/od/fileextensions/f/flac-file.htm

* Format name: WMA / Windows Media Audio
* Format type: Audio
* File extensions: .WMA
* Compression Type: Lossy

The term WMA (WindowsMedia Audio) is normally used to refer to Microsoft’s first audio codec that it originally developed to compete with the highly successful MP3 format. WMA is actually a set of audio compression codecs that form part of Microsoft’s multimedia framework; the other codecs that have been developed are, WMA Pro, WMA Lossless, and WMA Voice. For simplicity, WMA in this article will refer to the format used by media software players, and various compatible hardware devices such as media/MP3 players. The WMA format is a lossy audio format that has become one of the most popular formats in use today.

Popular Compatible Hardware:

There are now many hardware devices on the market that are compatible with the WMA format. As well as PlayForSure devices (a Microsoft certification for hardware and online services) there is also a significant number of uncertified hardware that supports WMA playback. The majority of these compatible devices are portable MP3/media players, but there are also other consumer electronics such as DVD players, cell phones, in-car/home entertainment systems that can handle WMA files. Incidentally, if you own a PlayStation Portable (V2.60 or higher), you can use it to listen to WMA audio files.

Sampling Frequencies & Bitrates:

The WMA format provides the following sampling frequency range:

* 8 – 48 kHz

The standard sample rate which is commonly used is 44.1 kHz – this is the same frequency as used by the compact disc.

The bitrates that can be used with the WMA format are flexible and gives this format an advantage over MP3 when encoding audio. The bitrate range is:

* 5 – 320 kbps.

Compression Algorithm:

The WMA format uses a lossy compression algorithm. It is a transform codec that uses a psychoacoustics model and employs the modified discrete cosine transform (MDCT) to process the WMA bit stream.
Digital Rights Management:

WMA files can be protected with DRM using Microsoft’s WMDRM system. The files that are downloaded from online digital music services are encapsulated using ASF (Advanced Systems Format); this is a container format that is used for audio and video files. DRM-protected WMA files are usually downloaded from subscription-based online services. If a subscription ends, then the files that the user has downloaded become unplayable. WMA protected audio files can also be downloaded from a la carte services and come with copy restrictions designed to inhibit distribution.

source: http://mp3.about.com/od/profiles/p/WMA_Profile.htm

The MP3 audio format has been the de facto standard for online digital music files since the early 1990′s. Almost every computer media player recognizes and plays MP3 files, even if its primary format is a proprietary one (such as AAC with iTunes, or WMA with Windows® Media Player). MP3 files work with virtually every brand of portable digital music player, which is why the term “MP3 player” is often used to describe these devices.

In order to get the most out of this ubiquitous format, it’s important to understand what an MP3 file is, and what it does to your music. With that knowledge, you’ll be better able to set up and manage your digital music collection to suit your needs.

How it works

MP3 is a digital audio codec; that is, it’s a method of compressing and decompressing digitized sound. The digital information on a standard audio CD requires about 10 megabytes per minute of music. When a song is ripped from a CD to a computer and converted to an MP3, that same minute of music typically is reduced to about 1 megabyte — a tenth of the original file size.

The MP3 codec shrinks the source file by removing portions of the original signal considered to be essentially inaudible — a technique know as “perceptual coding.” MP3 and other codecs are lossy formats of compression. That is, some frequencies are lost in the encoding process, and can’t be restored by reconverting the file to its original format.

The trade-off between sound quality and storage space

Despite its lossy nature, MP3 can still deliver near-CD sound quality. Just as you can save your digital photos at higher or lower resolutions to save memory space (with a proportional loss in detail), so too can you adjust the “resolution” of an MP3.

The bit_rate, or average amount of data required per second of music, determines the audio resolution of an MP3. The higher the number of kilobytes per second (kbps), the closer in sound quality the MP3 is to the original source — and the larger the file size.

Most Internet download sites are primarily concerned with facilitating faster downloads, which means keeping MP3 files as small as possible. Many offer MP3s at 128 kbps, generally considered to by the lowest acceptable level of sound quality.

128 kbps MP3s provide a balance between sound quality and convenience. The music sounds “good enough,” and even a small 1-gigabyte portable player can hold about 32 hours of 128 kbps MP3s, or roughly 480 songs.

You can also compromise by using a variable bit rate, or VBR. Rather than encoding audio at a constant rate, VBR encoding ups the bit rate for complex passages, and uses a lower rate for simple ones. A VBR-encoded sound file usually has more sonic detail than one encoded at the same constant bit rate, yet it’s a smaller file than a constant bit rate file of comparable resolution.

Hear more of the original detail with higher sound quality

Although the overall character of the music is preserved, the greater the compression, the more sonic details are lost. Extremely high and low frequencies usually get discarded with even slight compression. Although considered inaudible, they reinforce harmonic frequencies that “shade” the sound, giving it much of its fullness and presence. Further compression can diminish the differences between loud and soft passages, decreasing dramatic impact. Extreme compression — down to 64 kbps and lower — can completely flatten the sound, making it harsh and muddy.

By contrast, MP3 files of 192 kbps, 256 kbps or greater preserve most of the sonic information of the original WAV file. Acoustic instruments tend to keep their natural warmth at these resolutions, and electronic instruments sound fuller while retaining their punch.

So how can you make sure the MP3s you’re importing into your music library are at the best sound quality for you? Depending on the source, you can have some control over the process.

1. Importing music from CDs — Change the default settings of your PC’s media player. Almost all of these programs let you adjust the MP3 resolution from the standard 128 kbps up to at least 192 kbps. Many let you customize the setting by typing in your own number.
2. Purchasing music online — While most sites only offer songs at 128 kbps, some offer tracks at a higher resolution. Many of these higher-resolution tracks also come without copy protection, giving you the added advantage of enjoying your purchased downloads in more ways and on more players. Many of these tracks are offered at 256 kbps resolution.
3. Downloading from other sources — Band websites, podcast directories, and other sources of Internet music usually offer MP3s at 128 kbps. Try downloading other formats (discussed below) if your player supports them. You might find one that sounds better to you than the others.

Other codecs, too

As mentioned above, sometimes you aren’t offered a choice of MP3s with different levels of compression. Instead, you might be offered a choice between different formats.

Each codec has its own algorithms to determine what to discard, and so the same song saved in different formats can vary slightly in sound. Some people find they like the way a particular codec shapes the sound, and prefer that format over MP3.

There are also formats that retain all the information of the original files, and just store them more efficiently in a slightly smaller space. Because all of the information is retained, these are known as lossless codecs.
In the end, it’s up to you

Keep in mind that you can always compress a large file into a smaller one, but you can’t restore the resulting file back to its original form. The information lost in the compression process is permanent. So it’s best to choose the highest resolution when you first import a track. A 128 kbps file can’t be “bumped up” to 256 kbps any more than a low-resolution photo can be blown up to a poster-sized print.

The most important thing to remember when you’re changing settings to improve sound quality is this: your ears are the final judge. Some people can distinguish between a 256 kbps and a 192 kbps MP3. To others, there’s no appreciable difference between a 128 kbps file and the original CD track.

Only you can determine the ideal MP3 resolution for your music. But it’s best to do so only after you’ve given the higher resolutions a fair audition. You might be surprised at what you’ve been missing.

source: http://www.crutchfield.com/S-Vfy82PTIUG2/learn/learningcenter/home/mp3.html

* Format name: MPEG-4 AAC / Advanced Audio Coding.
* Format type: Audio.
* File extensions: .aac, .m4a, .mp4, .m4b, .m4p, .m4r, .m4v.
* Compression Type: Lossy.

The Advanced Audio Coding (AAC) format was developed by a group of companies with the goal of improving many of the drawbacks to using the MP3 standard. The AAC format is still a lossy audio format, just like MP3, but typically produces higher sound resolution (especially at low frequencies) due to the many improvements over MP3. There are many companies that have adopted this as their format of choice; Apple for example uses it for their iTunes Store.

Popular Hardware:

* As discussed, Apple’s iTunes Store uses AAC encoded files with DRM for most of its music catalog. Apart from a few exceptions, the iPod and iPhone are the only hardware devices that are capable of using the FairPlay DRM system.
* In the home video console market, Sony’s PlayStation 3, PSP, and Nintendo’s Wii utilize the AAC format in favor of MP3.
* Portable media players such as the Microsoft Zune and Cowon A3 have built-in support for AAC.
* Other hardware that has embraced the AAC format includes, cell phones, PDAs, and wireless audio streaming devices.

Sampling Frequencies & Bit Rates:

The AAC format provides the following sampling frequency range :

* 8 – 96 kHz and above.

As for the MP3 format, it is the 44.1 kHz sample rate that is used the most; this is the same frequency that is used for the compact disc.

The advantage of using the AAC format compared to MP3 is that it is significantly more flexible when it comes down to bitrates – especially at the lower end. The bitrate range is:

* 8 – 320 kbps.

Compression Algorithm:

The AAC format uses a lossy compression algorithm. Built into AAC is a suite of modules or profiles that make it a very flexible compression system. The most popular profile that is used to encode audio is the Low Complexity setting. This is the simplest module and the most commonly used.

source: http://mp3.about.com/od/profiles/p/AAC_Profile.htm

Format name: MPEG-1 Audio Layer 3
Format type: Audio
File extension: .MP3
Compression: Lossy

The MP3 format’s history is legendary when it comes to digital music and the Internet. It has single handedly revolutionized the way we listen to and share music. Even its name has become part of our everyday language; MP3 music and MP3 players are terms that are used to describe all kinds of formats and players. It is this impact on the world that makes the MP3 format truly impressive. Over a decade since it first appeared, the MP3 format continues to dominate over all other audio formats in terms of global use.

Sampling Frequencies & Bit Rates :

There are three sampling frequencies that are available for MP3 encoding which are:

* 32, 44.1 and 48 kHz.

It is the 44.1 kHz sample rate that is used the most; this is the same frequency that is used for the compact disc.

There are a selection of bit rates that the MPEG-1 Layer 3 standard sets out, and these are:

* 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256 and 320 kbps.

The most popular bit rate used for MP3 files is 128 kbps as this strikes a good balance between audio quality and file size.

Compression Algorithm:

The compression algorithm that the MP3 standard uses is a lossy one. The MP3 encoding process takes advantage of psychoacoustic algorithms that filter out frequencies that humans are unlikely to hear. Low frequencies are either filtered out or are converted to mono signals – this requires less storage space. The human ear can’t detect directions of low frequencies very well and so a stereo signal isn’t necessary. Another way that MP3 compression can reduce file size while keeping the quality is to drop quieter sounds in preference for louder sounds; it is unlikely that the listener will notice any difference.

Audio Quality Influencies:

Factors such as, sampling frequency, bit rate and compression algorithms all influence audio quality. The MP3 format is old compared to modern audio formats, like AAC and so its performance in certain scenarios isn’t as optimized. Using MP3 to encode audio at low bit rates produces a lower quality of sound compared to newer formats like AAC or WMA. Using CBR as opposed to VBR can also degrade sound quality. CBR produces MP3 files that are typically not as good as their VBR equivalents. VBR encoding can produce a higher quality of sound because the bit rate varies to optimize quality of sound and file size.

source: http://mp3.about.com/od/musicformats/p/MP3_profile.htm

Computers use sound cards to receive and send analog and digital sound. Without a working sound card, no sound will come out of your computer’s speakers because there is no outlet for it. Sound cards are also the foundation for voice chatting because they digitize your voice after it moves from the microphone and into your computer. Digital audio is then converted back to analog on the way out to your headset or speakers, or those of another user. To fully test a sound card and diagnose sound card or system issues, it is worthwhile to have on hand a second sound card that is known to be working.

Things You’ll Need:

* Working headset with microphone A sound card that is known to be working

Instructions

Step 1

Test the sound card by playing music on your computer’s media player to see if you can hear any sound through your headset. If you do not hear any sound through your headset, check your microphone jack to make sure your headset is plugged into the proper place, and test the jack to see if it is loose.

Step 2

If your jack is loose, replace your sound card with a working one and repeat the sound test with music.

Step 3

Open your computer’s sound recorder by going to “Start,” choosing “Programs,” choosing “Accessories,” choosing “Entertainment,” and then choosing “Sound Recorder.” Make sure your microphone is not muted. click on the record button and speak into the microphone. Click stop and playback the sound. If your voice plays back and you were able to hear music through your headsets during the sound check, your card is fully working.

Step 4

If your recorder does not play your voice back, check your microphone jack to make sure your microphone is plugged into the proper place, and test the jack to see if it is loose. If your jack is loose, replace your sound card with a working one and repeat the microphone test using the playback feature.

Step 5

Test the sound in a multimedia application like a movie or a computer game to make sure that the sound is functioning properly in a more sound-intensive environment. If you have no sound while testing a multimedia application, replace your sound card with a working one and retest your multimedia application.

Step 6

Test all sound jacks connected to your sound card because some cards will have jacks for rear, center, and front channels. If there is still no sound after checking the jacks, replace the sound card with a working card and rerun your sound checks.

Tips & Warnings

*
Check to see if your sound is muted by the computer’s master control. Look for your speaker in the bottom right portion of the screen and make sure it’s active and is not crossed out, which would indicate it is muted. You can also double-click on it and check if the master control has the “mute all” function selected under the master volume. Uncheck “mute all” and continue testing. If you are still are not receiving sound. Make sure the sound card’s updated drivers are installed. Download the updated sound drivers from the sound card manufacturer’s website.
*
Do not switch out sound cards when the computer is on. This could cause a short in your system and could hurt other components or ruin the whole system. When switching out the sound card, make sure you are statically discharged by touching your computer’s metal case so the system is not shorted out when you put the good sound card in. Make sure to anchor the sound card in the port with a computer screw. Components that are not anchored into the computer can cause electrical short circuits and destroy themselves and other components.

source: http://www.ehow.com/how_4811539_troubleshoot-computer-sound-cards.html

What Is a Sound Card?

1. A sound card is the device used to control the input and output of audio files on a computer. For the most part, the sound card is used by the computer user to hear the sound on music and video files and from video games. In the past, beeps were the only sound that could be produced by computers. Because of the invention of the sound card, more complex audio can be accessed and processed by a computer. Sound cards now come installed in all of the computers on the market. They have become common and important; they are now required by many mainstream computer systems and programs.

How Does a Sound Card Work?

2. Sound cards have a standard operation for controlling sound on computers–of processing the input and output of sound. The input comes from other devices such as microphones or MIDI musical instruments. The sound card takes the sound from the device, records it and then processes it. For output, the device takes instructions from the program and plays the sound through speakers connected to the computer.

Sound cards use the other elements that are installed inside of it to handle the tasks that are assigned to it. The digital signal processor (DSP) is a chip in the sound card that receives and handles the incoming sound signals. It then manipulates the signals to access the sound. The digital-to-analog converter (DAC) controls the outgoing sound. It takes the sound frequencies from the program and converts it into frequencies that are audible to the human ear. The analog-to-digital converter (ADC) does the same job as the DAC but in reverse. It changes the audio for the sound card to be able to process it. Digital audio is for human ears and analog audio is what the computer understands.
Other Components

3. The Musical Instrument Digital Interface (MIDI) takes the signals from connected musical instruments and processes it. Many musicians and songwriters are now using the same type of technology to write and record songs. Keyboards and guitars are connected to computer systems and sound is recorded with sound cards and special software programs.

The sound card has its own memory source. Like many of the devices used today, sound cards take advantage of the power of flash memory, or Random Access Memory (RAM). The memory is temporary and is only stored until the sound card has a chance to process the incoming and outgoing data. Once the stored data is not longer needed by the sound card, it is removed from the memory.

The sound card has evolved over the years since its creation. When first conceived, FM Synthesis was the method used by sound cards to create and interpret sound. This means that the sound card actually imitated the sound that closely resembled the audio file that it was accessing. Now a more realistic approach, called Wavetable Synthesis, is being used. This is when the sound card takes a piece of the real sound and copies it to create the full version of it. Sound cards are now able to produce a more realistic sound than earlier versions.

source: http://www.ehow.com/how-does_4568118_sound-card-work.html

Introduction

Coaxial and optical cables are used to make a digital audio connection between a CD or DVD player and a receiver. Both types transfer a digital signal from one component to the other. So, what is the difference between the two types and which is better? The answer depends on whom you ask, but in general the performance differences are negligible. Here are some facts about coaxial and optical digital cable connections.

Coaxial Digital Audio Cables

A coaxial (or coax) cable is hard-wired using shielded copper wire and is generally the most rugged. Each end of a coaxial cable uses familiar RCA jacks, which are reliable and stay firmly connected. However, coaxial cables may be susceptible to RFI (radio frequency interference) or EMI (electro-magnetic interference) and if there are any existing ‘hum’ problems with the system (eg, a ground loop) a coaxial cable may transfer the noise between components. Coaxial cables also lose signal over very long distances, but for home use that’s usually not a problem.

Optical Digital Audio Cables

An optical cable (also known as Toslink) transfers an audio signal using a red light beam through plastic fiber optic cable. The signal that travels through the cable must be converted from an electrical to an optical signal in the CD player, then back to an electrical signal in the receiver. Optical cables are not susceptible to RFI or EMI noise because it is light, not electricity that is traveling through the cable. Optical cables are more fragile than a coax cable and cannot be bent too tightly or pinched. The ends of an optical cable use an odd-shaped connector that must be inserted correctly and is not as tightly connected as a coaxial cable with RCA jacks.

Conclusions

Your decision about which cable to use may be based on the type of connectors on the components because not all audio components have both optical and coaxial. I prefer coaxial cables to optical because of sound quality, although the differences are very subtle and are probably noticeable mostly in high-end systems. In general, either type provides good sound quality.

source: http://stereos.about.com/od/accessoriesheadphones/a/digaudiocables.htm

Computer audio is one of the most overlooked aspects of a computer purchase. With little information from the manufacturers, users have a hard time figuring out exactly what it is they are getting. In the first segment of this series of articles on Understanding Computer Audio, we look at the basics of digital audio and the specifications may be listed. In addition, we will look at a couple of the standards that are used to describe the components.

Digital Audio

All audio that is recorded or played through a computer system is digital, but all audio that is played out of a speaker system is analog. The difference between these two forms of recording play an important role in determining the ability of sound processors.

Analog audio uses a variable scale of information to try and best reproduce the original sound waves from the source. This can produce a very accurate recording, but these recording degrade between connections and generations of recordings. Digital recording takes samples of the sound waves and records it as a series of bits (ones and zeros) that best approximate the wave pattern. This means that the quality of the digital recording will vary based on the bits and samples used for the recording, but the quality loss is much lower between equipment and recording generations.

Bits and Samples

When looking at sound processors and even digital recordings, the terms of bits and KHz will often come up. These two terms refer to the sample rate and audio definition that a digital recording can have. There are three primary standards used for commercial digital audio: 16-bit 44KHz for CD Audio, 16-bit 96KHz for DVD and 24-bit 192KHz for DVD-Audio.

The bit-rate refers to the number of bits used in the recording to determine the amplitude of the sound wave at each sample. Thus, a 16-bit bit-rate would allow for a range of 65,536 levels while a 24-bit allows for 16.7 million. The sample rate determines the number of points along the sound wave that are sampled over a period of one second. The greater the number of samples, the closer the digital representation will be to the analog sound wave.

With this general understanding, what exactly should one look for when examining the specifications for an audio processor? In general, it is best to look for one capable of 16-bit 96KHz sample rates. This is the level of audio used for the 5.1 surround sound channels on DVD movies. For those looking for the best audio definition, the new 24-bit 192KHz solutions are better.

Signal-to-Noise Ratio

Another aspect of audio components that users will come across is a Signal-to-Noise Ratio (SNR). This is a number represented by decibels (dB) to describe the ratio of an audio signal compared to the noise levels generated by the audio component. The higher the Signal-To-Noise ratio, the better the sound quality is. The average person generally cannot distinguish this noise if the SNR is greater than 90dB.

Standards

When reading the audio specifications, called AC’97 Compatible audio often comes up. This is a specification developed by Intel for audio processors integrated on computer motherboards. All AC’97 audio solutions support the 16-bit 96KHz, 6-channel playback for DVD surround sound recordings. This specification does not guarantee quality levels, so solutions may have various Signal-to-Noise Ratios. The specification also does not detail what surround sound modes are supported.

Another standard that might be referred to is 16-bit Sound Blaster compatible. Sound Blaster is a brand of audio cards created by Creative Labs. The Sound Blaster 16 was one of the first major sound cards to support the 16-bit 44KHz sampling rate for CD-Audio quality computer audio. This standard is below that of the newer AC’97 standard and will only be found on older computers. For those only worried about CD quality audio and not intending to use surround sound or DVD movies, this may be sufficient for them.

EAX or Environmental Audio Extensions is another standard that was developed by Creative Labs. Instead of a specific format for audio, it is a set of software extensions that modify audio to replicate the effects of specific environments. For example, the audio being played in a computer could be designed to sound as if it was being played in a cave with lots of echo. Support for this can exist in either software or hardware. If rendered in hardware, it uses fewer cycles from the CPU. There are currently three versions: EAX, EAX2 and EAX3.

Finally, some products may carry the THX logo. This is essentially a certification that THX Laboratories feels that the product meets or exceeds their minimum specifications. Just remember that a THX certified product will not necessarily have better performance or sound quality than one that does not. The manufacturers have to pay THX labs for the certification process.

Surround Sound Basics

The earliest form of surround sound came when audio was moved from a single channel source to stereophonic recordings. Later surround sound standards were developed by the movie industry to allow for a more immersive experience. Of course, the level of immersion by the listener is very dependent upon the number of speakers used and the number of channels in the surround sound encoding.

The most basic of surround sound configurations consists of at least four speakers. Two speakers are in front of the user and two behind. This gives the spatial appearance of audio in front of and behind the listener. Most modern systems also use a fifth speaker directly in front of the listener. This is typically used for dialog during movies to give the listener the sense that the dialogue is coming directly from the screen. Another common element in surround sound is a subwoofer. This is a large speaker that only puts out low frequency or bass audio that is non-directional. This also can give the loud booming effect.

The speaker layouts are denoted in a very specific manner when referring to surround sound. Typically it will be in the form of a dotted notation such as 5.1. The first number represents the number of distinct full range audio channels or speakers. The number after the dot represents the presence of a low frequency effect (LFE) or subwoofer channel.

Surround Standards

There are a variety of surround standards that have been developed over the years by the movie industry and are supported by computer audio processors for delivering surround sound. For those interested in surround sound with their computer system, the support for these different standards may be very important.

Dolby Pro Logic is a simple form of surround sound. This is actually a two-channel source that has encoding for certain frequencies to be played in a set of rear speakers. This allows the Dolby Pro Logic stream to be compatible with two speaker systems or to utilize a second set of rear speakers for a more immersive experience. A newer version called Dolby Pro Logic II was developed recently, but it is not very common.

AC-3 or Dolby Digital was developed by Dolby Labs as one of the first truly digital forms of surround sound for movies. It features 5 discrete channels (2 front, 2 rear, 1 center) plus a low-frequency effects channel. The primary benefit of this format is that it uses compression to take all this data and fit it in a very compact format with very little loss in quality. In order to playback sources with Dolby Digital (such as DVD movies), the computer audio processor will need to support this as well as have 5 speakers plus subwoofer. Audio processors require at least 16-bit 96KHz processing ability for true Dolby Digital.

DTS or Digital Theater Systems was a competing standard to Dolby Digital for 5.1 surround sound for movies. It also has five distinct speakers (2 front, 2 read, 1 center) plus a subwoofer. It uses a different encoding scheme than Dolby Digital that provides less loss in audio quality but at a larger audio stream. A newer standard called DTS-ES was recently released that supports a 6.1 speaker configurations and is backward compatible with the DTS 5.1 equipment.

Direct3D is an audio standard that is used by the Microsoft DirectX engines for generating 3D positional sound from the computer system. This is primary used in computer games to simulate the 3D environments that players are traveling through. This is not necessarily an encoding format more than a set of software standards that allows for the hardware manufacturers. It can support anything from two speaker configurations all the way up to 7.1 and 8.1 speaker configurations as long as the software programs will render the audio to that number of speakers.

Do You Need Surround?

For the majority of consumers, the answer to this question is no. Unless the computer is going to be used to watch movies or play 3D video games, the extra cost is not justified. Also, the environment that many computers exist in is not very conducive to listening to music or movies. The fans used for cooling the components of the computer will generally cause background noise that disrupts the listening experience.

For those who do plan on watching movies in surround sound or playing video games, it is generally best to make sure that the sound solutions you purchase fully support the Dolby Digital, DTS and Direct3D standards. This will ensure that whatever media you are likely to playback on your computer will work properly for surround sound.

Interconnects

Mini-Jacks

This is the most common form of interconnect between a computer system and speakers or stereo equipment and are the same connectors used on portable headphones. The reason that these are used so frequently is the size. It is possible to place upwards of 6 mini-jacks on a single PC card slot cover.

In addition to its size, mini-jacks are widely used for audio components. Portable audio has using these for many years making a wide range of headphones, external mini-speakers and amplified speakers compatible with the computer. With a simple cable, it is also possible to convert a mini-jack plug into RCA outputs for home stereo equipment.

Mini-jacks lack dynamic range though. Each mini-jack can only carry the signal for two channels or speakers. This means in the 5.1 surround setup, three mini-jack cables are required to carry the signal for the 6 channels of audio. Most audio solutions can do this without a problem, but sacrifice the audio-in and microphone jacks for output.

RCA Connectors

The RCA connector has been the standard for home stereo interconnects for a very, very long time. Each individual plug carries the signal for a single channel. This means a stereo output requires a cable with two RCA connectors. Since they have been in use for so long, there has also been a lot of development in the quality of cabling.

Of course, most computer system will not feature RCA connectors. The size of the connector is very large and the limited space of the PC card slot prevents many from being used. A 5.1 surround sound configuration would require 6 connectors. Since most computers are not hooked up to home stereo systems, the manufacturers generally opt to use the mini-jack connectors instead.

Digital Coax

With the advent of digital media such as CD and DVD, there was a need for preserving the digital signal. Constant switching between audio and digital signals induces noise into the sound. As a result, new digital interfaces were created for PCM (Pulse Code Modulation) signals from CD players to the AC-3 and DTS connections on the DVD players. Digital coax is one of the two methods for carrying the digital signal.

Digital coax looks identical to that of a RCA connector but it has a very different signal carried over it. With the digital signal traveling across the cable, it is able to pack a complete 5.1 signal into a single digital stream across the cable that would require six individual analog RCA connectors. This makes digital coax very efficient.

Of course, the drawback to using a digital coax connector is that the equipment that the computer hooks into must also be compatible. Typically, it requires either an amplified speaker system with digital decoders built into them or a home theatre receiver with the decoders. Since the digital coax can also carry different encoded streams, the device must be able to auto detect the type of signal. This can drive up the price of the connecting equipment.

SPDIF or Digital Optical

As good as digital coax is there are still some inherent problems. Digital coax is still limited to the problems of an electrical signal. They are affected by the materials they travel through and the electrical fields that are surrounded by. To combat these effects, an optical connector or SPDIF (Sony/Philips Digital Interface) was developed. This transmits the digital signal across a fiber optic cable to retain the signal integrity.

SPDIF connectors provide the cleanest form of signal transfer currently available, but there are limitations. First, it requires very specialized fiber optic cables that are very expensive. Second, the receiving equipment must also have the ability to receive the SPDIF connector. This is generally found on the home theater receivers, but it is very uncommon for amplified computer speaker sets.

Speakers

Notebook Speakers

Before going any further, its time to address notebooks. Currently no notebook system has a speaker solution that will provide a very high quality audio reproduction. Most solutions will sound very tinny and high pitched with very little low frequency. In addition, there tends to be a lot of noise transmitted over the speakers due to the limited space and close proximity to all of the internal components.

So, what can one do about notebook audio to improve things? The simplest solution is to get a good pair of portable audio headphones and plug them into the notebook. Another solution is to hook the notebook to either amplified external speakers or use new USB based sound solutions. These allow for more dynamic ranges to be produced then the internal solutions.

Amplification

Amplification is a tricky subject for consumers. Most systems are listed with their total wattage for all the speakers. The problem is one higher wattage system can end up sounding terrible when compared to one with half the rated wattage. This has to due with the current, type of amplification, size of speakers and even the speaker cable. Don’t be trapped into only looking at wattage.

The best thing to do to test the speakers is to listen to a set in a store and adjust the volume from low levels to much higher levels. You will likely annoy others in the store, but you want to see how the sound is reproduced at higher volume or amplification. Good speakers should sound clear through all volume levels. Poor speakers will tend to pop or have distortion especially on the lower frequencies.

Wires, Wires Everywhere

As the number of speakers in a system increases from the simple two speakers to the 5.1 surround configurations, the number of wires proportionally increases. When looking at computer speakers, it is important to look at what wires they use. Typical speakers will include interconnects that run from the computer to the subwoofer and amplifier. From here, all of the satellite speakers will connect into the subwoofer for the power.

Most inexpensive speakers will come with some sort of fixed wiring solution. These will be predetermined lengths of cable with a fixed connector such as a mini-jack or RCA connector that plugs into the subwoofer or amplifier. The advantage to this is that it helps reduce the costs by providing a standard system for the manufacturer. The user also has a very simple time connecting them as no wiring striping is involved. The problem is that the speaker wires have a fixed length that will restrict where the speakers can be located in relation to the computer and amplifier. This can be very problematic if rear speakers must be routed a certain way to hide wires and the cable lengths are not long enough.

More expensive speaker systems may come with standard home stereo speaker wiring. The advantage to this is that the speakers can be placed in any location as long as you have enough wire. The speakers tend to come with a thin gauge of wire that will provide adequate levels of sound, but for better quality the user can replace it with different specialized wire. Expect to have to spend more time in setting up these types of speakers.

Mounting

The majority of computer speaker systems are designed to rest on the desk that the computer sits on. The stands will typically be manufactured from plaster and have a fixed angle for the speaker. Some systems will come with adjustable stands that allows for better directional placement of the speakers. This can be important if the speakers need to reside above the listener and need to be angled down.

Wall mounting is an option for some high-end speakers, but be careful. Make sure that the speakers use a standard screw mount system so that adjustable mounts can be used to place them in locations with proper directional adjustments. Many systems will come with fixed wall mounts that allow for them to only point in a single direction.

Conclusions

Most computer systems come with a fairly generic set of amplified speakers that can leave room for much improvement. If the computer is not going to be used for music, movies or even gaming than these speakers will probably be sufficient, but for the other tasks third party speakers will improve the audio quality. When looking at the speakers, just be aware of what is included in the package and make sure those speakers fit with the way you intend to use the computer for audio and how those speakers will be placed.

source: http://compreviews.about.com/cs/soundcards/a/CompAudioPt1.htm