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Aedan 9th December, 2002 04:14 PM

Soundcard FAQ
The soundcard FAQ is divided into several sections, each as a seperate post, dealing with a seperate area. Whilst the FAQ hopes to answer questions you may have, it is entirely possible that you have a question that has not been answered here! If you cannot find the answer in this FAQ, then feel free to post a request here.

Aedan 9th December, 2002 04:14 PM

General questions

What is a soundcard?
A soundcard is typically a plug in daughterboard that provides various chips that can convert digital information into a signal that speakers and headphone can reproduce.

What's the difference between onboard sound and a soundcard?
In an ideal world, there would be no difference. Alas, we do not live in an ideal world! Often onboard sound has been added with little regard to anything but cost. This often means that it performs poorly. Poorly could mean muffled sound, poor bass, a high level of noise or a blatent inability to do what it's meant to do. Often onboard sound has less in the way of features compared to a soundcard.

Some soundcards also suffer from the same problems as onboard sound, but these are generally the very cheap generic parts. Generally, the more expensive a soundcard is, the better it performs.

Does the use of a soundcard make any difference?
Depending on what you wish to use your soundcard for, you may find a different soundcard is more suitable. Often soundcards designed for gaming do not reproduce sound as accurately or with the same level of quality as a soundcard designed for sound recording. A soundcard designed for sound recording will often not have the same type of acceleration required for gaming, despite generally being of higher quality. Decide what you want your soundcard to do before you buy it!

What is on a soundcard?
A soundcard is typically made up of several parts. However, generally there are two main parts, a Digital Controller and a CODEC. The Digital Controller is the interface to the computer, and the CODEC is the interface to the outside world.

What is a CODEC?
The abbreviation CODEC stands for COder/DECoder. The CODECs job in life is to convert one form of data into another form. It's also a generic term, so it is applicable to many things. In terms of sound, the following are the most common uses.

In terms of physical hardware on a soundcard, a CODEC is the device that turns the digital signal into an analog signal your speakers can handle. It can also convert an analog signal into a digital one, so the computer can deal with things like a microphone or a signal on the linein jack.

In terms of software, a CODEC is a piece of code which converts one form of data into another. For example, an MP3 player has a CODEC in, which converts the MP3 data into audio data the soundcard can play. Some CODECs can convert both ways - for instance, some MP3 CODECS can convert MP3 data into audio, and audio into MP3. If you do not have the correct CODEC for the sound you are trying to play, you cannot play it!

What is a Digital Controller?
The Digital Controller deals with all the computery stuff. It handles talking to the rest of the computer, and taking orders from the CPU. It is responsible for getting the information for the sound from the rest of the computer, doing whatever is necessary, and then passing it onto the CODEC.

As the Digital Controller is the heart of the soundcard, the features it supports will affect the features of the soundcard. Features like audio acceleration, DirectSound hardware support, wavetable synthesis and more are all handled by the Digital Controller.

What is AC97?
AC97 is a standard for linking a CODEC chip to an Digital Controller chip using just 5 wires. It is to CODEC chips what IDE is to hard disks! It defines how the Digital Controller Chip processor will communicate with the CODEC chip, down to how many bits, what frequency, how to set the volume and lots of other things. This means that any AC97 CODEC can be used with any AC97 capable audio processor.

At the time of writing, the latest AC97 spec (V2.3) supported the following list of features:
  • Up to 4 analog line level stereo inputs
  • Up to 2 analog line level mono inputs
  • Mic input with 20dB boost and programmable gain
  • Dedicated Stereo out (Line out), with support for 4 or 6 channels.
  • Additional Stereo out (Aux out), configurable as line out or headphone out.
  • Mono output for speakerphone or mono speaker.
  • 16 bit minimum, with support for 18 or 20bit converters
  • Optional tone and loudness
  • Optional 3D stereo enhancement
  • Optional input for dedicated voice input
  • Option S/PDIF for digital output
  • Optional Jack sensing
  • Support for 8Khz, 11.025Khz, 16Khz, 22.05Khz, 32Khz, 44.1Khz and 48KHz sampling rates.
  • Better than 85dB of dynamic range
  • Better than 17.64kHz frequency response
  • Less than -70dB THD and Noise

On the other hand, it does not specify how the AC97 Digital Controller will appear to the computer. This means that although the AC97 CODEC and AC97 link are standard features, each manufacturer can do what they like with the Digital Controller side. Hence, although manufactuers may use the same CODEC chip, the Digital Controller will provide the addition features such as hardware acceleration.

What is a mixer?
The mixer on a soundcard is the device that collects the various sounds together and mixes them into a signal to send to the speakers. It also acts as a selector for the recording system, choosing which sound source will be recorded.

Aedan 9th December, 2002 04:15 PM

Connection Questions

What is analog/lineout?
The analog output(s) are sometimes known as Line Out. This connection is a standard voltage level that allows the soundcard to be connected directly to the inputs on an stereo amplifier or something similar.

What is digital out?
Digital output is often in the form of a S/PDIF or Coax connection. This requires connecting to a seperate (or outboard) DAC in order to turn the digital signal back into an analog signal that the speakers/amplifier can understand. Often using an outboard DAC will improve the sound quality, as the converters no longer have to operate inside the electrically noisy environment within the PC.

What is S/PDIF?
S/PDIF stands for Sony/Philips Digital InterFace. It is a consumer standard designed to allow audio to be passed in digital form to other equipment. It supports several different formats, 2 channel uncompressed audio (such as CD audio), 5.1 channel Dolby Digital, 5.1 channel DTS, 6.1 Dobly Digital EX.

At the minimum S/PDIF will provide 2 channels of audio to another device. Better soundcards can also provide Dolby Digital and DTS from DVDs. The best soundcards can mix the surround audio into a Dolby Digital stream for a surround decoder to decode. If you want to plug your Surround Decoder on your cinema system into your PC, this is the best way to do it.

S/PDIF is generally provided as either an optical port or a RCA style connector.

What are Dolby Digital and DTS?
Both Dolby Digital and DTS are methods of providing digital surround sound. They compress up to 7 channels of audio into a single digital stream, which can be sent to another piece of equipment over a single cable or fibre optic.

Aedan 9th December, 2002 04:15 PM

Audio Quality questions

What are THD+N and THD?
THD stands for "Total Harmonic Distortion". It's a method for measuring how much distortion happens to an audio signal when it travels through a piece of equipment. The lower the percentage, the less distortion is occuring, and the cleaner the sound is.

THD+N is "Total Harmonic Distortion plus Noise". It's the same THD as above, but includes any noise generated by the equipment.

What is frequency response?
Frequency Response is a measure of how well a device can reproduce different frequencies of sound at the same level of loudness. Typically it states the lowest and highest frequencies (like 20Hz to 20kHz) and the "flatness" of the response (-3dB). Any soundcard that can really manage 20Hz to 20kHz with a flatness of -3dB is very high quality and probably supasses your ears! When looking at the flatness, ensure that the manufacturer is measuring to -3dB. If a larger figure is used, then the manufacturer is trying to make the frequency response look bigger than it really is.

What is Dynamic Range?
Dynamic range is a measure of the difference between the quietest (the noise floor) and loudest noise that can be played. The bigger the difference, the more capable the devices is of dealing with subtle sounds. A CD provides a Dynamic Range of about 90dB, a prerecorded cassette tape can manage about 50dB.

What is the noise floor?
The noise floor is the level of noise (or hiss) that the soundcard generate when it is not playing any sound. This represents the quietest sound that the soundcard can manage. The measurement is usually expressed as a negative number of decibels. The more negative the number, the less noise the soundcard introduces. (Look for a number better than -70dB)

How can I tell how good a soundcard is
The acid test for a soundcard is your ears. However, the specs can often provide an idea of how well the soundcard can manage sonic purity. The lower the THD and Noise figures, the better the card can perform in terms of keeping the signal pure. The flatter the frequency reponse, and the more true the sound will be. The bigger the dynamic range is, the more detail the soundcard can play.

Aedan 9th December, 2002 04:16 PM

Misc Questions

What is the diference between Analog and Digital systems?
Digital systems work with discrete finite numbers. Analog systems work with continously varying signals. Converting between the two is never completely accurate.

What is sample rate?
The sample rate is the frequency with which the analog signal is examined and converted into a number. The more often the signal is examined, the closer we can get to tracking how quickly it changes. The sample rate is expressed in kilohertz, or thousands of samples per second. However, the highest frequency that can be digitised is half that of the sample rate!

Why is the number of bits important?
Each bit that is used in a sample contributes to the overall accuracy of the sound. With only 1 bit, there are only two levels we can generate - sound and no sound. With 2 bits, we can generate 4 levels of sound. With 3 bits, we can generate 8 levels of sound. At 16bits, we can generate 65536 levels. At 20bits, we can generate 1048576 different levels!

What is Environmental Audio?
Environmental Audio is a technique for making a sound fit the environment it is supposedly occuring in. Imagine a sound like water dripping into a puddle. If you were inside a house, it would make one sort of dripping noise. If you were in a cave, it would make a different noise. Environmental Audio provides all the cues for the setting of the sound.

What is EAX?
EAX is a Creative Labs standard for Environmental Audio. EAX can apply effects like reverb and filtering in order to make the sound more realistic for the environment it is supposedly occuring in. For gaming, this makes the sound far more realistic than playing a standard sound.

What is DirectSound?
DirectSound is a part of Microsoft's DirectX. It provides methods for getting sound to a soundcard quickly, and also provides some environmental audio features[/b]

Aedan 9th December, 2002 04:16 PM

Encoding systems

What is an encoding system?
In terms of audio, an encoding system is a method for either ensuring the audio survives transmission, or a method for reducing the space the audio takes. On occasions, an encoding system can do both!

What types of compression are there?
There are two main types of compression that can be used. One form is lossless compression, and the other form is lossy compression.

Lossless compression manages to keep all the information that was originally recorded, but squeezes it into a smaller space. When at the other end, the squeezed version is expanded back into it's original form, and is exactly the same as the original recording.

Lossy compression uses various techniques to analyse the signal, and work out which bits you won't notice go missing. This throwing away of bits helps keep the information small, and can be used to help a lossless compression to it's job better. The most popular lossy compression techniques use a model of the human hearing to work out which parts can be thrown out. Lossy compression is often based on psychoacoustic compression.

Lossy Compression Techniques

Temporal masking
Temporal masking is an effect of the way that humans hear. If you play a single note at one frequency, it takes time before the ear can hear a another quieter note at a close frequency. This effect is the temporal masking.

Frequency masking
Frequency masking is an effect of the way humans hear. If you play two notes that are close, and one is louder than the other, the ear may only hear the louder of the two notes. This is frequency masking.

The Discrete Cosine Transform is a method of converting temporal information into the frequency domain. A spectrum analyser is a device that can use a DCT to convert the sound over time into a set of frequency bands.

Sony's Adaptive TRansform Acoustic Coding was designed for use with their MiniDisc players. As the MiniDisc only held about 80Mb of data, Sony needed some way of compressing 650Mb of data into that 80Mb space. ATRAC was the result of Sony's work.

Officially known as MPEG1 layer 1. Layer 1 uses a basic DCT filter with one frame, and only uses frequency masking.

Officially known as MPEG1 layer 2, layer 2 builds on the foundations of layer 1. Layer 2 uses three frames, and starts to model temporal masking.

Officially known as MPEG1 Layer 3, layer 3 builds on layer 2. Layer 3 improves the filtering, includes temporal masking, can take stereo redundancy into account, and uses a Huffman coder.

The WMA format is a Microsoft propriatory format. It includes digital rights management, and is probably based around similar techniques to MP3.

Real Audio
The RealAudio format is a Real Networks propriatory format. It includes digital rights management, and is also probably based around similar techniques to MP3.

Aedan 9th December, 2002 04:17 PM


The format that everyone knows and loves! The CD has been around for many years now, developed as a replacement for the Compact Cassette. A CD uses no compression at all! The CD stores audio in stereo at 16bits, sampled at 44.1kHz. This gives it a bit rate of around 1.76MBits/second.

Sony's MiniDisc format is based around a re-writeable optical disc. This disc can store up to 80MBytes, and uses Sony's ATRAC compression to reduce the storage requirements of audio.

DVD Audio
Unlike CD, DVD Audio supports multiple rates. In 2 channel stereo, DVD Audio can run up to 196kHz sampling rates in 24bits. In multichannel, DVD Audio can run up to 96kHz at 24bits, but is using Dolby Digital or DTS to provide up to 6 channels. This gives DVD Audio a data rate of around 9.6MBit/second. Unfortunately, DVD Audio uses so much bandwidth that a standard S/PDIF connection cannot handle the data. In this instance, FireWire can be used to move the digital audio from one device to another. DVD Audio promises better sound than CD, but currently has limited support.

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