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Old 3rd December, 2002, 12:50 PM
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Beginnings of Sound FAQ

Ok, I've put together a few things in the vague direction of a sound FAQ/glossery.

What do people think? What's missing? What would you like to see added?

Áedán
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Old 3rd December, 2002, 12:59 PM
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Computer Specific Questions

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, with support for 4 or 6 channels.
  • Mono output for speakerphone or mono speaker.
    16 bit DAC and ADC minimum, with support for 18 or 20bit DAC and ADCs
  • 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 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 peice of equipment over a single cable or fibre optic.


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!

What is Dynamic Range?
Dynamic range is a measure of the difference between the quietest 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.


Misc Questions

What is the diference between Analog and Digital?
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]
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Last edited by Áedán; 3rd December, 2002 at 05:59 PM.
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Old 3rd December, 2002, 04:06 PM
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EXCELLENT!!!

Perhaps a small mention of digital mediums? Like CDs, DVDs, and DVD-As? Their limits and such... Encoding schemes, like MP3, MP2, MP4, WMA, ASF...
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Old 3rd December, 2002, 05:39 PM
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Perhaps a few links to sites like Winamp??
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Old 3rd December, 2002, 05:44 PM
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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.

DCT
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.

ATRAC
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.

MP1
Officially known as MPEG1 layer 1. Layer 1 uses a basic DCT filter with one frame, and only uses frequency masking.

MP2
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.

MP3
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.

WMA
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.

Media

CD
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.

MiniDisc
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.
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Last edited by Áedán; 3rd December, 2002 at 05:48 PM.
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Old 3rd December, 2002, 05:52 PM
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Perhaps some mention of the sound available on newer motherboards, 5.1 etc, in comparison to that of add in cards would be useful??
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Old 3rd December, 2002, 06:01 PM
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Quote:
Originally posted by Daniel ~
Perhaps some mention of the sound available on newer motherboards, 5.1 etc, in comparison to that of add in cards would be useful??
Not sure exactly what you're looking for? Built in sound vs add in sound? Done right they should be equal!

Áedán
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Old 3rd December, 2002, 06:08 PM
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Well then say so damn it! LOL

A new guy, or someone mentally disfigured, such as myself, would find that useful information. ":O}

Superb work BTW.
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Old 3rd December, 2002, 06:11 PM
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Quote:
Originally posted by Daniel ~
Perhaps some mention of the sound available on newer motherboards, 5.1 etc, in comparison to that of add in cards would be useful??
As of now, I believe there are 5 available sound systems available as onboard audio for motherboards. AC97 (explained already above), ALC650, C-Media 8738, nForce APU, and the new nForce2 Soundstorm. I guess there might be others, but I only have boards with these on them. I don't really know how they compare to soundcards although I hear the C-Media and nForce tends to be at least the equivalents of Live!5.1.
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Old 3rd December, 2002, 07:47 PM
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Quote:
Originally posted by mookydooky


As of now, I believe there are 5 available sound systems available as onboard audio for motherboards. AC97 (explained already above), ALC650, C-Media 8738, nForce APU, and the new nForce2 Soundstorm. I guess there might be others, but I only have boards with these on them. I don't really know how they compare to soundcards although I hear the C-Media and nForce tends to be at least the equivalents of Live!5.1.
The ALC650 is just an AC97 CODEC, so it fits firmly in the AC97 category. Most of the nForce and nForce2 configs also fit firmly in the AC97 category too. They're just the Digital Controller side of the AC97 link!

I'll work out how to put stuff in there - This sound FAQ started because I was tired of people slagging off AC97 without realising that it's just a bus, not a sound implementation!

Áedán
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Old 4th December, 2002, 01:45 AM
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not to thread crap~ but nice name...... dint realize it was you :P
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Old 5th October, 2004, 02:02 AM
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Why are the support sampleing rates is 8Khz, 11.025Khz, 16Khz, 22.05Khz, 32Khz, 44.1Khz and 48KHz, not 20K/40K or else?
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Old 5th October, 2004, 08:08 AM
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The short answer is: History.

There are actually two sequences there:
8, 16, 32, 48, 96, 192
11.025, 22.05, 44.1

CD runs at 44.1kHz, as you probably already know. It seems a strange figure. It is certainly above the important 40kHz sampling rate needed to fulfill Nyquist. But why 44.1 exactly? Well, in the 1970s, when digital audio was starting to get off the ground, hard disks were expensive and low capacity. Expensive and low capacity means >$400 for 5Mb.

Another storage solution was needed, so video recorders were modified to handle digital audio signals. This was done by converting the digital signal to something that looked like a video signal, but actually contained the digital audio, encoded as black and white 'pixels'. As video recorders were in use, the sample rate had to relate to the TV standards, and survive either PAL or NTSC. This required there to be an integer number of samples per line. Using three samples per line, you get the first figure over the magic 40KHz, and that first figure just happens to be 44.1kHz.

For NTSC, there are 245 active lines at 60 fields per second. By fitting 3 samples (96 bits for stereo) on a line, you end up with 60 * 245 *3 which gives 44,100 samples a second.
For PAL, there are 294 active lines at 50 fields per second. By fitting 3 samples (96 bits for stereo) on a line you end up with 50 * 294 * 3, which gives 44,100 samples a second.

As to why use a 48kHz sampling rate - I'm not sure. This may have been an internal decision when R-DAT and S-DAT were in the lab, but I don't know.
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Old 5th October, 2004, 01:16 PM
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looking good but you don't mention anything about midi
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Old 5th October, 2004, 01:59 PM
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No, that's rather deliberate on my part. MIDI, in the hardware sense, is rather a slow archaic interface that's not really capable of keeping up with today's requirements.

Or did you mean something else?
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Old 5th October, 2004, 02:06 PM
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!!!! midi is still a massively used and essential tool when it comes to music creation obviously games don't use it any more but any one into audio editing and music creation will do
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Old 5th October, 2004, 02:14 PM
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The biggest problem with MIDI is it's latency. When it comes to playing a number of notes at the same instant, MIDI just can't handle it, and you end up being able to hear that some notes are delayed.
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Old 5th October, 2004, 02:20 PM
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maybe on cheap hardware or with computers (which can still get it down to <10ms) but a lot of artists still swear by it, I've never been in a studio without a massive selection of midi equipment, the mixing desk used for my recordings was controlled by an ST because they have such good timing
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Old 9th November, 2004, 02:16 AM
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why not the frequency bandwidth has low frequency, EXP: 250Hz-3400Hz, why not keep 0-250Hz signal, just to eliminate low frequency noise?
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Old 15th November, 2004, 03:10 PM
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Quote:
Originally Posted by Áedán
Computer Specific Questions

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!

[/b]
i thought codecs were intimidatory chips IE a converter, as a example, the DEC takes digital data source and converts it into a usable data stream for a DAC

DAC's and ADC's are the converters for digital streams and analouge signals
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