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Removed digital audio description (borrowed from OSS drivers)

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Jaroslav Kysela 2002-02-13 22:11:14 +00:00
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src/pcm/pcm.c
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@ -42,51 +42,7 @@
understanding it as general digital audio processing with volume samples
generated in continuous time periods.</P>
<P>Digital audio is the most commonly used method of representing
sound inside a computer. In this method sound is stored as a sequence of
samples taken from the audio signal using constant time intervals.
A sample represents volume of the signal at the moment when it
was measured. In uncompressed digital audio each sample require one
or more bytes of storage. The number of bytes required depends on number
of channels (mono, stereo) and sample format (8 or 16 bits, mu-Law, etc.).
The length of this interval determines the sampling rate. Commonly used
sampling rates are between 8kHz (telephone quality) and
48kHz (DAT tapes).</P>
<P>The physical devices used in digital audio are called the
ADC (Analog to Digital Converter) and DAC (Digital to Analog Converter).
A device containing both ADC and DAC is commonly known as a codec.
The codec device used in a Sound Blaster cards is called a DSP which
is somewhat misleading since DSP also stands for Digital Signal Processor
(the SB DSP chip is very limited when compared to "true" DSP chips).</P>
<P>Sampling parameters affect the quality of sound which can be
reproduced from the recorded signal. The most fundamental parameter
is sampling rate which limits the highest frequency that can be stored.
It is well known (Nyquist's Sampling Theorem) that the highest frequency
that can be stored in a sampled signal is at most 1/2 of the sampling
frequency. For example, an 8 kHz sampling rate permits the recording of
a signal in which the highest frequency is less than 4 kHz. Higher frequency
signals must be filtered out before feeding them to ADC.</P>
<P>Sample encoding limits the dynamic range of a recorded signal
(difference between the faintest and the loudest signal that can be
recorded). In theory the maximum dynamic range of signal is number_of_bits *
6dB. This means that 8 bits sampling resolution gives dynamic range of
48dB and 16 bit resolution gives 96dB.</P>
<P>Quality has price. The number of bytes required to store an audio
sequence depends on sampling rate, number of channels and sampling
resolution. For example just 8000 bytes of memory is required to store
one second of sound using 8kHz/8 bits/mono but 48kHz/16bit/stereo takes
192 kilobytes. A 64 kbps ISDN channel is required to transfer a
8kHz/8bit/mono audio stream in real time, and about 1.5Mbps is required
for DAT quality (48kHz/16bit/stereo). On the other hand it is possible
to store just 5.46 seconds of sound in a megabyte of memory when using
48kHz/16bit/stereo sampling. With 8kHz/8bits/mono it is possible to store
131 seconds of sound using the same amount of memory. It is possible
to reduce memory and communication costs by compressing the recorded
signal but this is beyond the scope of this document. </P>
<P>Write some description about digital audio here.</P>
\section pcm_general_overview General overview