From 8d95af2cb21e2ae9ceea247d35a97461a4a26562 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Feb 2002 22:11:14 +0000 Subject: [PATCH] Removed digital audio description (borrowed from OSS drivers) --- src/pcm/pcm.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 0cb1d751..8132ec59 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -42,51 +42,7 @@ understanding it as general digital audio processing with volume samples generated in continuous time periods.

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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).

- -

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).

- -

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.

- -

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.

- -

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.

+

Write some description about digital audio here.

\section pcm_general_overview General overview -- 2.47.1