From d91e0649c94f74451b88beb107adc0fa83f6be05 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Feb 2002 23:19:29 +0000 Subject: [PATCH] Changed PCM intro --- src/pcm/pcm.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 8132ec59..e7d85a26 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -42,7 +42,32 @@ understanding it as general digital audio processing with volume samples generated in continuous time periods.

-

Write some description about digital audio here.

+

The analog signal is recorded via analog to digital converters (ADC). +The digital value (de-facto a volume at a specific time) obtained +from ADC can be further processed. The following picture shows a perfect +sinus waveform:

+ +
+\image html wave1.gif + +

Next image shows digitized representation:

+ +
+\image html wave2.gif + +

As you may see, the quality of digital audio signal depends on the time +(recording rate) and voltage resolution (usually in an linear integer +representation with basic unit one bit).

+ +

The stored digital signal can be converted back to voltage (analog) +representation via digital to analog converters (DAC).

+ +

One digital value is called sample. More samples are collected to frames +(frame is terminology for ALSA) depending on count of converters used at one +specific time. One frame might contain one sample (when only one converter is +used - mono) or more samples (for example: stereo has signals from two converters +recorded at same time). Digital audio stream contains collection of frames +recorded at boundaries of continous time periods.

\section pcm_general_overview General overview -- 2.47.1