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<H2><A NAME="SECTION002410000000000000000">
4.1 PCM Terminology and Concepts</A>
</H2>

<P>
In order to use PCM devices it is useful to be familiar with some concepts and
terminology.

<P>
<DL>
<DT><STRONG>Sample</STRONG></DT>
<DD>PCM audio, whether it is input or output, consists at the lowest level
of a number of single samples. A sample represents the sound in a single channel in
a brief interval. If more than one channel is in use, more than one sample is required
for each interval to describe the sound. Samples can be of many different sizes, ranging
from 8 bit to 64 bit presition. The specific format of each sample can also vary - they 
can be big endian byte order, little endian byte order, or even floats.

<P>
</DD>
<DT><STRONG>Frame</STRONG></DT>
<DD>A frame consists of exactly one sample per channel. If there is only one
channel (Mono sound) a frame is simply a single sample. If the sound is stereo, each frame
consists of two samples, etc.

<P>
</DD>
<DT><STRONG>Frame size</STRONG></DT>
<DD>This is the size in bytes of each frame. This can vary a lot: if each sample is
8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with
64 bit floating point samples, the frame size is 48 bytes

<P>
</DD>
<DT><STRONG>Rate</STRONG></DT>
<DD>PCM sound consists of a flow of sound frames. The sound rate controls how often
the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played 
or captured 8000 times per second.

<P>
</DD>
<DT><STRONG>Data rate</STRONG></DT>
<DD>This is the number of bytes, which must be recorded or provided per second
at a certain frame size and rate. 

<P>
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s

<P>
At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples
has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second)

<P>
</DD>
<DT><STRONG>Period</STRONG></DT>
<DD>When the hardware processes data this is done in chunks of frames. The time interval
between each processing (A/D or D/A conversion) is known as the period. The size of the period has
direct implication on the latency of the sound input or output. For low-latency the period size should
be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the
CPU utilization is not impacted much by the period size, since the kernel layer buffers multiple
periods internally, so each period generates an interrupt and a memory copy, but userspace can be
slower and read or write multiple periods at the same time.

<P>
</DD>
<DT><STRONG>Period size</STRONG></DT>
<DD>This is the size of each period in Hz. <em>Not bytes, but Hz!.</em> In <tt class="module">alsaaudio</tt>
the period size is set directly, and it is therefore important to understand the significance of this 
number. If the period size is configured to for example 32, each write should contain exactly 32 frames
of sound data, and each read will return either 32 frames of data or nothing at all.

<P>
</DD>
</DL>

<P>
Once you understand these concepts, you will be ready to actually utilize PCM API. Read on.

<P>

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