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larsimmisch df89c12581 Decided that getenum should return the selected item and the available
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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>
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<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>
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</dl>
<p>
Once you understand these concepts, you will be ready to use the PCM
API. Read on.
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