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<div class="section" id="pcm-terminology-and-concepts">
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<h1>PCM Terminology and Concepts<a class="headerlink" href="#pcm-terminology-and-concepts" title="Permalink to this headline">¶</a></h1>
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<p>In order to use PCM devices it is useful to be familiar with some concepts and
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terminology.</p>
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<dl class="docutils">
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<dt>Sample</dt>
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<dd><p class="first">PCM audio, whether it is input or output, consists of <em>samples</em>.
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A single sample represents the amplitude of one channel of sound
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at a certain point in time. A lot of individual samples are
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necessary to represent actual sound; for CD audio, 44100 samples
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are taken every second.</p>
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<p>Samples can be of many different sizes, ranging from 8 bit to 64
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bit precision. The specific format of each sample can also vary -
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they can be big endian byte integers, little endian byte integers, or
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floating point numbers.</p>
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<p class="last">Musically, the sample size determines the dynamic range. The
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dynamic range is the difference between the quietest and the
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loudest signal that can be resproduced.</p>
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</dd>
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<dt>Frame</dt>
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<dd>A frame consists of exactly one sample per channel. If there is only one
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channel (Mono sound) a frame is simply a single sample. If the sound is
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stereo, each frame consists of two samples, etc.</dd>
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<dt>Frame size</dt>
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<dd><dl class="first last docutils">
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<dt>This is the size in bytes of each frame. This can vary a lot: if each sample</dt>
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<dd><dl class="first last docutils">
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<dt>is 8 bits, and we’re handling mono sound, the frame size is one byte.</dt>
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<dd>Similarly in 6 channel audio with 64 bit floating point samples, the frame
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size is 48 bytes</dd>
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</dl>
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</dd>
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</dl>
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</dd>
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<dt>Rate</dt>
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<dd>PCM sound consists of a flow of sound frames. The sound rate controls how
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often the current frame is replaced. For example, a rate of 8000 Hz
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means that a new frame is played or captured 8000 times per second.</dd>
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<dt>Data rate</dt>
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<dd><p class="first">This is the number of bytes, which must be recorded or provided per
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second at a certain frame size and rate.</p>
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<p>8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of
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8000 * 1 * 1 = 8 kb/s or 64kbit/s. This is typically used for telephony.</p>
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<p class="last">At the other end of the scale, 96000 Hz, 6 channel sound with 64
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bit (8 bytes) samples has a data rate of 96000 * 6 * 8 = 4608
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kb/s (almost 5 Mb sound data per second)</p>
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</dd>
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<dt>Period</dt>
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<dd>When the hardware processes data this is done in chunks of frames. The time
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interval between each processing (A/D or D/A conversion) is known
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as the period.
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The size of the period has direct implication on the latency of the
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sound input or output. For low-latency the period size should be
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very small, while low CPU resource usage would usually demand
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larger period sizes. With ALSA, the CPU utilization is not impacted
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much by the period size, since the kernel layer buffers multiple
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periods internally, so each period generates an interrupt and a
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memory copy, but userspace can be slower and read or write multiple
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periods at the same time.</dd>
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<dt>Period size</dt>
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<dd>This is the size of each period in Hz. <em>Not bytes, but Hz!.</em> In
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<a class="reference internal" href="libalsaaudio.html#module-alsaaudio" title="alsaaudio (Linux)"><code class="xref py py-mod docutils literal"><span class="pre">alsaaudio</span></code></a> the period size is set directly, and it is
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therefore important to understand the significance of this
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number. If the period size is configured to for example 32,
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each write should contain exactly 32 frames of sound data, and each
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read will return either 32 frames of data or nothing at all.</dd>
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</dl>
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<p>Once you understand these concepts, you will be ready to use the PCM API. Read
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on.</p>
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