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<H2><A NAME="SECTION002410000000000000000">
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4.1 PCM Terminology and Concepts</A>
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In order to use PCM devices it is useful to be familiar with some concepts and
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<DT><STRONG>Sample</STRONG></DT>
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<DD>PCM audio, whether it is input or output, consists at the lowest level
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of a number of single samples. A sample represents the sound in a single channel in
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a brief interval. If more than one channel is in use, more than one sample is required
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for each interval to describe the sound. Samples can be of many different sizes, ranging
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from 8 bit to 64 bit presition. The specific format of each sample can also vary - they
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can be big endian byte order, little endian byte order, or even floats.
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<DT><STRONG>Frame</STRONG></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 stereo, each frame
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consists of two samples, etc.
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<DT><STRONG>Frame size</STRONG></DT>
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<DD>This is the size in bytes of each frame. This can vary a lot: if each sample is
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8 bits, and we're handling mono sound, the frame size is one byte. Similarly in 6 channel audio with
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64 bit floating point samples, the frame size is 48 bytes
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<DT><STRONG>Rate</STRONG></DT>
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<DD>PCM sound consists of a flow of sound frames. The sound rate controls how often
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the current frame is replaced. For example, a rate of 8000 Hz means that a new frame is played
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or captured 8000 times per second.
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<DT><STRONG>Data rate</STRONG></DT>
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<DD>This is the number of bytes, which must be recorded or provided per second
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at a certain frame size and rate.
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8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s
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At the other end of the scale, 96000 Hz, 6 channel sound with 64 bit (8 bytes) samples
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has a data rate of 96000 * 6 * 8 = 4608 kb/s (almost 5 Mb sound data per second)
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<DT><STRONG>Period</STRONG></DT>
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<DD>When the hardware processes data this is done in chunks of frames. The time interval
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between each processing (A/D or D/A conversion) is known as the period. The size of the period has
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direct implication on the latency of the sound input or output. For low-latency the period size should
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be very small, while low CPU resource usage would usually demand larger period sizes. With ALSA, the
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CPU utilization is not impacted 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 memory copy, but userspace can be
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slower and read or write multiple periods at the same time.
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<DT><STRONG>Period size</STRONG></DT>
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<DD>This is the size of each period in Hz. <em>Not bytes, but Hz!.</em> In <tt class="module">alsaaudio</tt>
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the period size is set directly, and it is therefore important to understand the significance of this
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number. If the period size is configured to for example 32, each write should contain exactly 32 frames
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of sound data, and each read will return either 32 frames of data or nothing at all.
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Once you understand these concepts, you will be ready to actually utilize PCM API. Read on.
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