PCM Terminology and Concepts¶
In order to use PCM devices it is useful to be familiar with some concepts and terminology.
PCM audio, whether it is input or output, consists of samples. A single sample represents the amplitude of one channel of sound at a certain point in time. A lot of individual samples are necessary to represent actual sound; for CD audio, 44100 samples are taken every second.
Samples can be of many different sizes, ranging from 8 bit to 64 bit precision. The specific format of each sample can also vary - they can be big endian byte integers, little endian byte integers, or floating point numbers.
Musically, the sample size determines the dynamic range. The dynamic range is the difference between the quietest and the loudest signal that can be reproduced.
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.
- Frame size
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. For six channel audio with 64 bit floating point samples, the frame size is 48 bytes.
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.
- Data rate
This is the number of bytes which must be consumed or provided per second at a certain frame size and rate.
8000 Hz mono sound with 8 bit (1 byte) samples has a data rate of 8000 * 1 * 1 = 8 kb/s or 64kbit/s. This is typically used for telephony.
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).
If the data rate requirement is not met, an overrun (on capture) or underrun (on playback) occurs; the term “xrun” is used to refer to either event.
The CPU processes sample data in chunks of frames, so-called periods (also called fragments by some systems). The operating system kernel’s sample buffer must hold at least two periods (at any given time, one is processed by the sound hardware, and one by the CPU).
The completion of a period triggers a CPU interrupt, which causes processing and context switching overhead. Therefore, a smaller period size causes higher CPU resource usage at a given data rate.
A bigger size of the buffer improves the system’s resilience to xruns. The buffer being split into a bigger number of smaller periods also does that, as it allows it to be drained / topped up sooner.
On the other hand, a bigger size of the buffer also increases the playback latency, that is, the time it takes for a frame from being sent out by the application to being actually audible.
Similarly, a bigger period size increases the capture latency.
The trade-off between latency, xrun resilience, and resource usage must be made depending on the application.
- Period size
This is the size of each period in frames. Not bytes, but frames! In
alsaaudiothe 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.
Once you understand these concepts, you will be ready to use the PCM API. Read on.