PCM Terminology and Concepts — alsaaudio v0.6 documentation

### Navigation

- index
- modules |
- next |
- previous |
- alsaaudio v0.6 documentation »

# PCM Terminology and Concepts

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

Sample

  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 resproduced.

Frame
  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.
      Similarly in 6 channel audio with 64 bit floating point samples, the frame size is 48 bytes
Rate
  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 recorded 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)

Period
  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.
Period size
  This is the size of each period in Hz. Not bytes, but Hz!. In `alsaaudio` 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.

Once you understand these concepts, you will be ready to use the PCM API. Read on.

### Navigation

- index
- modules |
- next |
- previous |
- alsaaudio v0.6 documentation »