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Linear pulse code modulation
From Wikipedia, the free encyclopedia
Linear pulse code modulation (LPCM) is a method of encoding audio information digitally. The term also refers collectively to formats using this method of encoding. The term PCM, though strictly more general, is often used to describe data encoded as LPCM.
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LPCM is a particular method of pulse code modulation which represents an audio waveform as a sequence of amplitude values recorded at a sequence of times. LPCM specifies that the values stored are proportional to the amplitudes, rather than representing say the logarithm of the amplitude, or being related in some other manner. In practice these values will be quantized. Theoretically, there is no loss or error in conversion and reconstruction, as long as the sampling rate is just over twice the highest desired frequency component of the recorded signal [Claude Shannon; Harry Nyquist]. For example, if you want to record audio at up to 20 kHz, you would need a frequence of sampling (F/s) of a little more than 40 kHz. Some argue that 60 kHz F/s would be better as the rolloff frequency of the low pass filter which attenuates the echoes of the recorded signal, which otherwise would beat against the original, at multiples of the sampling rate, would be ultrasonic and therefore not smear the high end of the converted (audible range of the) audio. A stable clock, for conducting the regular timing of the sync pulses, which are scaled to represent the amplitude values of the original analog signal at each sampling instance, is essential to the good functioning of the digital audio system.
LPCM is the method of encoding generally used in conjunction with the WAV container format, the de facto standard for uncompressed audio on PCs. The term PCM and LPCM often refer explicitly to the format used in WAV files, though LPCM data may also be commonly stored in other formats such as AIFF. LPCM is further used for the lossless encoding of audio data in the compact disc Red Book standard. LPCM has been defined as a part of the DVD standard. AES3 is a particular format using LPCM.
Common sample resolutions for LPCM are 8, 16, 20, 24 or 32 bits per sample.
While two channels (stereo) is the most common format, some can support up to 8 audio channels (7.1 surround).
Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in compact discs. Sampling frequencies of 96 kHz or 192 kHz can be used on some newer equipment, with the higher value equating to 6.144 megabits per second per audio channel.
Most DVD players only support 48 kHz/16-bit capability. Only more high-end players have built-in 96 kHz/24-bit capabilities. The DVD-Audio standard supports 192 kHz/24-bit playback.
- Summary of LPCM – Contains links to information about implementations and their specifications.
- How to control internal/external hardware using Microsoft's Media Control Interface – Contains information about, and specifications for the implementation of LPCM used in WAV files.
