What is PCM? What does it have to do with. wav files?

PCM (pulse codemodulation----Pulse Code modulated recording). The so-called PCM recording is the sound and other analog signals into the symbolic pulse column, and then recorded. PCM signal is a digital signal composed of [1], [0] and other symbols, without any coding and compression processing. Compared with analog signal, it is not susceptible to the clutter and distortion of the transmission system. The dynamic range is wide and the effect of sound quality can be very good.

WAV is an audio format developed by Microsoft. WAV complies with PIFF Resource Interchange FileFormat specification. All WAV has a file header, which is the encoding parameter of the header audio stream. WAV does not have a hard-coded audio stream encoding, except for PCM, and almost any code that supports the ACM specification can encode WAV audio streams. Many friends do not have this concept, we take AVI to do a demonstration, because AVI and WAV in the file structure is very similar, but AVI more than a video stream. We are exposed to many kinds of AVI, so we often need to install some decode to watch some avi, we touch more DivX is a video encoding, AVI can use DivX encoding to compress the video stream, of course, can also use other encoding compression. Similarly, WAV can also use a variety of audio encoding to compress its audio stream, but we are common is the audio stream is PCM encoding processing of WAV, but this does not mean that WAV can only use PCM encoding, MP3 encoding can also be used in WAV, and AVI, as long as the corresponding decode installed, You can enjoy these wav.
Under the Windows platform, PCM-encoded WAV is the best supported audio format, all audio software can be perfectly supported, because it can achieve high sound quality requirements, therefore, WAV is also the preferred format for music editing, suitable for saving music material. As a result, PCM-encoded WAV is used as an intermediary format, often in the conversion of other encodings, such as MP3 to WMA.


In a nutshell, PCM is not a compressed format.
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Basic operating principle of PCM
Pulse modulation is a time continuous, value continuous analog signal transformed into time discrete, the value of discrete digital signal after the transmission in the channel. Pulse code modulation is the process of sampling the analog signal, then quantifying the magnitude of the sample value and coding.
The so-called sampling, is a periodic scan of the analog signal, the time of continuous signal into a discrete signal time. After sampling, the analog signal should contain all the information in the original signal, that is, the original analog signal can be restored without distortion. The lower limit of its sampling rate is determined by the sampling theorem. In this experiment, Sampling rate is 8kbit/s.
The so-called quantification, that is, the instantaneous value obtained by sampling is discrete, that is, with a set of specified level, the instantaneous sampling value with the closest level value to represent.
After sampling and quantization of an analog signal, a quantized pulse amplitude modulated signal is obtained, which is only a finite number.
The so-called coding, is to use a set of binary code group to represent each fixed level of the quantization value. However, in fact, the quantization is in the coding process at the same time, so the coding process is also known as the modulus/number transformation, can be recorded as A/d.
PCM voice signal first through anti-aliasing low-pass filter, pulse sampling, into a 8KHz repetition frequency sampling signal (that is, the discrete pulse amplitude-modulated Pam signal), and then the amplitude of continuous PAM signal with a "rounding" method to quantify the value of a finite range of signals, and then converted to binary code after encoding. For phones, CCITT a sampling rate of 8KHz, each sample value of 8-bit code, that is, a total of 28 = 256 quantization values, so that the standard digital rate of PCM encoding per session is 64kb/s. In order to solve the problem of large quantization error of small signal and poor sound quality in uniform quantization, In practice, the nonlinear quantization method of quantization interval is adopted, that is, the quantization characteristic is layered in small signal, the quantization interval is small, while the large signal is stratified and the quantization interval is large.