What is the difference between audio output PCM and lpcm?

Source: Internet
Author: User
Multi-channel lpcm: the original format of lossless audio tracks. It is equivalent to a wave file and does not require decoding. You can directly input a Power Amplifier for DA conversion. The fiber optic and coaxial interfaces can only transmit two-channel lpcm, multi-channel lpcm requires HDMI interface transmission. PCM: nonlinear pulse Coding Modulation
Lpcm: linear pulse Coding Modulation
They are an encoding method for converting analog speech signals to digital signals, which is lossless non-compression encoding.

Conversion process: Sampling --> quantification --> Encoding
The sampling process changes the continuous time analog signal into a discrete time, continuous range sampling signal.

The quantization process is a digital signal that changes the sampling signal to discrete time and discrete amplitude.
The quantization process is divided into linear quantization and non-linear quantization.
Linear quantization is within the entire quantization range, and the quantization interval is equal.
Non-linear quantization uses unequal quantization intervals

The encoding process is to encode the quantified signal into a binary code group for output.

Main differences:
Because they have different quantization intervals, the number of digits of the last binary encoding is different.

Other differences:
In PCM, corresponding technologies are sometimes used to reduce the data rate of digital signal sources, facilitating storage and data transmission.
Generally, more Dolby Surround Sound Replay technologies are used in the lpcm to reproduce the original sound more realistically. The common CD specification of lpcm (PCM) (linear pulse Coding Modulation) is 16 bit/44.1 kHz, and there are many DVD specifications. The quantization accuracy can be divided into 16 bit, 20 bit, and 24bit, the sampling frequency is divided into 48 khz and 96 kHz. In addition, the Dolby Surround Sound information can be entered in the lpcm signal for the existing Dolby directional logic surround sound system. A vob file consists of video, sound, and Subtitle data streams. Video data stream is MPEG2 format, audio data stream is a AC-3 or the lpcm, MPEG2, MP2, DTS and so on, AC3 is basically the standard of fact, MPEG2 multi-channel can be seen only on a very small number of 2-area discs (such as in the line of fire, Zone 2 ). PCM is mainly used for music DVDs, while MP2 is only available on cheap DVDs. PCM is a high-quality, non-Compressed digital audio. Therefore, it requires too much space and is not suitable for use on DVD and cine discs. The data rate of AC3 is between 192 ~ Between 448kbps, 384 kbps for dual-channel ~ 5.1 kbps is used for channels. The main methods for editing the digital audio of this segment are compression and non-compression. Early digital audio players, such as CD players and dat recorders, all use linear PCM encoding to store Music signals. They are not compressed. On high-quality audio workstations and digital recorders (such as dvcpro), non-compressed formats are also used. Currently, common MPEG, Dolby Digital, and DTs are compression methods. Compression can be divided into lossy compression and lossless compression. Lossy compression aims to increase the compression rate and reduce the occupation of system resources. Different sampling rates, sample resolution (precision), and data rates can be selected as needed. Today, Dolby Digital is part of the ATSC digital TV standard selected by the FCC for the United States, as well as the standard for High Definition TV (HDTV) and standard definition TV (sdtv) broadcast. MPEG is an audio standard for European digital video broadcast (DVB), digital audio broadcast (DAB), and Japanese broadcast and television. DVD supports three major standards: Dolby Digital (Dolby Digital), MPEG-2 and linear PCM (lpcm ). Other formats, such as DTS (Digital theatre sound) and SDDS (Sony dynamic digital sound), are optional. The evolution of Sound Replay Technology is from monophonic, dual-channel stereo (stereophonic) to four-channel stereo, and then to stereo surround, which is now generally in 5.1 mode. The fundamental purpose is to reproduce the original sound more realistically. Currently, Chinese TVs are widely used in single-channel mode, which is far behind the needs of people's daily lives. How to transmit multi-channel and high-quality sound with a low data rate is the development direction of digitization. So-called 5? 1 mode, that is, the recording, decoding, and sound recording adopt five channels: Left (L), center (C), right (R), left surround (LS), right surround RS), plus a low frequency effect channel (LFE), you can achieve the real three-dimensional surround sound effect-wide scene depth and overall sense of reality. Standard channels of ATSC and DVB are used in the 5.1 mode. The reason why non-compressed (PCM) sound can be digitalized is that the voice frequency that human ears can hear is not infinitely wide, mainly below 20 kHz. According to the sampling theorem, the original sound can be reconstructed without distortion only when the sampling frequency is greater than 40 kHz. For example, the sampling frequency of CD is 44.1khz, and the sampling frequency of others is 48 khz or 96 kHz. PCM (pulse Coding Modulation) is a method for converting analog speech signals into digital signals. There are three main processes: sampling, quantification, and coding. The sampling process changes the continuous time analog signal into a discrete time, continuous amplitude sampling signal, and the quantization process changes the sampling signal into a discrete time, discrete amplitude digital signal, the encoding process encodes the quantified signal into a binary code group output. Quantization is classified into linear quantization and non-linear quantization. Linear quantization is within the entire quantization range, and the quantization interval is equal. Non-linear quantization uses unequal quantization intervals. The Quantization interval is determined by the number of encoded binary digits. For example, if CD uses 16-bit linear quantization, the quantization interval is L = 65536. The more digits (n), the higher the accuracy, and the higher the SNR = 6.02n + 1.76 (db. However, the number of binary digits of the encoding is not unlimited and must be determined based on the required data rate. For example, can the data rate of a CD be 2x44.1x16 = 1411? 2 kbit/s. There are three types of commonly used encoding code groups: natural binary code group (NBC), collapsed binary code group (FBC), and gray binary code group (RBR ). International PCM standards mainly use FBC. Although the compressed PCM is lossless compression, the signal characteristics represented by typical audio signals are not optimal, and it is not well adapted to the specific requirements of the human ears auditory system. The data size of PCM is too high, which leads to storage and transmission obstacles. Therefore, it is necessary to use the corresponding technology to reduce the data rate of the digital signal source and avoid program damage as much as possible. This is the compression technology. Human ears have two characteristics: frequency masking and time masking. In a quiet environment, human ears have a threshold (threshold), which corresponds to the minimum sound intensity that can be felt within the frequency range that human ears can hear. Frequency masking, that is, when a single audio unit appears, a new curve is generated (same as the hearing threshold). In the frequency band near this frequency, the threshold is increased to varying degrees, the center frequency is the highest. Time masking: When a strong signal appears, the existing weak sound can be masked for a period of time before and after it. Audio Signals below the threshold do not need to be encoded. Turn: http://blog.chinaunix.net/uid-9688646-id-1998399.html

(Conversion) What is the difference between audio output PCM and lpcm?

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