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Method and equipment for coding and decoding of digital information signal

A digital information and signal technology, applied in the field of encoding and decoding of digital information signals and equipment, can solve problems such as length troubles

Inactive Publication Date: 2003-09-17
INTERDIGITAL CE PATENT HLDG
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Problems solved by technology

Differences in length are troublesome when editing audio signals or linking with precise timing

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  • Method and equipment for coding and decoding of digital information signal
  • Method and equipment for coding and decoding of digital information signal
  • Method and equipment for coding and decoding of digital information signal

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Embodiment Construction

[0023] In stereophonic or audio processing, the resulting analog signal (eg, the output of a microphone amplifier) ​​is converted to a digital signal using sampling and quantization theorems. 'Sampling' means that the amplitude value of the signal is taken at a certain interval. The reciprocal value of the time interval is the sampling rate. According to Nyquist's law or sampling theory, if the highest frequency contained in the signal is only half of the sampling rate, the original content of the sampled signal can be restored without error. Typical sampling rates for audio processing are 44.1kHz and 48kHz, which correspond to sampling intervals or clocks of 22.67μs and 20.83μs, respectively. 'Quantization' is based on the quantization characteristic, so that the reduction of the magnitude value can be basically distributed to the finely resolved signal sample value. Consequently, the resolution of the magnitude values ​​is limited, and an irreversible loss of information d...

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Abstract

A raw digital audio signal is represented as PCM samples, where the distance between values ​​corresponds to the sampling frequency. Digital signals only have lengths that are integer multiples of time elements. Specially coded digital audio signals are processed on a block basis, resulting in an overall length that is only an integer multiple of the block unit. According to the invention, the exact length of the original signal is transmitted with the encoded audio information. Furthermore, an information value representative of the total encoder and / or decoder delay is transmitted. The decoder takes these information items and cuts samples from the decoded program or audio-video track to adjust the overall length of the decoded signal.

Description

technical field [0001] The present invention relates to methods and devices for encoding and decoding signals at reduced bit rates, in particular digital audio signals. Background technique [0002] The digital representation of an analog audio signal has a temporal structure that originates from the sampling process. A digital audio signal represented in PCM format consists of a list of values, where the distance between the values ​​corresponds to the sampling frequency. This distance is the smallest element of the signal that can be defined in the time domain. Digital signals are only as long as integer multiples of this time element length. [0003] Coders and decoders that reduce the bit rate of digital audio signals (such as MPEG1 / 2 / 4-Audio, Dolby Digital Ac-3, mp3, ATRAC, Windows MediaAudio WMA or Real Audio) typically represent the signal in the short-term frequency domain to operate. To convert a signal to the frequency domain, typical numbers of signal elements...

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Application Information

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Patent Type & Authority Applications(China)
IPC IPC(8): G11B20/12G10L19/00G10L19/16H03M7/30
CPCG10L19/167G10L19/00H03M7/30
Inventor 恩斯特·F·施罗德约翰内斯·伯姆
Owner INTERDIGITAL CE PATENT HLDG
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