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Dynamic compensation of audio signals for improved perceived spectral imbalances

一种音频信号、补偿数据的技术，应用在数字信号音调/带宽控制、数字/编码信号组合控制、音调控制等方向，能够解决不能增亮低比特率编码的材料等问题

Active Publication Date: 2013-08-21
DTS
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  • Summary
  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

[0015] Applying any type of high frequency EQ may not brighten low bitrate encoded material

Method used

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  • Dynamic compensation of audio signals for improved perceived spectral imbalances
  • Dynamic compensation of audio signals for improved perceived spectral imbalances
  • Dynamic compensation of audio signals for improved perceived spectral imbalances

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Experimental program
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Embodiment Construction

[0043] definition. As used in this specification and the appended claims, unless the context requires otherwise, the following terms shall have the indicated meanings:

[0044] A continuous (analog) audio signal can be digitally sampled at a "sampling frequency" to form a stream of digital data. Common sampling frequencies include: 44.1 kHz used by MPEG-1 Audio including MP3; 48 kHz used by various professional digital video standards such as SDI; and 96 kHz used for DVD Audio, Blu-ray Audio and HD-DVD Audio. The digital data represents a "sample period" which is defined as the time between samples of the audio signal.

[0045] Digital data of a sampling period may be transformed from a time-based representation ("time domain") to a frequency-based representation ("frequency domain") by using well-known transforms such as the discrete cosine transform (DCT). While data values ​​in the time domain may represent a range of voltage magnitudes (for example), data values ​​in the...

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Abstract

An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth; The final, compensated audio signal is sent to the playback device for playback.

Description

[0001] (cross-reference to related application) [0002] This application claims the benefit of US Provisional Application No. 61 / 381831, filed September 10, 2010, which is hereby incorporated by reference in its entirety. technical field [0003] The present invention relates to methods for equalizing an audio signal for playback by using adaptive filtering. Background technique [0004] An audio signal can be described by its spectral balance or frequency response. When played in a playback device, the audio signal has an associated sound pressure level or "SPL". These two properties of an audio signal are logically independent: assuming a linear time-invariant reproduction system, changing the sound pressure level of an audio signal should not affect any objective measure of the spectral balance of that signal. [0005] However, from a subjective psychoacoustic point of view, we observe that changes in the sound pressure level produce significant changes in the perceive...

Claims

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Application Information

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Patent Type & Authority Applications(China)
IPC IPC(8): H03G3/00
CPCH03G5/005H03G5/025H03G5/165H03G9/005H03G9/025H03G3/00
Inventor M·沃尔什E·斯特因J-M·卓特
Owner DTS