Low-Complexity Audio Matrix Decoder for Mobile Devices

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Solution Overview

Problem

Existing matrix decoders, such as Dolby Pro Logic II, are too complex and resource-intensive for use in low-cost devices like multimedia mobile phones and portable game consoles, which require multichannel surround sound inputs while being limited to two-channel stereo outputs.

Innovation Solution

A low-complexity adaptive audio matrix decoder using a fixed-matrix-variable-gains approach that minimizes inter-channel leakage and avoids correlations between channels, employing a four-channel output configuration (Left, Right, Left Surround, and Right Surround) to achieve directional sound without increasing processing time, and incorporates a panning function to balance soundstage and directionality.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Device complexity

If existing matrix decoders (Dolby Pro Logic II) are used to decode multichannel surround sound into two-channel stereo output, then directional sound and soundstage are achieved, but device complexity and computational resources increase beyond what low-cost devices can handle

Engineering Contradiction:
Improvedecoder complexityVSAvoiddecoding performance
Core Design Contradiction:
Device complexityVSReliability

Solution Approach 1:

The patent extracts only the essential functions needed for multichannel-to-stereo decoding, removing complex adaptive matrix operations. It uses a fixed matrix with simple gain control and level normalization, keeping only the critical components for achieving directional sound while discarding resource-intensive features of existing decoders.

Inventive Principle:
Principle #2Taking out (Extraction)

Solution Approach 2:

The patent changes the operational parameters by using fixed matrix coefficients instead of adaptive ones, and applies level normalization to control output levels. This transforms the complex adaptive decoding process into a simpler fixed-parameter system that maintains perceptual quality while reducing computational burden.

Inventive Principle:
Principle #35Parameter changes

2Device complexity

If a fixed-matrix-variable-gains approach is used to reduce complexity, then computational resources are minimized, but inter-channel leakage and signal correlations may increase

Engineering Contradiction:
Improveprocessing complexityVSAvoidsignal separation
Core Design Contradiction:
Device complexityVSLoss of information

Solution Approach 1:

The patent introduces level normalization that monitors output levels and adjusts gains accordingly. This feedback mechanism prevents inter-channel leakage by ensuring that no single channel dominates the output, maintaining signal separation despite using a fixed matrix approach.

Inventive Principle:
Principle #23Feedback

Solution Approach 2:

While the matrix itself is fixed, the patent introduces dynamic gain control through level normalization. This allows the system to adapt to different input conditions and maintain proper signal separation without requiring a complex adaptive matrix, achieving a balance between simplicity and signal integrity.

Inventive Principle:
Principle #15Dynamics

3Reliability

If virtualizer processing is applied to multichannel audio signals, then directional sound perception is improved, but the requirement for multichannel surround sound input increases device complexity

Engineering Contradiction:
Improvedirectional sound perceptionVSAvoidsystem complexity
Core Design Contradiction:
ReliabilityVSDevice complexity

Solution Approach 1:

The patent combines the matrix decoding function with level normalization in a single integrated process. By merging these functions and using a fixed matrix, it eliminates the need for separate complex processing stages, enabling virtualizer-like directional sound perception in low-cost devices with limited computational resources.

Inventive Principle:
Principle #5Merging (Combining)

Data Source

PatentEP1964443B1Low-complexity audio matrix decoder
Publication Date: 2011.09.21 DOLBY LABORATORIES LICENSING CORP
  • EP1964443B1 patent drawingFigure 1~2
  • EP1964443B1 patent drawingFigure 3~4
  • EP1964443B1 patent drawingFigure 5~6

AI summary

Deriving n audio output signals from m audio input signals, where m and n are positive whole integers and the n audio output signals are derived using an adaptive matrix or matrixing process responsive to one or more control signals, which matrix or matrixing process produces n audio signals in response to m audio signals, and (b) deriving a plurality of time- varying control signals from the m audio input signals, wherein the control signals are derived from the m input audio signals using (i) a processor or process that produces a plurality of directional dominance signals in response to the m audio input signals, at least one directional dominance signal relating to a first directional axis and at least one other directional dominance signal relating to a second directional axis, and (iÊ) a processor or process that produces the control signals in response to the directional dominance signals.