Scalable Multi-Channel Audio Coding With Dominant and Residual Layers
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Solution Overview
Problem
Existing multi-channel audio coding technologies face challenges in achieving high quality audio at low bit rates while maintaining backward compatibility, particularly in 5.1 channel configurations, as they often require high bit rates and lack scalability.
Innovation Solution
The proposed solution involves an encoder that generates a dominant signal part and a residual signal part, using spatial parameters to create a scalable representation of audio signals. The dominant signal part can be decoded with existing decoders for acceptable quality, while the residual signal part, when used, enhances the quality significantly. This is achieved through a mathematical procedure that combines audio signals into orthogonal or least correlated parts, with the residual signal part being low-pass filtered and optionally included for improved sound quality.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If MPEG-2 Layer II is used to code multi-channel audio, then backward compatibility is maintained, but the bit rate is high (640 kbit/s)
Solution Approach 1:
The multi-channel audio signal is segmented into a dominant signal part and a residual signal part. The dominant part is transmitted in the base layer for backward compatibility, while the residual part is transmitted in the refinement layer to improve audio quality. This segmentation allows the system to maintain compatibility with legacy decoders while providing enhanced quality for advanced decoders at a lower overall bit rate.
Solution Approach 2:
The patent introduces a new dimension of scalability by organizing the bit stream into multiple layers (base layer and refinement layer). The base layer provides a standalone acceptable quality signal, while the refinement layer adds an additional dimension of quality enhancement. This layered structure enables backward compatibility while allowing for quality improvement without proportionally increasing bit rate.
2Quantity of substance
If AAC is used to code multi-channel audio, then bit rate is reduced (320 kbit/s), but backward compatibility is lost
Solution Approach 1:
The audio coding is segmented into a base layer that maintains backward compatibility and a refinement layer that provides quality enhancement. The base layer can be decoded by legacy decoders, while the refinement layer contains the advanced coding information. This segmentation resolves the contradiction by allowing both low bit rate operation and backward compatibility to coexist.
Solution Approach 2:
The bit stream structure is designed to serve multiple functions: it provides backward compatibility for legacy decoders, enables scalable quality enhancement for advanced decoders, and allows flexible bit rate adaptation. The same bit stream can be decoded at different quality levels depending on the decoder capabilities and available bandwidth.
3Quantity of substance
If parametric coding is used to reduce bit rate, then scalability is improved, but audio quality may be compromised
Solution Approach 1:
The audio signal is segmented into dominant and residual parts, where the dominant part captures the most perceptually important information and the residual part contains the remaining details. By transmitting both parts with appropriate precision, the system achieves low bit rate operation while maintaining high audio quality through the complementary nature of the two segments.
Solution Approach 2:
The patent employs parametric coding techniques to represent the residual signal efficiently by encoding parameters rather than full waveform data. This parameter-based representation significantly reduces the bit rate required for the refinement layer while maintaining the ability to reconstruct high-quality audio when combined with the base layer.
Data Source
AI summary
An audio encoder for encoding a multi-channel audio signal includes an encoder combination module (ECM) for generating a dominant signal part (m) and a residual signal part (s) being a combined representation of first and second audio signals (x1, x2), the dominant and residual signal parts (m, s) being obtained by applying a mathematical procedure to the first and second audio signals (x1, x2), wherein the mathematical procedure involves a first spatial parameter (SP1) including a description of spatial properties of the first and second audio signals (x1, x2), a parameter generator (PG) for generating a first parameter (PS1) set including a second spatial parameter (SP2), and a second parameter (PS2) set including a third spatial parameter (SP3), and an output generator for generating an encoded output signal having a first output part (OP1) including the dominant signal part (m) and the first parameter set (PS1), and a second output part (OP2) including the residual signal part (s) and the second parameter set (PS2).