Dual-Channel Echo Postfiltering for Mobile Audio Signal Processing
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Solution Overview
Problem
Existing echo cancellation methods in mobile terminals face limitations in suppressing echoes during double-talk periods, leading to high distortions in near-end speech signals due to the trade-off between echo suppression and speech quality, especially with limited computational complexity.
Innovation Solution
A dual-channel echo postfiltering method using two adaptive filters, one for each microphone path, to compute echo suppression gains based on power spectral density and signal-to-echo ratios, reducing residual echo and maintaining moderate computational complexity.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Object-generated harmful factors
If single-channel echo postfiltering is used, then echo suppression is achieved, but near-end speech distortion increases during double-talk periods
Solution Approach 1:
The patent transitions from single-channel to multi-channel echo postfiltering by utilizing signals from multiple microphones. This dimensional change allows the system to exploit spatial information and statistical independence between channels, achieving superior echo suppression while preserving near-end speech quality during double-talk periods through joint spectral analysis across multiple channels.
Solution Approach 2:
The patent combines information from multiple microphone channels by computing cross-spectral densities and integrating spectral estimates across channels. This merging of multi-channel signals enables the system to distinguish between echo and near-end speech more effectively, reducing speech distortion while maintaining echo suppression performance.
2Object-generated harmful factors
If multi-channel echo postfiltering is implemented, then echo suppression performance improves, but computational complexity increases
Solution Approach 1:
The patent segments the computational process into distinct stages: computing cross-spectral densities for each channel separately, then combining these segmented results through weighted averaging. This segmentation allows efficient parallel computation and reduces the overall computational burden compared to processing all channels simultaneously with full covariance matrices.
Solution Approach 2:
The patent employs parameter changes by using spectral estimates and cross-spectral densities as intermediate parameters to simplify the multi-channel filtering computation. By transforming the problem into the frequency domain and using these spectral parameters, the system achieves efficient computation while maintaining the benefits of multi-channel processing.
3Object-generated harmful factors
If aggressive echo suppression is applied, then echo level is reduced, but near-end speech distortion increases
Solution Approach 1:
The patent implements feedback mechanisms by continuously monitoring the spectral characteristics of multi-channel signals and dynamically adjusting the echo suppression gain. The system uses feedback from cross-spectral density computations to adaptively control the filtering strength, ensuring aggressive echo suppression when needed while automatically reducing suppression during double-talk periods to preserve speech quality.
Solution Approach 2:
The patent introduces dynamics by making the echo suppression gain time-varying and adaptive based on real-time spectral analysis. The system dynamically adjusts suppression levels by computing instantaneous spectral estimates and cross-spectral densities, allowing the filter to respond adaptively to changing acoustic conditions and distinguish between echo-only and double-talk periods for optimal speech preservation.
Data Source
AI summary
A method for processing audio signals is provided comprising outputting an audio signal; receiving the output audio signal via a first receiving path as a first received audio signal; receiving the output audio signal via a second receiving path as a second received audio signal; determining an echo suppression gain based on the first received audio signal and the second received audio signal; and filtering echo suppression of the audio signal based on the first received audio signal and the echo suppression gain.