Dual-Mode Audio Processing for Voice Fidelity and Legacy Playback
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Solution Overview
Problem
Current audio codec systems are often fragmented and lack coherence due to cumulative development, leading to inefficient performance, especially for voice signals, and struggle with backward compatibility with legacy equipment.
Innovation Solution
A unified audio processing system with a dual-mode architecture that includes a front-end component for dequantization and inverse transformation, and a processing stage for frequency-domain processing, capable of adapting to voice mode for improved voice signal fidelity and supporting legacy playback through phase-shifting and spectral band replication.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If a unified audio processing system with dual-mode architecture is implemented, then voice signal fidelity and system adaptability are improved, but device complexity increases
Solution Approach 1:
The system implements dynamic adaptability by switching between audio mode and voice mode based on signal characteristics. The dual-mode architecture allows the system to dynamically adjust its processing parameters, transform lengths, and coding strategies to optimize performance for different signal types (general audio vs. voice), thereby improving adaptability without requiring completely separate systems for each application.
Solution Approach 2:
The unified audio processing system is designed to handle multiple functions within a single architecture. It can process both general audio signals and voice signals using the same core components (front-end, processing stage, synthesis stage), while adapting its behavior through mode selection. This multi-functionality improves versatility while controlling complexity by avoiding the need for separate dedicated systems.
2Ease of operation
If parametric multichannel audio codec systems provide backward compatibility with legacy equipment, then ease of operation is improved, but manufacturing precision and system coherence deteriorate
Solution Approach 1:
The system extracts and processes only the necessary components for backward compatibility. By implementing a downmix signal path that can be independently controlled and processed, the system can provide legacy-compatible output to legacy equipment while maintaining the full parametric multichannel capability for modern systems. This separation allows compatibility without compromising the coherence of the main system architecture.
Solution Approach 2:
The downmix signal acts as an intermediary between the parametric multichannel audio system and legacy playback equipment. By providing a separately processable downmix path, the system enables legacy equipment to receive compatible signals without requiring modifications to the core parametric encoding architecture, thus maintaining system coherence while achieving backward compatibility.
3Ease of manufacture
If cumulative and uncoordinated development continues, then ease of manufacture is improved, but device complexity and system performance worsen
Solution Approach 1:
The audio processing system is segmented into distinct functional stages: front-end component, processing stage, and synthesis stage. Each stage has clearly defined inputs and outputs, allowing for modular development and implementation. This segmentation enables coordinated development across different teams while maintaining overall system coherence, as each module can be developed independently but interfaces are well-defined.
Solution Approach 2:
The system uses parameter changes to adapt its behavior across different modes and operating conditions. By controlling parameters such as transform length, processing type, and coding strategy based on signal characteristics, the system achieves coordinated behavior across different development contributions. This parameter-based control allows cumulative development while maintaining system-wide coherence through centralized parameter management.
Data Source
AI summary
An audio processing system (100) comprises a front-end component (102, 103), which receives quantized spectral components and performs an inverse quantization, yielding a time-domain representation of an intermediate signal. The audio processing system further comprises a frequency-domain processing stage (104, 105, 106, 107, 108), configured to provide a time-domain representation of a processed audio signal, and a sample rate converter (109), providing a reconstructed audio signal sampled at a target sampling frequency. The respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the time-domain representation of the processed audio signal are equal. In particular embodiments, the processing stage comprises a parametric upmix stage which is operable in at least two different modes and is associated with a delay stage that ensures constant total delay.