Adaptive Audio Encoding Mode Selection for Compression Quality
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Solution Overview
Problem
Conventional voice/audio compression modes suffer from timbre deterioration when compressing signals that are not of their intended type, leading to suboptimal compression performance.
Innovation Solution
An adaptive time/frequency-based encoding apparatus that determines the encoding mode for each frequency band by extracting long-term and short-term features from both time and frequency domains, allowing for efficient switching between time-based and frequency-based encoding modes to optimize compression performance.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If audio codec (frequency-based encoding) is used to compress voice signal, then compression ratio is improved, but timbre is deteriorated
Solution Approach 1:
The patent changes the encoding parameter (time-domain vs frequency-domain) based on the signal type parameters (voice-like vs audio-like characteristics). By analyzing signal properties and dynamically selecting the appropriate encoding domain, the system optimizes both compression ratio and timbre quality for different signal types.
Solution Approach 2:
The patent implements dynamic mode switching between time-based and frequency-based encoding. The encoding mode is not fixed but adapts in real-time based on the input signal characteristics, allowing the system to transition between ACELP and TCX modes to maintain high quality while achieving good compression.
2Manufacturing precision
If voice codec (time-based encoding) is used to compress audio signal, then timbre is improved, but compression ratio is deteriorated
Solution Approach 1:
The system changes the encoding parameter based on signal type detection. When the input is detected as audio-like rather than voice-like, the system switches from time-based to frequency-based encoding, thereby improving compression ratio while maintaining acceptable timbre quality for the specific signal type.
Solution Approach 2:
The patent implements dynamic adaptation of the encoding approach based on real-time signal analysis. The system continuously evaluates the input signal characteristics and adjusts the encoding mode accordingly, enabling optimal performance for both voice and audio signals rather than being constrained to a single encoding approach.
3Adaptability or versatility
If AMR-WB+ mode determines encoding mode for each frame, then adaptability is improved, but timbre deterioration and compression ratio degradation occur for audio-like signals
Solution Approach 1:
The patent employs parameter-based signal classification to determine the appropriate encoding mode. By analyzing specific signal characteristics and changing the encoding parameter accordingly, the system achieves better results for audio-like signals compared to frame-based mode determination, reducing timbre deterioration while maintaining adaptability.
Data Source
AI summary
An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.