Audio Encoder Multi-Mode Coding for Bitrate and Quality Tradeoffs
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Conventional audio coding methods face challenges in achieving improved audio quality due to tradeoffs between quantization errors and reconstruction artifacts, particularly in variable bit rate scenarios where dynamic detection of quantization and reconstruction regions is not sufficient.
Innovation Solution
An encoder device and method that apply multiple modes to an input signal, determining an optimum mode based on processed outputs to minimize distortion, allowing for selective quantization and reconstruction, thereby improving audio quality while managing computational complexity.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If quantization precision is increased to reduce quantization errors, then audio quality improves, but the number of bits required increases
Solution Approach 1:
The spectrum is divided into multiple frequency bands, with different quantization strategies applied to each band. Low-frequency bands receive higher quantization precision while high-frequency bands use lower precision, allowing bit allocation to be optimized across different frequency regions rather than uniform quantization across the entire spectrum.
Solution Approach 2:
Different quantization precision levels are applied to different parts of the signal spectrum based on local characteristics. The system dynamically adjusts the number of bits allocated to each frequency band according to the signal energy and importance, providing high precision where needed and lower precision where acceptable, thus improving overall audio quality without proportionally increasing total bit rate.
2Quantity of substance
If reconstruction is used to reduce bit rate, then the number of bits decreases, but reconstruction artifacts are introduced
Solution Approach 1:
The system dynamically selects between full quantization and reconstruction modes for each frequency band based on signal characteristics and available bit rate. Rather than statically applying reconstruction to all high-frequency components, the encoder adapts the reconstruction strategy in real-time, choosing to quantize certain bands fully while using reconstruction for others, thereby minimizing artifacts while maintaining low bit rate.
Solution Approach 2:
The system changes the reconstruction parameters (such as the reconstruction filter characteristics and the decision threshold for when to reconstruct versus when to quantize) based on the input signal properties. By adjusting these parameters dynamically, the system optimizes the trade-off between bit rate reduction and artifact generation for different types of audio content.
3Adaptability or versatility
If dynamic detection of quantization and reconstruction regions is implemented, then adaptability improves, but device complexity increases
Solution Approach 1:
The frequency spectrum is segmented into a fixed number of predetermined bands, eliminating the need for complex dynamic boundary detection. The encoder simply evaluates signal characteristics within each predefined band and makes quantization/reconstruction decisions independently for each segment, reducing computational complexity compared to continuous region detection while maintaining good adaptability.
Solution Approach 2:
The system uses the signal's own energy distribution across frequency bands to automatically determine which regions to quantize and which to reconstruct, without requiring external control or complex analysis. Each frequency band self-determines its processing mode based on local energy measures, simplifying the overall control logic while maintaining dynamic adaptability to different audio content.
Data Source
AI summary
The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the first processed output and the second processed output, and the output according to the optimum mode is selected.