QMF Audio Codec Resampling via Reconfigurable Filter Banks
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Solution Overview
Problem
Current audio processing technologies face challenges in efficiently handling different sampling rates, particularly at low bit rates, which require additional computational resources and are not compatible with low-cost hardware, limiting audio bandwidth and increasing complexity.
Innovation Solution
The proposed solution involves a configurable audio signal processor with analysis and synthesis filter banks that adjust the number of channels based on configuration settings, allowing for different sampling rates and reducing computational resources by modifying the frequency domain representation and using low-complexity resamplers.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Adaptability or versatility
If additional resampling functional modules are used to adapt the output sampling rate, then the sampling rate compatibility is improved, but the computational resources and device complexity increase significantly
Solution Approach 1:
The patent merges the resampling functionality with the existing analysis and synthesis filter banks already present in the audio codec. By integrating resampling into these existing structures rather than adding separate resampling modules, the system achieves sampling rate adaptation without proportionally increasing device complexity. The filter banks serve dual purposes: their traditional signal processing function and the resampling function.
Solution Approach 2:
The analysis and synthesis filter banks are designed to perform multiple functions: traditional audio coding/decoding operations and sampling rate adaptation. This multi-functionality allows a single set of filter banks to handle both signal processing and resampling tasks, reducing the need for additional dedicated resampling hardware or software modules.
2Adaptability or versatility
If additional resampling functional modules are used to adapt the output sampling rate, then the sampling rate compatibility is improved, but the computational resources increase significantly
Solution Approach 1:
The patent merges the resampling functionality with the existing analysis and synthesis filter banks already present in the audio codec. By integrating resampling into these existing structures rather than adding separate resampling modules, the system achieves sampling rate adaptation without proportionally increasing device complexity. The filter banks serve dual purposes: their traditional signal processing function and the resampling function.
Solution Approach 2:
The filter banks perform resampling as a byproduct of their normal operation. The analysis filter bank decomposes the input signal into subbands, and the synthesis filter bank reconstructs it, and this inherent frequency domain processing automatically provides resampling capability when the number of analysis and synthesis channels differ, without requiring additional computational effort beyond what's already needed for audio coding.
3Adaptability or versatility
If the number of analysis and synthesis filter bank channels are increased to improve resampling flexibility, then the sampling rate adaptation capability is improved, but the computational resources and device complexity increase
Solution Approach 1:
The patent employs dynamic control of the filter bank channel configurations. The controller adaptively selects and switches between different numbers of analysis and synthesis filter bank channels based on the desired sampling rate conversion ratio. This dynamic reconfiguration allows the system to optimize the balance between resampling flexibility and computational efficiency for each specific resampling task, rather than being fixed at a single channel configuration.
Solution Approach 2:
The system changes the operational parameters of the filter banks, specifically the number of analysis and synthesis channels, to achieve different sampling rate conversions. By varying these parameters dynamically based on the resampling requirements, the system can adapt to different sampling rate ratios while maintaining computational efficiency, avoiding the need for a fixed high-channel configuration that would always consume maximum resources.
Data Source
AI summary
An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.


