Audio Resampling Support Vector for Low-Delay Codec Switching
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Solution Overview
Problem
Existing audio frequency signal resampling methods introduce algorithmic delays that can result in audible artifacts and are not effectively compensated, particularly in codecs like AMR-WB and G.722, where delay compensation techniques either degrade signal quality or introduce periodic discontinuities.
Innovation Solution
An adaptive linear prediction method is employed to determine the number of future signal samples based on the chosen resampling delay, constructing a resampling support vector from current and predicted samples, and applying a resampling filter to reduce delays while maintaining performance close to continuous resampling.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If FIR filter resampling is used to ensure linear phase and preserve waveform, then signal quality is improved, but algorithmic delay increases (1 to a few ms)
Solution Approach 1:
The patent applies preliminary action by predicting future signal samples before they are actually available. The linear prediction module generates predicted future samples that are used in the resampling filter, allowing the system to compensate for the filter delay in advance. This enables the output signal to be available earlier without sacrificing the quality benefits of linear phase filtering.
Solution Approach 2:
The patent introduces an intermediary mechanism - the predicted future samples - that mediates between the current input samples and the resampling filter output. These predicted samples act as a bridge, allowing the filter to operate with its full impulse response length while compensating for the delay introduced by the filtering process.
2Device complexity
If IIR filter resampling is used to reduce filter order and complexity, then device complexity is reduced, but phase linearity is lost unless additional all-pass filtering is added
Solution Approach 1:
The patent uses preliminary action by predicting future samples before they are needed in the resampling process. This allows the use of simpler IIR filters without requiring complex all-pass phase compensation stages, as the prediction mechanism inherently compensates for the non-linear phase characteristics of IIR filters.
3Loss of time
If resampling delay is reduced by using fewer future samples, then loss of time is reduced, but measurement precision of signal quality deteriorates
Solution Approach 1:
The patent applies dynamics by making the number of predicted future samples configurable and adaptable. The system can dynamically adjust the prediction horizon based on the specific application requirements, allowing optimization between delay reduction and signal quality maintenance. This dynamic approach enables the system to adapt to different operating conditions and quality requirements.
Data Source
AI summary
A method and device for resampling an audio frequency signal in an audio frequency signal coding or decoding. The method includes the following acts for each signal block to be resampled: determining, by adaptive linear prediction, a number of future signal samples, this number being defined as a function of a chosen resampling delay; constructing a resampling support vector from at least samples of the current block and determined future signal samples; applying a resampling filter to the samples of the resampling support vector.