Audio Decoder Gain Interpolation for Continuous Frame Waveforms
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Solution Overview
Problem
Existing audio coding techniques, such as MPEG AAC, face issues with discontinuous temporal waveforms in audio signals due to uniform gain correction across frames, leading to degradation in sound quality when significant gain changes occur between frames.
Innovation Solution
A decoding apparatus and method that reads encoded gain values and interpolation information to apply either linear or non-linear interpolation between gain sample positions, allowing for more precise gain value calculation at each sample position, thereby smoothing the audio waveform and reducing code amount.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Device complexity
If uniform gain correction is applied across all samples within one frame, then the coding complexity is reduced and processing is simplified, but the temporal waveform becomes discontinuous between frames when gain changes significantly, degrading sound quality
Solution Approach 1:
The frame is divided into multiple sub-sections (e.g., 4 sub-frames of 256 samples each within a 1024-sample frame), and separate gain values are calculated and applied to each sub-section. This segmentation allows the gain to vary within a frame, maintaining waveform continuity while keeping the overall structure manageable and not excessively complex.
Solution Approach 2:
The gain value transitions from being static (uniform across the entire frame) to dynamic (varying across different sub-sections within the frame). By applying different gain values to different temporal segments, the system adapts to local variations in the audio signal, ensuring smooth transitions and continuous waveforms while preserving sound quality.
2Reliability
If gain values are adjusted frequently to maintain waveform continuity, then sound quality is improved, but the code amount increases and coding efficiency decreases
Solution Approach 1:
Instead of calculating gain for every sample or every small segment, the frame is segmented into a moderate number of sub-sections (e.g., 4 sub-frames). This segmentation strikes a balance between achieving sufficient waveform continuity and limiting the number of gain values that need to be encoded, thereby controlling the code amount.
Solution Approach 2:
Rather than applying gain correction at every possible point (which would ensure perfect continuity but increase code amount), the invention applies gain correction at selected sub-section boundaries. This partial action provides sufficient waveform continuity for perceptual quality while avoiding the excessive code overhead of full-sample-level gain adjustment.
Data Source
AI summary
The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality.A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.