Audio Decoder Clipping Prevention via Dynamic Level Shifting
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Solution Overview
Problem
Modern audio coding systems face challenges in preventing clipping of audio signals during decoding, particularly due to quantization errors and loudness normalization, which can result in audible artifacts like clicks or distortion, especially when using fixed-point arithmetic and limited bit resolution.
Innovation Solution
An audio signal decoder and encoder with a clipping estimator, level shifter, and level shift compensator are implemented to analyze side information and adjust signal levels dynamically, shifting information towards the least significant bit to create headroom and prevent clipping without increasing word length, allowing for efficient frequency-to-time and time-to-frequency conversions.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If quantization is applied to reduce transmission data rate, then transmission efficiency is improved, but clipping occurs due to accumulated quantization errors
Solution Approach 1:
The patent applies preliminary action by performing level adjustment before the frequency-to-time conversion process. The decoder estimates the signal level from frequency domain parameters, determines an appropriate shift factor, and adjusts the signal level in advance to prevent clipping from occurring during reconstruction. This proactive approach addresses clipping before it happens rather than correcting it afterward.
Solution Approach 2:
The patent implements feedback by using side information from the encoder about signal characteristics to guide the decoder's level adjustment process. The encoder provides information about signal level and potential clipping risk, and the decoder uses this feedback to dynamically adjust the reconstruction level, creating a closed-loop system that adapts to varying signal conditions.
2Device complexity
If fixed-point arithmetic with limited bit resolution is used, then computational complexity is reduced, but clipping occurs due to insufficient dynamic range
Solution Approach 1:
The patent applies dynamics by making the signal level adjustment adaptive rather than fixed. The shift factor is determined dynamically based on the estimated signal level from frequency domain parameters, allowing the system to optimize the balance between computational efficiency and clipping prevention for each specific signal condition. This dynamic adjustment enables fixed-point arithmetic to handle varying dynamic range requirements.
Solution Approach 2:
The patent changes the parameter of signal level (through shift factor adjustment) to prevent clipping. By modifying the gain parameter applied to the frequency domain coefficients before conversion, the system adapts the effective dynamic range to match the capabilities of fixed-point arithmetic, thereby preventing overflow without requiring higher precision data types.
3Manufacturing precision
If loudness normalization is applied to achieve unified playback levels, then audio quality consistency is improved, but clipping occurs when normalization gain increases signal levels
Solution Approach 1:
The patent performs preliminary level adjustment to account for upcoming loudness normalization operations. By estimating the signal level and applying an appropriate shift factor before frequency-to-time conversion, the system creates headroom that accommodates subsequent normalization gain application without causing clipping. This ensures that both quality consistency and clipping prevention are achieved.
Solution Approach 2:
The patent applies beforehand cushioning by introducing level adjustment that creates protective headroom in the signal. This cushioning effect allows loudness normalization to be applied safely by ensuring the signal has sufficient dynamic range buffer to absorb the normalization gain without exceeding the maximum representable level in fixed-point arithmetic.
4Reliability
If word length is increased in filter banks to prevent clipping, then clipping prevention is improved, but memory usage and power consumption increase
Solution Approach 1:
The patent performs preliminary level adjustment in the frequency domain before the computationally intensive frequency-to-time conversion. By preventing clipping at this earlier stage through shift factor application, the system eliminates the need for higher word length in the filter bank, thereby reducing power consumption and memory usage while maintaining clipping prevention.
Solution Approach 2:
The patent introduces an intermediary level adjustment operation that mediates between the frequency domain representation and the time domain conversion. This intermediary step prevents the need for high-precision arithmetic in the filter bank by ensuring the input signal is already scaled to an appropriate level, thus reducing the computational burden and resource requirements of the conversion process.
Data Source
AI summary
An audio signal decoder for providing a decoded audio signal representation on the basis of an encoded audio signal representation has a decoder preprocessing stage for obtaining a plurality of frequency band signals from the encoded audio signal representation, a clipping estimator, a level shifter, a frequency-to-time-domain converter, and a level shift compensator. The clipping estimator analyzes the encoded audio signal representation and/or side information relative to a gain of the frequency band signals in order to determine a current level shift factor. The level shifter shifts levels of the frequency band signals according to the level shift factor. The frequency-to-time-domain converter converts the level shifted frequency band signals into a time-domain representation. The level shift compensator acts on the time-domain representation for at least partly compensating a corresponding level shift and for obtaining a substantially compensated time-domain representation.