Audio Encoding with Transient Detection for Pre-Echo Control
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Solution Overview
Problem
Existing audio codec technologies fail to effectively control or eliminate pre-echo artifacts, especially noticeable in audio with sharp impulses and transient signals, due to inaccuracies during time domain to frequency domain conversion and back, leading to suboptimal sound quality in dynamic and distributed IP-based multimedia systems.
Innovation Solution
A computer-implemented system and method that encodes sampled audio signals by identifying potential pre-echo events, generating an error signal, and encoding this information into a bitstream, allowing for the removal of pre-echo events during downstream decoding, using techniques such as pulse code modulation, adaptive pulse code modulation, and modified discrete cosine transform.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If MDCT transform is used to convert time domain signal to frequency domain and back, then audio encoding efficiency is improved, but pre-echo artifacts are introduced due to error spreading across block size
Solution Approach 1:
The audio signal is divided into multiple blocks for MDCT processing, and the patent identifies specific blocks containing transient signals. By segmenting the processing approach - using different block sizes (short blocks for transients, long blocks for steady-state) and different coding modes (transform coding vs. predictive coding), the patent prevents error spreading across entire long blocks, thereby reducing pre-echo while maintaining encoding efficiency.
Solution Approach 2:
The patent dynamically adjusts the MDCT block size based on the detected signal characteristics. When transient signals are detected, short blocks are used to minimize pre-echo; when steady-state signals are present, long blocks are used to maintain encoding efficiency. This dynamic adaptation resolves the contradiction between encoding efficiency and pre-echo prevention.
2Quantity of substance
If quantization is applied during time domain to frequency domain conversion, then bit rate is reduced, but audio quality deteriorates due to quantization inaccuracies
Solution Approach 1:
The patent applies different quantization strategies to different signal regions. For transient-containing blocks, short-block transform coding with finer quantization is used to preserve precision where needed. For steady-state blocks, longer blocks with coarser quantization are used to achieve bit rate reduction. This local differentiation resolves the contradiction between bit rate reduction and audio quality preservation.
Solution Approach 2:
The patent changes quantization parameters dynamically based on signal characteristics and bit rate conditions. By adjusting quantization step sizes, block sizes, and coding modes according to the local signal properties and available bit rate, the system optimizes the trade-off between compression efficiency and audio quality.
3Stability of the object's composition
If fixed bit rate encoding is used, then audio quality consistency is maintained, but adaptability to dynamic network conditions is reduced
Solution Approach 1:
The patent employs dynamic bit rate allocation and variable block size strategies that adapt to network conditions while maintaining perceptual audio quality. The system can switch between different coding modes and adjust quantization parameters in real-time based on available bandwidth, providing both quality consistency through perceptual optimization and adaptability through dynamic parameter adjustment.
Solution Approach 2:
The patent incorporates feedback mechanisms that monitor both audio signal characteristics and network conditions, then adjust encoding parameters accordingly. This feedback loop enables the system to maintain audio quality consistency by learning from signal patterns while adapting to dynamic network conditions through real-time parameter adjustment.
Data Source
AI summary
An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events. The encoder operable to transform the error signal into high resolution frequency components using the MDCT block sizes, quantize the scale factors and frequency components, and encode the quantized lines, block sizes, and quantized scale factors for inclusion in the bitstream.