Audio Encoder Pre-Echo Detection Using Error Signal and MDCT Blocks
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Solution Overview
Problem
Existing audio codec technologies fail to effectively control or eliminate pre-echo artifacts, particularly 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 artifacts during 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 normal MDCT for most blocks and a different approach (such as time-domain processing or modified MDCT parameters) for blocks containing transients - the patent prevents error spreading that causes pre-echo while maintaining encoding efficiency for the majority of the signal.
Solution Approach 2:
The patent performs preliminary analysis to identify transient signals and potential pre-echo events before completing the MDCT encoding process. By detecting transient characteristics in advance and marking these events, the system can apply corrective measures during decoding or re-encoding, preventing the formation of pre-echo artifacts while maintaining the efficiency benefits of MDCT for non-transient portions.
2Quantity of substance
If quantization is applied during time domain to frequency domain conversion, then data compression is achieved, but audio quality deteriorates due to quantization errors
Solution Approach 1:
The patent applies different quantization strategies to different portions of the audio signal based on local characteristics. For blocks containing transient signals, the system uses higher quantization precision or alternative encoding methods to preserve audio quality, while for blocks with steady-state signals, more aggressive quantization is applied to achieve better compression. This local differentiation resolves the contradiction by adapting compression intensity to signal characteristics.
Solution Approach 2:
The patent dynamically adjusts quantization parameters based on the detected signal characteristics. When transient signals are identified, the system modifies quantization step sizes, block sizes, or other encoding parameters to minimize quantization errors that would otherwise degrade audio quality. For non-transient portions, standard quantization parameters are used to maintain compression efficiency.
3Reliability
If fixed bit rate encoding is used for traditional broadcast services, then transmission reliability is improved, but adaptability to distributed IP networks deteriorates
Solution Approach 1:
The patent introduces dynamic bit rate allocation and encoding parameter adjustment based on network conditions and signal characteristics. Instead of using fixed bit rate encoding, the system adapts encoding parameters such as bit rate, block size, and quantization precision in real-time based on detected transient events and network conditions. This dynamic approach maintains reliability for critical signal portions while providing adaptability to varying network conditions in distributed IP environments.
Solution Approach 2:
The patent creates a multi-functional encoding system that can operate in different modes depending on requirements. The same encoding framework can provide fixed bit rate operation for reliable broadcast applications while also supporting variable bit rate operation for adaptive IP network streaming. The system universally handles both traditional and modern distribution requirements through a single adaptable architecture that detects and responds to different operational contexts.
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.