Audio Signal Synchronization Using Time Offset Estimation
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Solution Overview
Problem
Existing audio signal processing methods for multiple microphones struggle to effectively synchronize and combine audio signals from different sources, leading to suboptimal sound mixing and processing outcomes, especially in environments with spatial audio applications.
Innovation Solution
The method involves obtaining clean and processed signals from multiple microphones, determining timing information such as intra-channel and inter-channel time offsets, and using this information to synchronize the signals before further processing, such as spatial audio mixing, ensuring accurate alignment and combination of audio inputs from different microphones.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If audio signals from multiple microphones are processed without precise timing alignment, then processing complexity is reduced, but synchronization accuracy and sound mixing quality deteriorate
Solution Approach 1:
The patent performs preliminary timing offset estimation and compensation before audio signal mixing. By calculating intra-channel and inter-channel time offsets using correlation methods on the raw audio signals from multiple microphones, the system prepares synchronized signal streams in advance, which then can be mixed with high quality without requiring complex real-time synchronization during the mixing process itself.
Solution Approach 2:
The patent introduces timing offset parameters as intermediary variables that mediate between the raw audio signals from multiple microphones and the final mixed output. These time offset parameters are estimated separately and used to adjust the phase alignment of signals from different microphones, enabling precise synchronization without directly complicating the mixing operation.
2Reliability
If multiple audio signals from different microphones are combined without time offset compensation, then processing speed is maintained, but sound mixing quality and spatial audio accuracy deteriorate
Solution Approach 1:
The system performs time offset estimation and compensation in a preliminary stage before the actual audio mixing. By pre-calculating the timing relationships between signals from different microphones and applying corrections in advance, the patent ensures high sound mixing quality while keeping the main processing pipeline efficient and avoiding time losses during critical mixing operations.
Solution Approach 2:
The patent divides the audio processing into separate segments: timing offset estimation, time compensation, and signal mixing. This segmentation allows each function to be optimized independently - the timing analysis can be performed on short reference segments, and the corrected signals can then be mixed efficiently, reducing overall processing time while maintaining quality.
3Measurement precision
If intra-channel and inter-channel time offsets are calculated and applied, then signal alignment accuracy is improved, but computational complexity increases
Solution Approach 1:
The patent uses time offset parameters as intermediary variables that separate the complex task of signal alignment from the mixing process. By estimating intra-channel and inter-channel time offsets as distinct intermediate steps and applying them as phase corrections, the system achieves precise signal alignment without embedding complex synchronization logic throughout the entire processing chain.
Solution Approach 2:
The patent creates simplified copies or representations of the timing relationships through correlation functions and time offset estimates. Instead of directly manipulating the full complexity of multi-microphone signal relationships, the system works with condensed timing parameter representations that capture the essential alignment information needed for accurate signal mixing.
Data Source
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Figure 3A~3C
AI summary
A method comprising: obtaining a first clean signal (S1) and a first processed signal (S4) dependent upon audio input (24) at a first microphone (25); obtaining a second clean signal (S2) and a second processed signal (S3) dependent upon audio input (26) at a second microphone (27); using the first clean signal and the second clean signal to enable further processing (37) of at least the first processed signal and the second processed signal.