Echo Cancellation Using Beamforming Nulling and Acoustic Delay Compensation
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Conventional echo cancellation methods in home theater systems with multiple loudspeakers fail to effectively reduce echo due to increased delays between near-end and far-end signals, which conventional adaptive filters cannot handle, leading to ineffective echo suppression.
Innovation Solution
An audio processing device is configured to operate in a calibration mode to determine acoustic delays and direction of arrival (DOA) for each loudspeaker, using beamforming to null out specific audio paths and apply tunable delays to improve echo cancellation, especially in home theater systems with multiple speakers.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Illumination intensity
If multiple loudspeakers are used in home theater systems, then audio quality and surround sound capability are improved, but echo generation increases and conventional echo cancellation methods fail
Solution Approach 1:
The patent segments the echo cancellation process into two distinct stages: calibration mode and runtime mode. During calibration mode, the system individually characterizes each loudspeaker's acoustic path to the microphone, storing impulse responses and delay values. At runtime, it mixes these pre-characterized paths and applies adaptive filtering. This segmentation allows the system to handle multiple loudspeakers effectively by processing their acoustic contributions separately during calibration and then combining them during operation, resolving the echo problem without compromising audio quality.
Solution Approach 2:
The system performs preliminary acoustic characterization of each loudspeaker during calibration mode before actual audio processing. It determines impulse responses, delays, and spatial positions of loudspeakers in advance, storing this information for use during runtime. This preliminary action enables the runtime echo canceller to efficiently compensate for acoustic delays and null out loudspeaker signals without real-time complexity, allowing effective echo cancellation in multi-loudspeaker configurations.
2Device complexity
If conventional adaptive filters are used for echo cancellation, then simplicity is maintained, but they cannot handle increased delays between near-end and far-end signals in multi-loudspeaker systems
Solution Approach 1:
The system performs preliminary acoustic characterization of each loudspeaker during calibration mode, determining impulse responses, delays, and spatial positions in advance. This pre-computed information is stored and reused during runtime, allowing the adaptive filter to focus only on coefficient adaptation rather than recalculating acoustic paths. This approach maintains filter simplicity while significantly improving reliability by providing accurate delay compensation for multi-loudspeaker configurations.
Solution Approach 2:
The patent introduces an intermediary calibration process that acts as a bridge between the physical acoustic environment and the digital signal processing. The calibration mode measures and stores acoustic path characteristics, serving as an intermediary layer that translates complex acoustic delays into manageable filter parameters. This intermediary step enables conventional adaptive filters to effectively handle multi-loudspeaker systems without requiring complex real-time acoustic modeling.
3Measurement precision
If acoustic delays and DOA determination are performed during calibration, then echo cancellation accuracy is improved, but processing time and system complexity increase
Solution Approach 1:
The system performs acoustic delay and DOA determination during an initial calibration phase, storing these parameters for reuse during runtime audio processing. This preliminary measurement of acoustic characteristics (impulse responses, delays, spatial positions) is conducted once during setup rather than continuously during operation. The calibration time is traded for significantly improved echo cancellation accuracy during actual use, as the pre-computed parameters enable precise delay compensation without recurring processing overhead.
Solution Approach 2:
The calibration process is designed as a periodic or one-time operation that establishes acoustic parameters, after which the system operates in runtime mode using these pre-determined values. The calibration can be repeated if environmental changes occur, but normally serves as a setup phase. This periodic approach concentrates the time-consuming measurements into discrete calibration events rather than continuous processing, improving real-time performance while maintaining measurement precision when needed.
Applied Scientific Principles
This section explains which scientific principles are used to turn an abstract innovation direction into a practical engineering solution.
Function Achieved in This Case
The solution effectively reduces echo in home theater systems by accurately determining and compensating for acoustic delays and DOAs, enabling successful echo cancellation even in environments where conventional methods fail, thereby enhancing audio quality during conferencing and other audio applications.
Implementation Method 1
using beamforming to null out specific audio paths
Implementation Method 2
apply tunable delays to improve echo cancellation
Data Source
Figure 1
Figure 2
Figure 3
AI summary
A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.