Live Event Headphone Audio Synchronization With Adaptive Filtering
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Solution Overview
Problem
Audiences at live events often struggle to clearly hear individual instruments or voices amidst ambient noise, as existing sound enhancement systems cannot accommodate individual preferences and are limited by microphone placement and conventional noise cancellation methods.
Innovation Solution
An audio processing system that uses a wireless interface to receive and adaptively filter audio signals, compensating for propagation time differences between sound emitted by loudspeakers and received via headphones, allowing for personalized sound enhancement by delaying and modifying the audio signals to improve sound clarity and reduce ambient noise.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If audio signals are transmitted wirelessly to headphones for sound enhancement at live events, then individual hearing impression and sound clarity are improved, but propagation time difference between sound emitted by loudspeakers and received via headphones causes synchronization issues and disturbs the hearing impression
Solution Approach 1:
The patent applies preliminary action by determining the propagation time difference between sound emitted by loudspeakers and received via headphones before processing the audio signal. The system calculates this time difference in advance and uses it to delay the audio signal played through headphones, ensuring synchronous playback with the acoustic signal from loudspeakers. This preliminary timing adjustment prevents synchronization issues and maintains accurate hearing impression.
2Stability of the object's composition
If the second audio signal is delayed to compensate for propagation time difference, then synchronization with acoustic signal is improved, but the delay processing increases computational complexity and processing time
Solution Approach 1:
The system applies self-service by automatically determining the propagation time difference between the acoustic signal from loudspeakers and the wireless audio signal received by headphones. The device autonomously calculates the time difference, applies the appropriate delay to the audio signal, and ensures synchronization without requiring manual intervention or complex external processing systems.
3Object-affected harmful factors
If adaptive filtering is applied to model acoustic transmission, then noise reduction and sound quality are improved, but the filtering process increases computational load and processing time
Solution Approach 1:
The patent implements feedback by using the determined propagation time difference and acoustic signal characteristics to continuously adjust the adaptive filtering parameters. The system processes the audio signal through the adaptive filter, monitors the resulting hearing impression, and refines the filtering to optimize noise reduction while maintaining processing efficiency. This feedback mechanism ensures high sound quality without excessive computational load.
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
This system enables individualized audio processing that enhances sound clarity and reduces ambient noise, allowing listeners to better hear specific instruments or voices while maintaining ambient sound awareness, even in noisy environments like rock concerts.
Implementation Method 1
an audio mixer or mixing console receives an audio signal that may have two or more audio tracks, mixes and transmits this signal as a second audio signal via a wireless interface, in particular a WLAN interface or mobile network interface
Implementation Method 2
a first audio signal is received. The first audio signal is herein preferably a sound signal that is converted into an electrical audio signal by one or more audio sensors, in particular microphones
Implementation Method 3
a propagation time difference between the first and the second audio signal is determined, and the second audio signal is adaptively filtered and delayed, based on the determined propagation time difference
Implementation Method 4
this signal is provided to one or more loudspeakers, such as e.g. a public address system, which emits a corresponding sound signal
Data Source
AI summary
A method for generating and providing an audio signal, including receiving a first audio signal via an external microphone of a headphone or earphone, and receiving a second audio signal via a wireless interface. The first audio signal includes a portion reproduced via loudspeakers. The second audio signal corresponds to the portion reproduced via loudspeakers and is received before the corresponding portion of the first audio signal. A propagation time difference is determined between the first audio signal and the second audio signal. The second audio signal is modified by adaptive filtering and temporal shifting such that the propagation time difference between the first and second modified audio signal is substantially compensated. The adaptive filtering models an acoustic transmission of the first audio signal and a modified second audio signal is obtained. The modified second audio signal is inverted, then it is provided via the headphone or earphone.