Audio Harmonic Reinjection for Low-Frequency Restoration
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Solution Overview
Problem
Existing methods for reconstituting low frequencies in audio signals, such as those used in sound playback devices, often result in non-linear processing, intermodulation artifacts, and phase shifting, failing to accurately preserve the nuances and dynamic range of the original signal, particularly due to variations in human hearing perception with frequency.
Innovation Solution
A method that filters the audio signal using a lowpass filter to detect fundamental frequencies, generates harmonics, and adapts the dynamic range of the time envelope to reinject the harmonic signal in phase, utilizing a compression/expansion system with a feedback loop to dynamically adjust the offset and ensure the signal's energy is maximized within defined limits, thereby maintaining the signal's natural tone and dynamic range.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Loss of information
If a filter is applied to amplify low frequencies attenuated by the loudspeaker, then the low frequency content is restored, but the diaphragm excursion becomes too great and the loudspeaker is damaged
Solution Approach 1:
The patent replaces the mechanical approach of forcing the loudspeaker diaphragm to radiate low frequencies with an electronic signal processing approach. By detecting fundamental frequencies and synthesizing harmonic signals that exploit the missing fundamental effect, the system restores low frequency perception without requiring the physical loudspeaker to reproduce those frequencies, thereby avoiding diaphragm damage.
Solution Approach 2:
The patent changes the parameter space by working in the frequency domain through harmonic synthesis rather than in the time domain through direct signal amplification. By generating harmonic signals at higher frequencies that perceive as the fundamental frequency through the missing fundamental effect, the system achieves low frequency restoration without the physical constraints of the loudspeaker's frequency response and mechanical limits.
2Loss of information
If non-linear filtering is used to generate harmonics, then higher harmonics are produced, but intermodulation artifacts degrade audio performance
Solution Approach 1:
The patent extracts only the fundamental frequency information from the audio signal using linear filtering and zero-crossing detection, then selectively synthesizes only the necessary harmonic components. This approach avoids the intermodulation artifacts that occur in non-linear systems by not processing the entire signal through non-linear operations, thereby maintaining audio fidelity while still generating the required harmonic content.
3Difficulty of detecting and measuring
If lowpass filtering is applied to detect fundamental frequencies, then frequency detection is simplified, but phase shifting occurs and harmonic reinjection is out of phase
Solution Approach 1:
The patent transitions from working solely in the time domain to incorporating frequency domain analysis. By using zero-crossing detection on the lowpass filtered signal to determine fundamental frequencies, then synthesizing harmonics in the frequency domain and reinjecting them with proper phase alignment, the system overcomes the phase shifting problem while maintaining the simplicity of lowpass filtering for detection.
4Loss of information
If the audio signal is amplified to compensate for low frequency attenuation, then deep sounds are restored, but the loudspeaker diaphragm excursion becomes excessive
Solution Approach 1:
The patent replaces the mechanical amplification approach with an electronic harmonic synthesis approach. Instead of amplifying the fundamental low frequency signal and forcing the diaphragm to move extensively, the system synthesizes harmonic signals at higher frequencies that the loudspeaker can reproduce with minimal diaphragm excursion, while the human ear perceives the fundamental frequency through the missing fundamental effect.
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 approach effectively extends the passband of sound playback devices to restore low frequencies, ensuring the signal's nuances and tone color are preserved, providing a natural sound reproduction that accounts for human hearing variations, without introducing distortion or phase issues.
Implementation Method 1
filtering the audio signal by means of a lowpass filter with a cutoff frequency substantially equal to said cutoff frequency of the sound playback device
Implementation Method 2
adapting the dynamic range of said time envelope as a function of the frequency band under consideration
Data Source
AI summary
The method comprises the steps of: filtering the audio signal by means of a lowpass filter (101) with a cutoff frequency substantially equal to said cutoff frequency (F0) of the sound playback device; determining a fundamental frequency for reconstituting from the lowpass filtered audio signal; and generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted. It also comprises the steps of: detecting a time envelope (env(t)) of the lowpass filtered audio signal; adapting the dynamic range of said time envelope (env(t)) as a function of the frequency band under consideration; and reinjecting said harmonic signal in phase into said audio signal by addition after multiplying said harmonic signal (Sharm) with the adapted time envelope (envadapt(t)). The adaptation is performed by compression/expansion of the time envelope with feedback loop control that is adjusted automatically on the value of the envelope as a function of the mean energy of the input signal to a value that maximizes said energy within a defined limit.