Frontend Audio Capture Gain Staging for Consistent Conferencing Input

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Solution Overview

Problem

Video conferencing systems face challenges in maintaining optimal audio signal quality due to variations in hardware specifications between client devices and server expectations, particularly with third-party video conferencing devices providing lower signal levels that need to be adjusted to meet server requirements, often resulting in suboptimal audio transmission.

Innovation Solution

The implementation of a frontend capture module (FCM) within the video conferencing application that includes an acoustic echo cancellation module, noise estimation module, input stage, suppression module, and output stage to adjust and process audio signals, ensuring they meet server specifications by amplifying low signal levels, compressing high signal levels, and maintaining a consistent noise floor.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Ease of operation

If third-party video conferencing devices are used to provide audio input, then device compatibility and ease of operation are improved, but audio signal level and quality deteriorate because these devices provide lower signal levels that do not meet server requirements

Engineering Contradiction:
Improvedevice compatibilityVSAvoidaudio signal level
Core Design Contradiction:
Ease of operationVSManufacturing precision

Solution Approach 1:

The frontend capture module acts as an intermediary between third-party video conferencing devices and the server. It receives audio input from devices with varying signal levels, processes the signal through amplification and compression stages, and outputs a standardized signal that meets server requirements. This mediator approach allows compatibility with diverse third-party devices while ensuring consistent audio quality at the server端.

Inventive Principle:
Principle #24Intermediary (Mediator)

Solution Approach 2:

The system dynamically changes audio signal parameters including gain levels, compression ratios, and noise floor thresholds. The input stage amplifies low signal levels, the compressor adjusts dynamic range, and the noise floor is maintained at optimal levels. These parameter adjustments transform incompatible audio signals from third-party devices into server-compliant signals.

Inventive Principle:
Principle #35Parameter changes

2Manufacturing precision

If signal amplification is applied to low signal levels, then audio quality is improved, but noise amplification and distortion occur

Engineering Contradiction:
Improveaudio signal qualityVSAvoidnoise and distortion
Core Design Contradiction:
Manufacturing precisionVSObject-generated harmful factors

Solution Approach 1:

The system performs preliminary noise estimation and characterization before signal amplification. The noise estimation module analyzes the audio signal to determine noise floor levels and characteristics in advance. This preliminary action allows the amplification stage to selectively amplify desired audio signals while minimizing noise amplification, and enables the compressor to prevent distortion by pre-establishing appropriate compression thresholds.

Inventive Principle:
Principle #10Preliminary action

Solution Approach 2:

The system implements feedback mechanisms where the output of each processing stage is monitored and used to adjust subsequent processing. The noise estimation feedback allows dynamic adjustment of gain and compression parameters. The compressor uses feedback from signal level monitoring to maintain output within optimal ranges, preventing distortion while preserving audio quality.

Inventive Principle:
Principle #23Feedback

3Reliability

If compression is applied to high signal levels, then clipping is prevented, but audio quality and dynamic range are reduced

Engineering Contradiction:
Improveclipping preventionVSAvoidaudio quality
Core Design Contradiction:
ReliabilityVSManufacturing precision

Solution Approach 1:

The compression parameters are made dynamic rather than static. The compression ratio, threshold, and attack/release times are adjusted in real-time based on signal characteristics and noise floor levels. This dynamic compression approach prevents clipping during high signal levels while maintaining better audio quality and dynamic range compared to fixed compression, as the compressor engages only when necessary and with optimized parameters for each situation.

Inventive Principle:
Principle #15Dynamics

Data Source

PatentUS11837254B2Frontend capture with input stage, suppression module, and output stage
Publication Date: 2023.12.05 ZOOM COMMUNICATIONS INC
  • US11837254B2 patent drawing
  • US11837254B2 patent drawing
  • US11837254B2 patent drawing

AI summary

Disclosed are systems and methods for a frontend capture module of a video conferencing application, which can modify an input signal, received from a microphone device to match predetermined signal characteristics, such as voice signal level and expected noise floor. An Input stage, a suppression module and an output stage amplify the voice signal portion of the input signal and suppress the noise signal of input signal to predetermined ranges. The input stage selectively applies gains defined by a gain table, based on signal level of the input signal. The suppression module selectively applies a suppression gain to the input signal based on presence or absence of voice signal in the input signal. The output stage further amplifies the input signal in portions having a voice signal and applies a gain table to maintain a consistent noise floor.