Dual-Microphone ADC Data Multiplexing for Lower DSP Load
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Solution Overview
Problem
Existing analog-to-digital converters in sound processing devices face inefficiencies in processing load and circuit integration due to suboptimal design, which complicates the conversion of analog audio signals to digital signals.
Innovation Solution
The proposed solution involves an analog-to-digital converter with a toggle detection module, pre-amplifier, and 1-bit ΔΣ modulator, which receives a mode signal, channel selection signal, and clock signal to efficiently convert audio signals, and a signal coding module that encodes data bits based on the channel selection signal's logic level, simplifying output data handling.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If conventional analog-to-digital converters are used in sound processing devices, then the conversion function is provided, but the processing load on digital signal processors increases and circuit integration becomes complex
Solution Approach 1:
The patent segments the analog-to-digital conversion process into distinct functional modules: a pre-amplifier stage for signal conditioning, a ΔΣ modulator for quantization and oversampling, and a digital filter for decimation. This modular segmentation allows each component to be optimized independently, reducing overall circuit complexity while maintaining high conversion efficiency.
Solution Approach 2:
The patent introduces an intermediate digital signal processing stage between the analog input and final digital output. The ΔΣ modulator acts as an intermediary that converts the analog signal to a high-rate digital signal, which is then processed by digital filters. This intermediary stage simplifies the final conversion stage and reduces the processing burden on downstream digital signal processors.
2Measurement precision
If the analog-to-digital converter processes all audio signals through complex encoding, then signal quality is maintained, but the processing load increases
Solution Approach 1:
The patent applies partial encoding by selectively processing only the necessary signal components through the full encoding chain. The ΔΣ modulator performs oversampling and quantization on the full-band signal, but subsequent digital filtering removes frequencies outside the audio band before final encoding. This partial processing approach maintains signal accuracy for audible frequencies while reducing unnecessary processing overhead.
Solution Approach 2:
The patent dynamically adjusts processing parameters based on signal characteristics. The digital filter cutoff frequencies and decimation ratios are optimized according to the input signal bandwidth and desired output sample rate. This parameter optimization allows the system to maintain high conversion accuracy while adapting processing throughput to match actual signal requirements.
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 design reduces the processing load on digital signal processors, simplifies circuit integration, and enhances the efficiency of analog-to-digital conversion by effectively managing data output based on clock signal oscillations, improving overall sound processing performance.
Implementation Method 1
The 1-bit ΔΣ modulator converts the second audio signal from analog to digital to obtain a third audio signal
Data Source
AI summary
The invention provides a sound processing device. In one embodiment, the sound processing device comprises a first microphone, a first analog-to-digital converter, a second microphone, and a second analog-to-digital converter. The first microphone detects a first sound pressure to generate a first analog audio signal. The first analog-to-digital converter converts the first analog audio signal to a first digital audio signal. The second microphone detects a second sound pressure to generate a second analog audio signal. The second analog-to-digital converter converts the second analog audio signal to a second digital audio signal, encodes a third digital audio signal according to the second digital audio signal, receives the first digital audio signal and a clock signal, outputs data bits of the third digital audio signal when the clock signal oscillates to a logic low level, and outputs data bits of the first digital audio signal when the clock signal oscillates to a logic high level.