Personal Hearing Amplifier Product

Overview

A personal hearing amplifier is a non-medical sound amplification system designed for casual hearing support—conversation assistance in noisy restaurants, lecture halls, or small group settings. Unlike a hearing aid, which is individually tuned and prescribed, an amplifier is a consumer device for ad-hoc temporary use. The system comprises a pocket-sized amplifier unit with a microphone, a wireless earpiece receiver, and a charging case.

The amplifier processes ambient sound in real-time: it captures audio from the environment through a directional microphone, applies DSP filtering to suppress background noise, amplifies speech frequencies, and transmits the processed stream to one or two wireless earpieces the user wears. All processing is analog and digital combined, with latency under 50 ms to avoid the unnatural "delay" sensation of hearing aids.

How it works

The Microphone Array consists of two MEMS elements in a small PCB array. The Microphone Preamp front-ends both mics with a low-noise amplifier and analog high-pass filter to reject wind noise and low-frequency rumble. The preamp output feeds the Audio Codec, which samples the audio at 16 kHz and quantizes to 16 bits.

The Processor Board's Microcontroller and Audio DSP IC run a real-time DSP pipeline. First, a noise gate detector measures the RMS energy in silence periods; if the current frame is below the gate threshold, its amplitude is scaled down. Second, a 3-band equalizer—typically parametric filters around 250 Hz (low), 1 kHz (mid), and 4 kHz (high)—can boost speech intelligibility by lifting the 1–4 kHz midrange where consonants live. Third, a compressor with adjustable knee and ratio prevents loud sounds (traffic, sirens) from clipping while preserving dynamic range.

The processed audio stream exits the Audio Codec's DAC and is buffered by the Audio Power Stage's Audio Power Amp IC, a Class-AB audio power amplifier. This stage can drive the earpieces to ~94 dB SPL (sound pressure level)—loud enough for restaurants but not so loud as to cause ear fatigue.

Simultaneously, the Microcontroller packetizes the processed audio samples and transmits them via the Wireless Link Module's BLE Transceiver. Each BLE audio packet contains 20 ms of audio and is timestamped; the earpiece buffers incoming packets and plays them back with a constant latency (typically 40–60 ms).

The Wireless Earpiece is a miniature receiver earbud with its own Earpiece BLE Module transceiver and tiny Earpiece Speaker driver (8–10 mm diameter). The earpiece decodes incoming BLE packets and immediately plays the audio through the speaker into the Ear Tips Set. There is no transmit link from the earpiece to the main unit; it is strictly receive-only, so no feedback or latency from the earbud back to the amplifier.

Power & Charging

The Battery Module in the main unit is a 3.7 V li-ion cell with a BMS Board for protection. The Charge Controller is a simple linear charger; USB power (typically 500 mA) trickle-charges the cell. The battery supplies power to the microphone preamp, DSP/MCU, codec, power amp, and BLE transceiver; total average consumption is ~200–300 mW during active listening.

The Earpiece Battery is a much smaller cell (50–100 mAh), as it only drives the speaker and low-power BLE receiver. With a 2 mW average draw, it can operate 4–8 hours on a single charge.

The Charging Case is a power bank and storage dock. It integrates a larger li-ion battery (typically 2000+ mAh) and a USB charger controller. Gold [[hearing-amplifier-charging-contacts|pogo pins]] dock the main unit and earpiece(s), and the case's charger IC detects each device, determines its battery state, and applies appropriate charge current (500 mA for the main unit, 100 mA for the tiny earpiece). A full case can recharge both devices 2–3 times before the case itself needs USB power.

Noise Management & Speech Enhancement

The Audio DSP Module must balance competing goals: amplify quiet speech without magnifying background noise, and avoid feedback if the earpiece speaker comes near the microphone. Real-time noise gating (via Audio DSP IC) detects speech bursts (high variance, rapid on-off) versus steady noise (constant hum), and applies more gain reduction to the latter.

In multi-speaker environments (parties, classrooms), the directional Microphone Array array can reject sound from behind, creating a forward-facing "listening beam." However, with only two mics spaced ~1 cm apart, the beamforming gain is modest (5–10 dB rejection of off-axis noise).

Compression prevents sudden loud events (dropped fork, door slam) from causing startling peaks. A typical compressor with a 2:1 ratio and 10 ms attack time will catch transients and reduce their peak by 3 dB relative to softer passages.

User Interaction

The Control Interface allow adjustment of gain (volume), EQ preset, and noise gate sensitivity. Most models have 5–10 dB volume steps and 3–5 EQ presets (e.g., "Restaurant," "Lecture," "Nature"). Mode cycling via the button changes the DSP program and resets the gate threshold and compressor state.