Project

# Title Team Members TA Documents Sponsor
88 Catching Z's
Prineet Parhar
Srikar Palani
Suprathik Vinayakula
Zhuchen Shao proposal1.pdf
# Title
**Catching Z’s**

## Team Members
- Suprathik Vinayakula (sv53)
- Srikar Palani (palani3)
- Prineet Parhar (pparhar2)

## Problem
Sudden environmental noises such as sirens, loud neighbors, barking dogs, or door slams are a primary cause of sleep fragmentation, which negatively impacts cognitive performance and long-term health. Conventional white noise machines operate continuously at a fixed volume, which can be unnecessary or ineffective against short, intermittent disturbances. There is a need for a smart bedside system that continuously monitors room acoustics and activates noise masking only when disruptive sounds occur, while remaining off during quiet periods.

## Solution
We propose **Catching Z’s**, a bedside embedded system that monitors ambient audio in real time and adaptively generates masking noise in response to disruptive sound events. Using a high-sensitivity microphone and onboard signal processing, the system establishes a baseline ambient noise profile and detects sudden sound spikes based on amplitude and frequency characteristics. When a disturbance is detected, Catching Z’s smoothly fades in white, pink, or brown noise to mask the event, then gradually fades out once the environment returns to baseline. This adaptive response minimizes unnecessary noise while preventing the masking system itself from waking the user.

## Solution Components

### Acoustic Sensing Subsystem
This subsystem continuously monitors the ambient sound environment.
- **Microphone Module:** Electret microphone with pre-amplifier (MAX4466) to capture low-level room noise with sufficient gain and low distortion.
- **Analog-to-Digital Conversion:** The ESP32-S3’s built-in ADC samples the microphone signal at 10–20 kHz for envelope and spectral analysis.

### Processing and Audio Output Subsystem
This subsystem performs sound analysis and generates masking audio.
- **Microcontroller:** ESP32-S3-WROOM-1, selected for dual-core operation, allowing one core to handle real-time audio sensing while the other manages audio synthesis and playback.
- **Audio Amplifier / DAC:** I2S Class-D amplifier (MAX98357A) for efficient digital-to-audio conversion and speaker drive.
- **Speaker:** 4 Ω, 3 W full-range speaker (50 mm) for producing broadband masking noise.

### User Interface and Power Subsystem
This subsystem provides user control and power regulation.
- **User Input:** Rotary encoder (PEC11R-4215F-S0024) to adjust detection sensitivity and masking intensity thresholds.
- **Power:** 5 V USB-C input with on-board regulation to 3.3 V using an AMS1117-3.3 LDO regulator.
- **Indicators:** Status LEDs to indicate detection events and system state.

## Criterion for Success
1. **Detection Latency:** The system shall trigger masking noise playback within **100 ms** of detecting a sound event exceeding the ambient baseline by **≥ 10 dB**.
2. **Output Capability:** The audio subsystem shall produce masking noise over a controllable range of **40 dB to 75 dB SPL** at the bedside.
3. **Continuous Operation:** The system shall operate continuously for overnight use without performance degradation or audible artifacts.

## Risks and Mitigation
- **Overreaction to brief harmless sounds:** Mitigated by minimum-duration thresholds.
- **Environmental variability:** Adaptive baseline recalibration during extended quiet periods.

Master Bus Processor

Clay Kaiser, Philip Macias, Richard Mannion

Master Bus Processor

Featured Project

General Description

We will design a Master Bus Processor (MBP) for music production in home studios. The MBP will use a hybrid analog/digital approach to provide both the desirable non-linearities of analog processing and the flexibility of digital control. Our design will be less costly than other audio bus processors so that it is more accessible to our target market of home studio owners. The MBP will be unique in its low cost as well as in its incorporation of a digital hardware control system. This allows for more flexibility and more intuitive controls when compared to other products on the market.

Design Proposal

Our design would contain a core functionality with scalability in added functionality. It would be designed to fit in a 2U rack mount enclosure with distinct boards for digital and analog circuits to allow for easier unit testings and account for digital/analog interference.

The audio processing signal chain would be composed of analog processing 'blocks’--like steps in the signal chain.

The basic analog blocks we would integrate are:

Compressor/limiter modes

EQ with shelf/bell modes

Saturation with symmetrical/asymmetrical modes

Each block’s multiple modes would be controlled by a digital circuit to allow for intuitive mode selection.

The digital circuit will be responsible for:

Mode selection

Analog block sequence

DSP feedback and monitoring of each analog block (REACH GOAL)

The digital circuit will entail a series of buttons to allow the user to easily select which analog block to control and another button to allow the user to scroll between different modes and presets. Another button will allow the user to control sequence of the analog blocks. An LCD display will be used to give the user feedback of the current state of the system when scrolling and selecting particular modes.

Reach Goals

added DSP functionality such as monitoring of the analog functions

Replace Arduino boards for DSP with custom digital control boards using ATmega328 microcontrollers (same as arduino board)

Rack mounted enclosure/marketable design

System Verification

We will qualify the success of the project by how closely its processing performance matches the design intent. Since audio 'quality’ can be highly subjective, we will rely on objective metrics such as Gain Reduction (GR [dB]), Total Harmonic Distortion (THD [%]), and Noise [V] to qualify the analog processing blocks. The digital controls will be qualified by their ability to actuate the correct analog blocks consistently without causing disruptions to the signal chain or interference. Additionally, the hardware user interface will be qualified by ease of use and intuitiveness.

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