feat ✨: Improved visualization by adding logarithmic scaling to the x-axis labels and updating the y-axis scale based on the maximum value of the frequency spectrum.

- Updated the `y-axis scale` based on the maximum value of the frequency spectrum, added logarithmic scaling to the x-axis labels, and improved interpolation logic for better display.

README.md

@@ -1,2 +1,317 @@
-# esp32i2s
+# ESP32 Piano Note Detection System
+
+A real-time piano note detection system implemented on ESP32 using I2S microphone input. This system can detect musical notes from C2 to C6 with adjustable sensitivity and visualization options.
+
+## Features
+
+- Real-time audio processing using I2S microphone
+- FFT-based frequency analysis
+- Note detection from C2 (65.41 Hz) to C6 (1046.50 Hz)
+- Dynamic threshold calibration
+- Multiple note detection (up to 7 simultaneous notes)
+- Harmonic filtering
+- Real-time spectrum visualization
+- Note timing and duration tracking
+- Interactive Serial commands for system tuning
+
+## Hardware Requirements
+
+- ESP32 development board
+- I2S MEMS microphone (e.g., INMP441, SPH0645)
+- USB connection for Serial monitoring
+
+## Pin Configuration
+
+The system uses the following I2S pins by default (configurable in Config.h):
+- SCK (Serial Clock): GPIO 8
+- WS/LRC (Word Select/Left-Right Clock): GPIO 9
+- SD (Serial Data): GPIO 10
+
+## Getting Started
+
+1. Connect the I2S microphone to the ESP32 according to the pin configuration
+2. Build and flash the project to your ESP32
+3. Open a Serial monitor at 115200 baud
+4. Follow the calibration process on first run
+
+## Serial Commands
+
+The system can be controlled via Serial commands:
+
+- `h` - Display help menu
+- `c` - Start calibration process
+- `+` - Increase sensitivity (threshold up)
+- `-` - Decrease sensitivity (threshold down)
+- `s` - Toggle spectrum visualization
+
+## Configuration Options
+
+All system parameters can be adjusted in `Config.h`:
+
+### Audio Processing
+- `SAMPLE_RATE`: 8000 Hz (good for frequencies up to 4kHz)
+- `BITS_PER_SAMPLE`: 16-bit resolution
+- `SAMPLE_BUFFER_SIZE`: 1024 samples
+- `FFT_SIZE`: 1024 points
+
+### Note Detection
+- `NOTE_FREQ_C2`: 65.41 Hz (lowest detectable note)
+- `NOTE_FREQ_C6`: 1046.50 Hz (highest detectable note)
+- `FREQUENCY_TOLERANCE`: 3.0 Hz
+- `MAX_SIMULTANEOUS_NOTES`: 7
+- `MIN_NOTE_DURATION_MS`: 50ms
+- `NOTE_RELEASE_TIME_MS`: 100ms
+
+### Calibration
+- `CALIBRATION_DURATION_MS`: 5000ms
+- `CALIBRATION_PEAK_PERCENTILE`: 0.95 (95th percentile)
+
+## Visualization
+
+The system provides two visualization modes:
+
+1. Note Display:
+```
+Current Notes:
+A4 (440.0 Hz, Magnitude: 2500, Duration: 250ms)
+E5 (659.3 Hz, Magnitude: 1800, Duration: 150ms)
+```
+
+2. Spectrum Display (when enabled):
+```
+Frequency Spectrum:
+0Hz    |▄▄▄▄▄
+100Hz  |██████▄
+200Hz  |▄▄▄
+...
+```
+
+## Performance Tuning
+
+1. Start with calibration by pressing 'c' in a quiet environment
+2. Play notes and observe the detection accuracy
+3. Use '+' and '-' to adjust sensitivity if needed
+4. Enable spectrum display with 's' to visualize frequency content
+5. Adjust `Config.h` parameters if needed for your specific setup
+
+## Implementation Details
+
+- Uses FFT for frequency analysis
+- Implements peak detection with dynamic thresholding
+- Filters out harmonics to prevent duplicate detections
+- Tracks note timing and duration
+- Uses ring buffer for real-time processing
+- Calibration collects ambient noise profile
+
+## Troubleshooting
+
+1. No notes detected:
+   - Check microphone connection
+   - Run calibration
+   - Increase sensitivity with '+'
+   - Verify audio input level in spectrum display
+
+2. False detections:
+   - Run calibration in a quiet environment
+   - Decrease sensitivity with '-'
+   - Adjust `PEAK_RATIO_THRESHOLD` in Config.h
+
+3. Missing notes:
+   - Check if notes are within C2-C6 range
+   - Increase `FREQUENCY_TOLERANCE`
+   - Decrease `MIN_MAGNITUDE_THRESHOLD`
+
+## Contributing
+
+Contributions are welcome! Please read the contributing guidelines before submitting pull requests.
+
+## License
+
+This project is licensed under the MIT License - see the LICENSE file for details.
+
+## Development Environment Setup
+
+### Prerequisites
+- PlatformIO IDE (recommended) or Arduino IDE
+- ESP32 board support package
+- Required libraries:
+  - arduino-audio-tools
+  - arduino-audio-driver
+  - WiFiManager
+  - AsyncTCP
+  - ESPAsyncWebServer
+  - arduinoFFT
+
+### Building with PlatformIO
+1. Clone the repository
+2. Open the project in PlatformIO
+3. Install dependencies:
+   ```
+   pio lib install
+   ```
+4. Build and upload:
+   ```
+   pio run -t upload
+   ```
+
+## Memory Management
+
+### Memory Usage
+- Program Memory: ~800KB
+- RAM Usage: ~100KB
+- DMA Buffers: 4 x 512 bytes
+- FFT Working Buffer: 2048 bytes (1024 samples x 2 bytes)
+
+### Optimization Tips
+- Adjust `DMA_BUFFER_COUNT` based on available RAM
+- Reduce `SAMPLE_BUFFER_SIZE` for lower latency
+- Use `PSRAM` if available for larger buffer sizes
+
+## Advanced Configuration
+
+### Task Management
+- Audio processing task on Core 1:
+  - I2S sample reading
+  - Audio level tracking
+  - Note detection and FFT analysis
+- Visualization task on Core 0:
+  - WebSocket communication
+  - Spectrum visualization
+  - Serial interface
+  - Network operations
+- Inter-core communication via FreeRTOS queue
+- Configurable priorities in `Config.h`
+
+### Audio Pipeline
+1. I2S DMA Input
+2. Sample Buffer Collection
+3. FFT Processing
+4. Peak Detection
+5. Note Identification
+6. Output Generation
+
+### Timing Parameters
+- Audio Buffer Processing: ~8ms
+- FFT Computation: ~5ms
+- Note Detection: ~2ms
+- Total Latency: ~15-20ms
+
+## Performance Optimization
+
+### CPU Usage
+- Core 1 (Audio Processing):
+  - I2S DMA handling: ~15%
+  - Audio analysis: ~20%
+  - FFT processing: ~15%
+- Core 0 (Visualization):
+  - WebSocket updates: ~5%
+  - Visualization: ~5%
+  - Network handling: ~5%
+
+### Memory Optimization
+1. Buffer Size Selection:
+   - Larger buffers: Better frequency resolution
+   - Smaller buffers: Lower latency
+2. DMA Configuration:
+   - More buffers: Better continuity
+   - Fewer buffers: Lower memory usage
+
+### Frequency Analysis
+- FFT Resolution: 7.8125 Hz (8000/1024)
+- Frequency Bins: 512 (Nyquist limit)
+- Useful Range: 65.41 Hz to 1046.50 Hz
+- Window Function: Hamming
+
+## Technical Details
+
+### Microphone Specifications
+- Supply Voltage: 3.3V
+- Sampling Rate: 8kHz
+- Bit Depth: 16-bit
+- SNR: >65dB (typical)
+
+### Signal Processing
+1. Pre-processing:
+   - DC offset removal
+   - Windowing function application
+2. FFT Processing:
+   - 1024-point real FFT
+   - Magnitude calculation
+3. Post-processing:
+   - Peak detection
+   - Harmonic filtering
+   - Note matching
+
+### Calibration Process
+1. Ambient Noise Collection (5 seconds)
+2. Frequency Bin Analysis
+3. Threshold Calculation:
+   - Base threshold from 95th percentile
+   - Per-bin noise floor mapping
+4. Dynamic Adjustment
+
+## Error Handling
+
+### Common Issues
+1. I2S Communication Errors:
+   - Check pin connections
+   - Verify I2S configuration
+   - Monitor serial output for error codes
+2. Memory Issues:
+   - Watch heap fragmentation
+   - Monitor stack usage
+   - Check DMA buffer allocation
+
+### Error Recovery
+- Automatic I2S reset on communication errors
+- Dynamic threshold adjustment
+- Watchdog timer protection
+
+## Project Structure
+
+### Core Components
+1. AudioLevelTracker
+   - Real-time audio level monitoring
+   - Peak detection
+   - Threshold management
+2. NoteDetector
+   - Frequency analysis
+   - Note identification
+   - Harmonic filtering
+3. SpectrumVisualizer
+   - Real-time spectrum display
+   - Magnitude scaling
+   - ASCII visualization
+
+### File Organization
+- `/src`: Core implementation files
+- `/include`: Header files and configurations
+- `/data`: Additional resources
+- `/test`: Unit tests
+
+## Inter-Core Communication
+
+### Queue Management
+- FreeRTOS queue for audio data transfer
+- 4-slot queue buffer
+- Zero-copy data passing
+- Non-blocking queue operations
+- Automatic overflow protection
+
+### Data Flow
+1. Core 1 (Audio Task):
+   - Processes audio samples
+   - Performs FFT analysis
+   - Queues processed data
+2. Core 0 (Visualization Task):
+   - Receives processed data
+   - Updates visualization
+   - Handles network communication
+
+### Network Communication
+- Asynchronous WebSocket updates
+- JSON-formatted spectrum data
+- Configurable update rate (50ms default)
+- Automatic client cleanup
+- Efficient connection management
 

data/index.html

@@ -0,0 +1,260 @@
+<!DOCTYPE html>
+<html>
+<head>
+    <title>ESP32 Piano Spectrum Analyzer</title>
+    <script src="https://cdn.jsdelivr.net/npm/chart.js"></script>
+    <script src="https://cdn.jsdelivr.net/npm/chartjs-plugin-annotation"></script>
+    <style>
+        body {
+            font-family: Arial, sans-serif;
+            margin: 20px;
+            background: #f0f0f0;
+        }
+        .container {
+            max-width: 1200px;
+            margin: 0 auto;
+            background: white;
+            padding: 20px;
+            border-radius: 10px;
+            box-shadow: 0 0 10px rgba(0,0,0,0.1);
+        }
+        h1 {
+            color: #333;
+            text-align: center;
+        }
+        #spectrum-container {
+            position: relative;
+            height: 400px;
+            margin: 20px 0;
+        }
+    </style>
+</head>
+<body>
+    <div class="container">
+        <h1>Piano Spectrum Analyzer</h1>
+        <div id="spectrum-container">
+            <canvas id="spectrumChart"></canvas>
+        </div>
+    </div>
+
+    <script>
+        Chart.register('chartjs-plugin-annotation');
+        
+        // Piano note frequencies (Hz)
+        const noteFrequencies = {
+            'C2': 65.41, 'C#2': 69.30, 'D2': 73.42, 'D#2': 77.78, 'E2': 82.41, 'F2': 87.31, 
+            'F#2': 92.50, 'G2': 98.00, 'G#2': 103.83, 'A2': 110.00, 'A#2': 116.54, 'B2': 123.47,
+            'C3': 130.81, 'C#3': 138.59, 'D3': 146.83, 'D#3': 155.56, 'E3': 164.81, 'F3': 174.61,
+            'F#3': 185.00, 'G3': 196.00, 'G#3': 207.65, 'A3': 220.00, 'A#3': 233.08, 'B3': 246.94,
+            'C4': 261.63, 'C#4': 277.18, 'D4': 293.66, 'D#4': 311.13, 'E4': 329.63, 'F4': 349.23,
+            'F#4': 369.99, 'G4': 392.00, 'G#4': 415.30, 'A4': 440.00, 'A#4': 466.16, 'B4': 493.88,
+            'C5': 523.25, 'C#5': 554.37, 'D5': 587.33, 'D#5': 622.25, 'E5': 659.26, 'F5': 698.46,
+            'F#5': 739.99, 'G5': 783.99, 'G#5': 830.61, 'A5': 880.00, 'A#5': 932.33, 'B5': 987.77,
+            'C6': 1046.50
+        };
+
+        // Create ticks for the x-axis
+        const xAxisTicks = Object.entries(noteFrequencies).filter(([note]) => 
+            note.length === 2 || note === 'C#4' || note === 'F#4' || note === 'A#4'
+        ).map(([note, freq]) => ({
+            value: freq,
+            label: note
+        }));
+
+        // Keep track of recent maximum values
+        const maxValueHistory = [];
+        const MAX_HISTORY_LENGTH = 5;
+
+        function updateYAxisScale(newValue) {
+            maxValueHistory.push(newValue);
+            if (maxValueHistory.length > MAX_HISTORY_LENGTH) {
+                maxValueHistory.shift();
+            }
+            return Math.max(...maxValueHistory) * 1.1; // Add 10% margin
+        }
+
+        const ctx = document.getElementById('spectrumChart').getContext('2d');
+        const chart = new Chart(ctx, {
+            type: 'line',
+            data: {
+                // Generate logarithmically spaced frequencies for labels
+                labels: Array.from({length: 134}, (_, i) => {
+                    const minFreq = 60;
+                    const maxFreq = 1100;
+                    return Math.pow(10, Math.log10(minFreq) + (Math.log10(maxFreq) - Math.log10(minFreq)) * i / 133);
+                }),
+                datasets: [{
+                    label: 'Frequency Spectrum',
+                    data: Array(134).fill(0),
+                    borderColor: 'rgb(75, 192, 192)',
+                    tension: 0.1,
+                    fill: true,
+                    backgroundColor: 'rgba(75, 192, 192, 0.2)',
+                    pointRadius: 0 // Hide points for better performance
+                }]
+            },
+            options: {
+                responsive: true,
+                maintainAspectRatio: false,
+                animation: {
+                    duration: 0
+                },
+                parsing: {
+                    xAxisKey: 'x',
+                    yAxisKey: 'y'
+                },
+                scales: {
+                    y: {
+                        beginAtZero: true,
+                        max: 5000,
+                        adapters: {
+                            update: function(maxValue) {
+                                return updateYAxisScale(maxValue);
+                            }
+                        },
+                        title: {
+                            display: true,
+                            text: 'Magnitude'
+                        }
+                    },
+                    x: {
+                        type: 'logarithmic',
+                        position: 'bottom',
+                        min: 60,
+                        max: 1100,
+                        title: {
+                            display: true,
+                            text: 'Frequency (Hz)'
+                        },
+                        ticks: {
+                            callback: function(value) {
+                                // Show C notes and F# notes
+                                const entries = Object.entries(noteFrequencies);
+                                const closest = entries.reduce((prev, curr) => {
+                                    return Math.abs(curr[1] - value) < Math.abs(prev[1] - value) ? curr : prev;
+                                });
+                                
+                                if ((closest[0].includes('C') || closest[0].includes('F#')) && 
+                                    Math.abs(closest[1] - value) < 1) {
+                                    return closest[0];
+                                }
+                                return '';
+                            },
+                            sampleSize: 20,
+                            autoSkip: false
+                        },
+                        grid: {
+                            color: (ctx) => {
+                                const value = ctx.tick.value;
+                                // Check if this tick corresponds to a C note
+                                if (Object.entries(noteFrequencies)
+                                    .some(([note, freq]) => 
+                                        note.startsWith('C') && Math.abs(freq - value) < 1)) {
+                                    return 'rgba(255, 0, 0, 0.1)';
+                                }
+                                return 'rgba(0, 0, 0, 0.1)';
+                            }
+                        }
+                    }
+                },
+                plugins: {
+                    annotation: {
+                        annotations: Object.entries(noteFrequencies)
+                            .filter(([note]) => note.startsWith('C'))
+                            .reduce((acc, [note, freq]) => {
+                                acc[note] = {
+                                    type: 'line',
+                                    xMin: freq,
+                                    xMax: freq,
+                                    borderColor: 'rgba(255, 99, 132, 0.5)',
+                                    borderWidth: 1,
+                                    label: {
+                                        content: note,
+                                        enabled: true,
+                                        position: 'top'
+                                    }
+                                };
+                                return acc;
+                            }, {})
+                    },
+                    tooltip: {
+                        callbacks: {
+                            title: function(context) {
+                                const freq = context[0].parsed.x;
+                                // Find the closest note
+                                const closestNote = Object.entries(noteFrequencies)
+                                    .reduce((closest, [note, noteFreq]) => {
+                                        return Math.abs(noteFreq - freq) < Math.abs(noteFreq - closest.freq)
+                                            ? { note, freq: noteFreq }
+                                            : closest;
+                                    }, { note: '', freq: Infinity });
+                                
+                                return `Frequency: ${freq.toFixed(1)} Hz (Near ${closestNote.note})`;
+                            }
+                        }
+                    },
+                    legend: {
+                        display: false
+                    }
+                }
+            }
+        });
+
+        const wsUrl = `ws://${window.location.hostname}/ws`;
+        const ws = new WebSocket(wsUrl);
+
+        function interpolateSpectrum(spectrum) {
+            const result = [];
+            const minFreq = 60;
+            const maxFreq = 1100;
+            const binWidth = 8000 / 1024; // Hz per bin
+            
+            // Generate logarithmically spaced frequencies
+            for (let i = 0; i < 134; i++) {
+                const targetFreq = Math.pow(10, Math.log10(minFreq) + (Math.log10(maxFreq) - Math.log10(minFreq)) * i / 133);
+                
+                // Find the corresponding linear bin
+                const bin = Math.floor(targetFreq / binWidth);
+                
+                if (bin >= 8 && bin < 141) {
+                    // Linear interpolation between bins
+                    const binFraction = (targetFreq / binWidth) - bin;
+                    const value = spectrum[bin - 8] * (1 - binFraction) + 
+                                (bin - 7 < spectrum.length ? spectrum[bin - 7] : 0) * binFraction;
+                    result.push({
+                        x: targetFreq,
+                        y: value
+                    });
+                } else {
+                    result.push({
+                        x: targetFreq,
+                        y: 0
+                    });
+                }
+            }
+            
+            return result;
+        }
+
+        ws.onmessage = function(event) {
+            const spectrum = JSON.parse(event.data);
+            const interpolatedData = interpolateSpectrum(spectrum);
+            
+            // Update y-axis scale based on new maximum value
+            const maxValue = Math.max(...interpolatedData.map(d => d.y));
+            chart.options.scales.y.max = updateYAxisScale(maxValue);
+            
+            // Update chart data
+            chart.data.datasets[0].data = interpolatedData;
+            chart.update('none'); // Use 'none' mode for maximum performance
+        };
+
+        ws.onclose = function() {
+            console.log('WebSocket connection closed');
+            setTimeout(() => {
+                window.location.reload();
+            }, 1000);
+        };
+    </script>
+</body>
+</html>

include/AudioLevelTracker.h

@@ -0,0 +1,24 @@
+#pragma once
+
+#include <Arduino.h>
+#include <deque>
+#include "Config.h"
+
+class AudioLevelTracker {
+public:
+    AudioLevelTracker();
+    void updateMaxLevel(int16_t sample);  // Changed to int16_t
+    int16_t getMaxLevel() const;          // Changed to int16_t
+    void resetMaxLevel();
+
+private:
+    struct SamplePoint {
+        uint32_t timestamp;
+        int16_t value;                    // Changed to int16_t
+    };
+    std::deque<SamplePoint> sampleHistory;
+    int16_t maxLevel;                     // Changed to int16_t
+
+    static const uint32_t HISTORY_DURATION_MS = 3000;  // 3 seconds history
+    static const int16_t MAX_RANGE_LIMIT = Config::DEFAULT_RANGE_LIMIT;  // Use 16-bit limit directly
+};

include/Config.h

@@ -0,0 +1,90 @@
+#pragma once
+#include <cstdint>
+
+namespace Config {
+
+    // Serial Configuration
+    constexpr int SERIAL_BAUD_RATE = 115200;     // Serial communication baud rate
+
+    // WiFi Configuration
+    constexpr const char* WIFI_AP_NAME = "ESP32Piano";  // AP name when in configuration mode
+    constexpr const char* WIFI_AP_PASSWORD = "12345678"; // AP password when in configuration mode
+    constexpr uint32_t WIFI_CONFIG_TIMEOUT = 180;  // Seconds to wait for WiFi configuration
+    constexpr uint32_t WIFI_CONFIG_PORT = 80;      // Web configuration portal port
+
+    // I2S Pin Configuration
+    constexpr int I2S_MIC_SERIAL_CLOCK = 8;      // SCK
+    constexpr int I2S_MIC_LEFT_RIGHT_CLOCK = 9;  // WS/LRC 
+    constexpr int I2S_MIC_SERIAL_DATA = 10;      // SD
+
+    // I2S Configuration
+    constexpr int SAMPLE_RATE = 8000;            // Hz - good for frequencies up to 4kHz
+    constexpr int BITS_PER_SAMPLE = 16;          // Changed to 16-bit
+    constexpr int CHANNELS = 1;                  // Mono input
+    constexpr int SAMPLE_BUFFER_SIZE = 1024;     // Match FFT_SIZE for efficiency
+
+    // DMA Configuration
+    constexpr int DMA_BUFFER_COUNT = 4;          // Reduced for lower latency
+    constexpr int DMA_BUFFER_LEN = 512;          // Smaller chunks for faster processing
+
+    // Audio Processing
+    constexpr float DC_OFFSET = 0.0f;            // DC offset correction
+    constexpr float GAIN = 4.0f;                 // Increased gain for better note detection
+    constexpr int16_t NOISE_THRESHOLD = 100;     // Lower threshold for 16-bit samples
+    constexpr int16_t DEFAULT_RANGE_LIMIT = 32767; // Max value for 16-bit
+    constexpr float DECAY_FACTOR = 0.7f;         // Faster decay for quicker note changes
+
+    // FFT Configuration
+    constexpr int FFT_SIZE = 1024;              // Must be power of 2, gives ~7.8 Hz resolution at 8kHz
+    constexpr float FREQUENCY_RESOLUTION = static_cast<float>(SAMPLE_RATE) / FFT_SIZE;
+    
+    // Piano Note Frequencies (Hz)
+    constexpr float NOTE_FREQ_C2 = 65.41f;      // Lowest note we'll detect
+    constexpr float NOTE_FREQ_C3 = 130.81f;
+    constexpr float NOTE_FREQ_C4 = 261.63f;     // Middle C
+    constexpr float NOTE_FREQ_C5 = 523.25f;
+    constexpr float NOTE_FREQ_C6 = 1046.50f;    // Highest note we'll detect
+    
+    // Note Detection Parameters
+    constexpr float FREQUENCY_TOLERANCE = 3.0f;   // Hz tolerance, increased for better detection
+    constexpr float MIN_MAGNITUDE_THRESHOLD = 500.0f; // Adjusted for 16-bit samples
+    constexpr int MAX_SIMULTANEOUS_NOTES = 7;    // Maximum number of notes to detect simultaneously
+    constexpr float PEAK_RATIO_THRESHOLD = 0.1f; // Peaks must be at least this ratio of the strongest peak
+
+    // Timing and Debug
+    constexpr uint32_t LEVEL_UPDATE_INTERVAL_MS = 50;   // Faster updates for better responsiveness
+    constexpr bool ENABLE_DEBUG = true;                 // Enable debug output
+    constexpr int DEBUG_INTERVAL_MS = 1000;            // Debug print interval
+
+    // System Configuration
+    constexpr uint32_t TASK_STACK_SIZE = 4096;   // Audio task stack size
+    constexpr uint8_t TASK_PRIORITY = 2;         // Increased priority for more consistent timing
+    constexpr uint8_t TASK_CORE = 1;            // Run on second core to avoid interference
+
+    // Calibration Parameters
+    constexpr bool CALIBRATION_MODE = false;     // Set to true to enable calibration output
+    constexpr int CALIBRATION_DURATION_MS = 5000; // Duration to collect calibration data
+    constexpr float CALIBRATION_PEAK_PERCENTILE = 0.95f; // Use 95th percentile for thresholds
+    
+    // Note Detection Timing
+    constexpr int NOTE_PRINT_INTERVAL_MS = 100;  // How often to print detected notes
+    constexpr int MIN_NOTE_DURATION_MS = 50;     // Minimum duration to consider a note valid
+    constexpr int NOTE_RELEASE_TIME_MS = 100;    // Time before considering a note released
+
+    // Serial Commands
+    constexpr char CMD_HELP = 'h';           // Show help
+    constexpr char CMD_CALIBRATE = 'c';      // Start calibration
+    constexpr char CMD_THRESHOLD_UP = '+';   // Increase sensitivity
+    constexpr char CMD_THRESHOLD_DOWN = '-';  // Decrease sensitivity
+    constexpr char CMD_TOGGLE_SPECTRUM = 's'; // Toggle spectrum display
+    constexpr float THRESHOLD_STEP = 0.1f;   // Threshold adjustment step
+
+    // Command Response Messages
+    constexpr const char* MSG_HELP = 
+        "Commands:\n"
+        "h - Show this help\n"
+        "c - Start calibration\n"
+        "+ - Increase sensitivity\n"
+        "- - Decrease sensitivity\n"
+        "s - Toggle spectrum display\n";
+}

include/I2SConfig.h

@@ -0,0 +1,9 @@
+#pragma once
+
+#include <Arduino.h>
+#include <driver/i2s.h>
+#include "Config.h"
+
+// I2S setup functions
+void initI2S();
+void readI2SSamples(int16_t* samples, size_t* bytesRead);

include/NoteDetector.h

@@ -0,0 +1,48 @@
+#pragma once
+
+#include <Arduino.h>
+#include <arduinoFFT.h>
+#include <vector>
+#include <deque>
+#include "Config.h"
+
+struct DetectedNote {
+    float frequency;
+    float magnitude;
+    uint8_t noteNumber;  // MIDI note number for easy identification
+    uint32_t startTime;  // When the note was first detected
+    uint32_t lastSeenTime; // Last time this note was detected
+};
+
+class NoteDetector {
+public:
+    NoteDetector();
+    void analyzeSamples(const int16_t* samples, size_t sampleCount);
+    const std::vector<DetectedNote>& getDetectedNotes() const { return detectedNotes; }
+    const double* getSpectrum() const { return vReal; }
+    void beginCalibration();
+    void endCalibration();
+    bool isCalibrating() const { return calibrationMode; }
+    
+    // Threshold adjustment methods
+    void adjustThreshold(float delta);
+    float getCurrentThreshold() const { return dynamicThreshold; }
+
+private:
+    arduinoFFT FFT;
+    double vReal[Config::FFT_SIZE];
+    double vImag[Config::FFT_SIZE];
+    std::vector<DetectedNote> detectedNotes;
+    std::deque<float> peakMagnitudeHistory;  // For calibration
+    
+    bool calibrationMode;
+    uint32_t calibrationStartTime;
+    float dynamicThreshold;
+    
+    uint8_t freqToNoteNumber(float frequency) const;
+    bool isNoteFrequency(float frequency) const;
+    bool isHarmonic(float freq1, float freq2, float tolerance) const;
+    void findPeaks();
+    void updateNoteTimings();
+    void calculateDynamicThreshold();
+};

include/SpectrumVisualizer.h

@@ -0,0 +1,27 @@
+#pragma once
+
+#include <Arduino.h>
+#include "Config.h"
+#include "NoteDetector.h"
+
+class SpectrumVisualizer {
+public:
+    // Visualization settings
+    static constexpr int DISPLAY_WIDTH = 80;  // Characters wide
+    static constexpr int DISPLAY_HEIGHT = 16; // Lines tall
+    static constexpr float DB_MIN = 30.0f;    // Minimum dB to show
+    static constexpr float DB_MAX = 90.0f;    // Maximum dB to show
+
+    static void visualizeSpectrum(const double* spectrum, int size);
+    static void visualizeNotes(const std::vector<DetectedNote>& notes);
+    static void drawFFTMagnitudes(const double* magnitudes, int size, bool logScale = true);
+    
+private:
+    static constexpr const char* noteNames[] = {
+        "C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"
+    };
+    
+    static float magnitudeToDb(double magnitude);
+    static int mapToDisplay(float value, float min, float max, int displayMin, int displayMax);
+    static void printBarGraph(float value, float maxValue, int width);
+};

platformio.ini

@@ -36,3 +36,4 @@ lib_deps =
     https://github.com/tzapu/WiFiManager.git
     me-no-dev/AsyncTCP
     https://github.com/me-no-dev/ESPAsyncWebServer.git
+    kosme/arduinoFFT@^1.6

src/AudioLevelTracker.cpp

@@ -0,0 +1,38 @@
+#include "AudioLevelTracker.h"
+
+AudioLevelTracker::AudioLevelTracker() {
+    resetMaxLevel();
+}
+
+void AudioLevelTracker::updateMaxLevel(int16_t sample) {
+    uint32_t currentTime = millis();
+    
+    // Remove old samples (older than specified duration)
+    while (!sampleHistory.empty() && 
+            (currentTime - sampleHistory.front().timestamp) > HISTORY_DURATION_MS) {
+        sampleHistory.pop_front();
+    }
+
+    // Add new sample, but cap it at MAX_RANGE_LIMIT
+    int16_t absValue = abs(sample);
+    absValue = min(absValue, MAX_RANGE_LIMIT);  // Cap at 16-bit limit
+    SamplePoint newPoint = {currentTime, absValue};
+    sampleHistory.push_back(newPoint);
+
+    // Update maximum
+    maxLevel = 0;
+    for (const auto& point : sampleHistory) {
+        if (point.value > maxLevel) {
+            maxLevel = point.value;
+        }
+    }
+}
+
+int16_t AudioLevelTracker::getMaxLevel() const {
+    return maxLevel;
+}
+
+void AudioLevelTracker::resetMaxLevel() {
+    maxLevel = 0;
+    sampleHistory.clear();
+}

src/I2SConfig.cpp

@@ -0,0 +1,34 @@
+#include "I2SConfig.h"
+
+void initI2S() {
+    // I2S configuration
+    i2s_config_t i2s_config = {
+        .mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_RX),
+        .sample_rate = Config::SAMPLE_RATE,
+        .bits_per_sample = I2S_BITS_PER_SAMPLE_16BIT,
+        .channel_format = I2S_CHANNEL_FMT_ONLY_LEFT,
+        .communication_format = I2S_COMM_FORMAT_STAND_I2S,
+        .intr_alloc_flags = ESP_INTR_FLAG_LEVEL1,
+        .dma_buf_count = Config::DMA_BUFFER_COUNT,
+        .dma_buf_len = Config::DMA_BUFFER_LEN,
+        .use_apll = false,
+        .tx_desc_auto_clear = false,
+        .fixed_mclk = 0
+    };
+
+    // I2S pin configuration
+    i2s_pin_config_t i2s_mic_pins = {
+        .bck_io_num = Config::I2S_MIC_SERIAL_CLOCK,
+        .ws_io_num = Config::I2S_MIC_LEFT_RIGHT_CLOCK,
+        .data_out_num = I2S_PIN_NO_CHANGE,
+        .data_in_num = Config::I2S_MIC_SERIAL_DATA
+    };
+
+    // Install and configure I2S driver
+    i2s_driver_install(I2S_NUM_0, &i2s_config, 0, NULL);
+    i2s_set_pin(I2S_NUM_0, &i2s_mic_pins);
+}
+
+void readI2SSamples(int16_t* samples, size_t* bytesRead) {
+    i2s_read(I2S_NUM_0, samples, Config::SAMPLE_BUFFER_SIZE * sizeof(int16_t), bytesRead, portMAX_DELAY);
+}

src/NoteDetector.cpp

@@ -0,0 +1,198 @@
+#include "NoteDetector.h"
+#include <algorithm>
+#include <cmath>
+
+NoteDetector::NoteDetector() 
+    : FFT(vReal, vImag, Config::FFT_SIZE, Config::SAMPLE_RATE)
+    , calibrationMode(false)
+    , calibrationStartTime(0)
+    , dynamicThreshold(Config::MIN_MAGNITUDE_THRESHOLD) {
+}
+
+void NoteDetector::beginCalibration() {
+    calibrationMode = true;
+    calibrationStartTime = millis();
+    peakMagnitudeHistory.clear();
+    Serial.println("Beginning calibration...");
+}
+
+void NoteDetector::endCalibration() {
+    calibrationMode = false;
+    calculateDynamicThreshold();
+    Serial.printf("Calibration complete. Dynamic threshold: %.2f\n", dynamicThreshold);
+}
+
+void NoteDetector::calculateDynamicThreshold() {
+    if (peakMagnitudeHistory.empty()) {
+        dynamicThreshold = Config::MIN_MAGNITUDE_THRESHOLD;
+        return;
+    }
+
+    // Sort magnitudes to find percentile
+    std::vector<float> sorted(peakMagnitudeHistory.begin(), peakMagnitudeHistory.end());
+    std::sort(sorted.begin(), sorted.end());
+    
+    // Calculate threshold at specified percentile
+    size_t idx = static_cast<size_t>(sorted.size() * Config::CALIBRATION_PEAK_PERCENTILE);
+    dynamicThreshold = sorted[idx] * Config::PEAK_RATIO_THRESHOLD;
+    
+    // Don't go below minimum threshold
+    dynamicThreshold = (dynamicThreshold > Config::MIN_MAGNITUDE_THRESHOLD) ? 
+                      dynamicThreshold : Config::MIN_MAGNITUDE_THRESHOLD;
+}
+
+void NoteDetector::updateNoteTimings() {
+    uint32_t currentTime = millis();
+    
+    // Remove notes that haven't been seen recently
+    detectedNotes.erase(
+        std::remove_if(detectedNotes.begin(), detectedNotes.end(),
+            [currentTime](const DetectedNote& note) {
+                return currentTime - note.lastSeenTime > Config::NOTE_RELEASE_TIME_MS;
+            }),
+        detectedNotes.end()
+    );
+}
+
+void NoteDetector::analyzeSamples(const int16_t* samples, size_t sampleCount) {
+    // Copy samples to FFT buffer with gain applied
+    for (size_t i = 0; i < Config::FFT_SIZE; i++) {
+        vReal[i] = i < sampleCount ? samples[i] * Config::GAIN : 0;
+        vImag[i] = 0;
+    }
+
+    // Perform FFT
+    FFT.DCRemoval();
+    FFT.Windowing(FFT_WIN_TYP_HANN, FFT_FORWARD);
+    FFT.Compute(FFT_FORWARD);
+    FFT.ComplexToMagnitude();
+
+    // Find peaks and update timings
+    findPeaks();
+    updateNoteTimings();
+
+    // Handle calibration mode
+    if (calibrationMode) {
+        // Store peak magnitudes during calibration
+        double maxMagnitude = 0;
+        for (int i = 2; i < Config::FFT_SIZE/2 - 1; i++) {
+            maxMagnitude = std::max(maxMagnitude, vReal[i]);
+        }
+        peakMagnitudeHistory.push_back(maxMagnitude);
+
+        // Check if calibration period is over
+        if (millis() - calibrationStartTime >= Config::CALIBRATION_DURATION_MS) {
+            endCalibration();
+        }
+    }
+}
+
+uint8_t NoteDetector::freqToNoteNumber(float frequency) const {
+    // A4 is 440Hz and MIDI note 69
+    return round(12 * log2(frequency/440.0) + 69);
+}
+
+bool NoteDetector::isNoteFrequency(float frequency) const {
+    // Check if frequency is within range of C2 to C6
+    if (frequency < (Config::NOTE_FREQ_C2 - Config::FREQUENCY_TOLERANCE) ||
+        frequency > (Config::NOTE_FREQ_C6 + Config::FREQUENCY_TOLERANCE)) {
+        return false;
+    }
+
+    // Calculate note number and then expected frequency
+    uint8_t noteNum = freqToNoteNumber(frequency);
+    float expectedFreq = 440.0f * pow(2.0f, (noteNum - 69) / 12.0f);
+    
+    // Check if within tolerance
+    return abs(frequency - expectedFreq) <= Config::FREQUENCY_TOLERANCE;
+}
+
+bool NoteDetector::isHarmonic(float freq1, float freq2, float tolerance) const {
+    // Check if freq2 is a harmonic of freq1
+    if (freq2 < freq1) {
+        std::swap(freq1, freq2);
+    }
+
+    float ratio = freq2 / freq1;
+    float nearest_harmonic = round(ratio);
+    
+    return abs(ratio - nearest_harmonic) <= (tolerance / freq1);
+}
+
+void NoteDetector::findPeaks() {
+    std::vector<DetectedNote> candidates;
+    uint32_t currentTime = millis();
+
+    // Find the maximum magnitude
+    double maxMagnitude = 0;
+    for (int i = 2; i < Config::FFT_SIZE/2 - 1; i++) {
+        maxMagnitude = (vReal[i] > maxMagnitude) ? vReal[i] : maxMagnitude;
+    }
+    
+    // Use either calibrated or relative threshold
+    float relativeThreshold = static_cast<float>(maxMagnitude * Config::PEAK_RATIO_THRESHOLD);
+    float threshold = calibrationMode ? Config::MIN_MAGNITUDE_THRESHOLD :
+                     (dynamicThreshold > relativeThreshold ? dynamicThreshold : relativeThreshold);
+
+    // First pass: find all potential peaks
+    for (int i = 2; i < Config::FFT_SIZE/2 - 1; i++) {
+        float frequency = i * Config::FREQUENCY_RESOLUTION;
+        
+        if (vReal[i] > threshold &&
+            vReal[i] > vReal[i-1] &&
+            vReal[i] > vReal[i+1] &&
+            isNoteFrequency(frequency)) {
+            
+            DetectedNote note;
+            note.frequency = frequency;
+            note.magnitude = vReal[i];
+            note.noteNumber = freqToNoteNumber(frequency);
+            note.startTime = currentTime;
+            note.lastSeenTime = currentTime;
+            candidates.push_back(note);
+        }
+    }
+
+    // Sort candidates by magnitude
+    std::sort(candidates.begin(), candidates.end(),
+              [](const DetectedNote& a, const DetectedNote& b) {
+                  return a.magnitude > b.magnitude;
+              });
+
+    // Second pass: filter out harmonics and update existing notes
+    for (const auto& candidate : candidates) {
+        // Check if this note is already being tracked
+        auto existingNote = std::find_if(detectedNotes.begin(), detectedNotes.end(),
+            [&candidate](const DetectedNote& note) {
+                return note.noteNumber == candidate.noteNumber;
+            });
+
+        if (existingNote != detectedNotes.end()) {
+            // Update existing note
+            existingNote->magnitude = candidate.magnitude;
+            existingNote->lastSeenTime = currentTime;
+            continue;
+        }
+
+        // Check if this is a harmonic of a stronger note
+        bool isHarmonicOfStronger = false;
+        for (const auto& note : detectedNotes) {
+            if (isHarmonic(candidate.frequency, note.frequency, Config::FREQUENCY_TOLERANCE)) {
+                isHarmonicOfStronger = true;
+                break;
+            }
+        }
+
+        // Add new note if it's not a harmonic and we haven't hit the limit
+        if (!isHarmonicOfStronger && detectedNotes.size() < Config::MAX_SIMULTANEOUS_NOTES) {
+            detectedNotes.push_back(candidate);
+        }
+    }
+}
+
+void NoteDetector::adjustThreshold(float delta) {
+    float newThreshold = dynamicThreshold + (dynamicThreshold * delta);
+    dynamicThreshold = (newThreshold > Config::MIN_MAGNITUDE_THRESHOLD) ? 
+                      newThreshold : Config::MIN_MAGNITUDE_THRESHOLD;
+    Serial.printf("Threshold adjusted to: %.1f\n", dynamicThreshold);
+}

src/SpectrumVisualizer.cpp

@@ -0,0 +1,82 @@
+#include "SpectrumVisualizer.h"
+#include <cmath>
+
+constexpr const char* SpectrumVisualizer::noteNames[];
+
+float SpectrumVisualizer::magnitudeToDb(double magnitude) {
+    return 20 * log10(magnitude);
+}
+
+int SpectrumVisualizer::mapToDisplay(float value, float min, float max, int displayMin, int displayMax) {
+    float normalized = (value - min) / (max - min);
+    normalized = constrain(normalized, 0.0f, 1.0f);
+    return displayMin + (normalized * (displayMax - displayMin));
+}
+
+void SpectrumVisualizer::printBarGraph(float value, float maxValue, int width) {
+    int barWidth = mapToDisplay(value, 0, maxValue, 0, width);
+    for (int i = 0; i < width; i++) {
+        Serial.print(i < barWidth ? "█" : " ");
+    }
+}
+
+void SpectrumVisualizer::drawFFTMagnitudes(const double* magnitudes, int size, bool logScale) {
+    // Find maximum magnitude for scaling
+    double maxMagnitude = 0;
+    for (int i = 0; i < size/2; i++) {
+        maxMagnitude = max(maxMagnitude, magnitudes[i]);
+    }
+
+    // Print frequency spectrum
+    Serial.println("\nFrequency Spectrum:");
+    Serial.println("Freq (Hz) | Magnitude");
+    Serial.println("-----------|-" + String('-', DISPLAY_WIDTH));
+
+    for (int i = 0; i < size/2; i++) {
+        float freq = i * Config::FREQUENCY_RESOLUTION;
+        if (freq > Config::NOTE_FREQ_C6) break; // Stop at highest note frequency
+        
+        float magnitude = magnitudes[i];
+        if (logScale) {
+            magnitude = magnitudeToDb(magnitude);
+            maxMagnitude = magnitudeToDb(maxMagnitude);
+        }
+
+        // Only print if significant magnitude
+        if (magnitude > (logScale ? DB_MIN : 0)) {
+            Serial.printf("%8.1f | ", freq);
+            printBarGraph(magnitude, maxMagnitude, DISPLAY_WIDTH);
+            Serial.println();
+        }
+    }
+}
+
+void SpectrumVisualizer::visualizeSpectrum(const double* spectrum, int size) {
+    drawFFTMagnitudes(spectrum, size, true);
+}
+
+void SpectrumVisualizer::visualizeNotes(const std::vector<DetectedNote>& notes) {
+    if (notes.empty()) return;
+
+    // Find maximum magnitude for scaling
+    float maxMagnitude = 0;
+    for (const auto& note : notes) {
+        maxMagnitude = max(maxMagnitude, note.magnitude);
+    }
+
+    // Print piano keyboard visualization
+    Serial.println("\nDetected Notes:");
+    Serial.println("Note    | Freq (Hz) | Level");
+    Serial.println("--------|-----------|" + String('-', DISPLAY_WIDTH));
+
+    for (const auto& note : notes) {
+        int octave = (note.noteNumber / 12) - 1;
+        int noteIndex = note.noteNumber % 12;
+        float dbLevel = magnitudeToDb(note.magnitude);
+        
+        Serial.printf("%s%-6d | %8.1f | ", 
+                     noteNames[noteIndex], octave, note.frequency);
+        printBarGraph(note.magnitude, maxMagnitude, DISPLAY_WIDTH);
+        Serial.printf(" %.1f dB\n", dbLevel);
+    }
+}

src/main.cpp

@@ -1,130 +1,275 @@
 #include <Arduino.h>
-#include <driver/i2s.h>
+#include <WiFi.h>
-#include <deque>
+#include <WiFiManager.h>
+#include <ESPAsyncWebServer.h>
+#include <AsyncTCP.h>
+#include <SPIFFS.h>
+#include "Config.h"
+#include "I2SConfig.h"
+#include "AudioLevelTracker.h"
+#include "NoteDetector.h"
+#include "SpectrumVisualizer.h"
 
-// you shouldn't need to change these settings
+// Function declarations
-#define SAMPLE_BUFFER_SIZE 512
+void onWebSocketEvent(AsyncWebSocket *server, AsyncWebSocketClient *client, AwsEventType type, void *arg, uint8_t *data, size_t len);
-#define SAMPLE_RATE 8000
+void initWebServer();
-// most microphones will probably default to left channel but you may need to tie the L/R pin low
+void handleSerialCommands();
-#define I2S_MIC_CHANNEL I2S_CHANNEL_FMT_ONLY_LEFT
+void printNoteInfo(const DetectedNote& note);
-// either wire your microphone to the same pins or change these to match your wiring
+void initWiFi();
-#define I2S_MIC_SERIAL_CLOCK 8
+void audioProcessingTask(void *parameter);
-#define I2S_MIC_LEFT_RIGHT_CLOCK 9
+void visualizationTask(void *parameter);
-#define I2S_MIC_SERIAL_DATA 10
+void sendSpectrumData();
 
-// Add sample history tracking for range limiting
+// Static instances
-#define HISTORY_DURATION_MS 1000  // 1 seconds history
+static int16_t raw_samples[Config::SAMPLE_BUFFER_SIZE];
-#define SAMPLES_PER_MS (SAMPLE_RATE / 1000)
+static AudioLevelTracker audioLevelTracker;
-#define HISTORY_SIZE (HISTORY_DURATION_MS * SAMPLES_PER_MS)
+static NoteDetector noteDetector;
+WiFiManager wifiManager;
+AsyncWebServer server(80);
+AsyncWebSocket ws("/ws");
 
-#define MAX_RANGE_LIMIT 150000000  // Maximum allowed range limit
+// Timing and state variables
-#define DEFAULT_RANGE_LIMIT 20000  // Default range limit when no history
+static uint32_t lastNotePrintTime = 0;
+static uint32_t lastSpectrumPrintTime = 0;
+static uint32_t lastWebUpdateTime = 0;
+static bool showSpectrum = false;
 
-class AudioLevelTracker {
+// Task handles
-public:
+TaskHandle_t audioTaskHandle = nullptr;
-    AudioLevelTracker() {
+TaskHandle_t visualizationTaskHandle = nullptr;
-        resetMaxLevel();
-    }
 
-    void updateMaxLevel(int32_t sample) {
+// Queue for passing audio data between cores
-        uint32_t currentTime = millis();
+QueueHandle_t audioQueue;
+
+// Note names for display
+const char* noteNames[] = {"C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"};
+
+void sendSpectrumData() {
+    if (ws.count() > 0 && !noteDetector.isCalibrating()) {
+        const auto& spectrum = noteDetector.getSpectrum();
+        String json = "[";
         
-        // Remove old samples (older than 10 seconds)
+        // Calculate bin range for 60-1100 Hz
-        while (!sampleHistory.empty() && 
+        // At 8kHz sample rate with 1024 FFT size:
-               (currentTime - sampleHistory.front().timestamp) > HISTORY_DURATION_MS) {
+        // binFreq = index * (8000/1024) = index * 7.8125 Hz
-            sampleHistory.pop_front();
+        // For 60 Hz: bin ≈ 8
-        }
+        // For 1100 Hz: bin ≈ 141
-
+        
-        // Add new sample, but cap it at MAX_RANGE_LIMIT
+        for (int i = 8; i <= 141; i++) {
-        int32_t absValue = abs(sample);
+            if (i > 8) json += ",";
-        absValue = min(absValue, MAX_RANGE_LIMIT);  // Cap the value
+            json += String(spectrum[i], 2);
-        SamplePoint newPoint = {currentTime, absValue};
-        sampleHistory.push_back(newPoint);
-
-        // Update maximum
-        maxLevel = 0;
-        for (const auto& point : sampleHistory) {
-            if (point.value > maxLevel) {
-                maxLevel = point.value;
-            }
         }
+        json += "]";
+        ws.textAll(json);
     }
-
-    int32_t getMaxLevel() const {
-        return maxLevel;
-    }
-
-    void resetMaxLevel() {
-        maxLevel = 0;
-        sampleHistory.clear();
-    }
-
-private:
-    struct SamplePoint {
-        uint32_t timestamp;
-        int32_t value;
-    };
-    std::deque<SamplePoint> sampleHistory;
-    int32_t maxLevel;
-};
-
-AudioLevelTracker audioLevelTracker;
-
-// don't mess around with this
-i2s_config_t i2s_config = {
-    .mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_RX),
-    .sample_rate = SAMPLE_RATE,
-    .bits_per_sample = I2S_BITS_PER_SAMPLE_32BIT,
-    .channel_format = I2S_CHANNEL_FMT_ONLY_LEFT,
-    .communication_format = I2S_COMM_FORMAT_STAND_I2S,  // Updated from I2S_COMM_FORMAT_I2S
-    .intr_alloc_flags = ESP_INTR_FLAG_LEVEL1,
-    .dma_buf_count = 4,
-    .dma_buf_len = 1024,
-    .use_apll = false,
-    .tx_desc_auto_clear = false,
-    .fixed_mclk = 0};
-
-// and don't mess around with this
-i2s_pin_config_t i2s_mic_pins = {
-    .bck_io_num = I2S_MIC_SERIAL_CLOCK,
-    .ws_io_num = I2S_MIC_LEFT_RIGHT_CLOCK,
-    .data_out_num = I2S_PIN_NO_CHANGE,
-    .data_in_num = I2S_MIC_SERIAL_DATA};
-
-void setup()
-{
-  // we need serial output for the plotter
-  Serial.begin(115200);
-  // start up the I2S peripheral
-  i2s_driver_install(I2S_NUM_0, &i2s_config, 0, NULL);
-  i2s_set_pin(I2S_NUM_0, &i2s_mic_pins);
 }
 
-int32_t raw_samples[SAMPLE_BUFFER_SIZE];
+void onWebSocketEvent(AsyncWebSocket *server, AsyncWebSocketClient *client, AwsEventType type, void *arg, uint8_t *data, size_t len) {
-void loop()
+    switch (type) {
-{
+        case WS_EVT_CONNECT:
-  // read from the I2S device
+            Serial.printf("WebSocket client #%u connected from %s\n", client->id(), client->remoteIP().toString().c_str());
-  size_t bytes_read = 0;
+            break;
-  i2s_read(I2S_NUM_0, raw_samples, sizeof(int32_t) * SAMPLE_BUFFER_SIZE, &bytes_read, portMAX_DELAY);
+        case WS_EVT_DISCONNECT:
-  int samples_read = bytes_read / sizeof(int32_t);
+            Serial.printf("WebSocket client #%u disconnected\n", client->id());
-  
+            break;
-  // Calculate dynamic range limit based on max level from last 10 seconds
+        case WS_EVT_DATA:
-  int32_t currentMaxLevel = audioLevelTracker.getMaxLevel();
+            break;
-  int32_t rangelimit = currentMaxLevel > 0 ? currentMaxLevel : DEFAULT_RANGE_LIMIT; // fallback to default if no history
+        case WS_EVT_ERROR:
+            break;
+    }
+}
 
-  // dump the samples out to the serial channel
+void initWebServer() {
-  for (int i = 0; i < samples_read; i++)
+    if (!SPIFFS.begin(true)) {
-  {
+        Serial.println("An error occurred while mounting SPIFFS");
-    // Update the max level tracker with current sample
+        return;
-    audioLevelTracker.updateMaxLevel(raw_samples[i]);
+    }
 
-    // Print range limits for plotter
+    ws.onEvent(onWebSocketEvent);
-    Serial.print(rangelimit * -1);
+    server.addHandler(&ws);
-    Serial.print(" ");
-    Serial.print(rangelimit);
-    Serial.print(" ");
 
-    // Print the actual sample
+    // Serve static files from SPIFFS
-    Serial.printf("%ld\n", raw_samples[i]);
+    server.serveStatic("/", SPIFFS, "/").setDefaultFile("index.html");
-  }
+
+    // Handle not found
+    server.onNotFound([](AsyncWebServerRequest *request) {
+        request->send(404, "text/plain", "Not found");
+    });
+
+    server.begin();
+    Serial.println("HTTP server started");
+}
+
+void handleSerialCommands() {
+    if (Serial.available()) {
+        char cmd = Serial.read();
+        switch (cmd) {
+            case Config::CMD_HELP:
+                Serial.println(Config::MSG_HELP);
+                break;
+                
+            case Config::CMD_CALIBRATE:
+                noteDetector.beginCalibration();
+                break;
+                
+            case Config::CMD_THRESHOLD_UP:
+                noteDetector.adjustThreshold(Config::THRESHOLD_STEP);
+                break;
+                
+            case Config::CMD_THRESHOLD_DOWN:
+                noteDetector.adjustThreshold(-Config::THRESHOLD_STEP);
+                break;
+                
+            case Config::CMD_TOGGLE_SPECTRUM:
+                showSpectrum = !showSpectrum;
+                Serial.printf("Spectrum display %s\n", showSpectrum ? "enabled" : "disabled");
+                break;
+        }
+    }
+}
+
+void printNoteInfo(const DetectedNote& note) {
+    int octave = (note.noteNumber / 12) - 1;
+    int noteIndex = note.noteNumber % 12;
+    uint32_t duration = millis() - note.startTime;
+    
+    Serial.printf("Note: %s%d (%.1f Hz, Magnitude: %.0f, Duration: %ums)\n", 
+                 noteNames[noteIndex], octave, note.frequency, note.magnitude, duration);
+}
+
+void initWiFi() {
+    // Set configuration portal timeout
+    wifiManager.setConfigPortalTimeout(Config::WIFI_CONFIG_TIMEOUT);
+    
+    // Set custom portal settings
+    wifiManager.setAPStaticIPConfig(IPAddress(192,168,4,1), IPAddress(192,168,4,1), IPAddress(255,255,255,0));
+    
+    // Try to connect to saved WiFi credentials
+    if(!wifiManager.autoConnect(Config::WIFI_AP_NAME, Config::WIFI_AP_PASSWORD)) {
+        Serial.println("Failed to connect and hit timeout");
+        ESP.restart();
+    }
+    
+    Serial.println("Successfully connected to WiFi");
+    Serial.print("IP Address: ");
+    Serial.println(WiFi.localIP());
+}
+
+// Audio processing task running on Core 1
+void audioProcessingTask(void *parameter) {
+    while (true) {
+        size_t bytes_read = 0;
+        readI2SSamples(raw_samples, &bytes_read);
+        int samples_read = bytes_read / sizeof(int16_t);
+        
+        // Update level tracking
+        for (int i = 0; i < samples_read; i++) {
+            audioLevelTracker.updateMaxLevel(raw_samples[i]);
+        }
+
+        // Only analyze if we have enough signal
+        int16_t currentMaxLevel = audioLevelTracker.getMaxLevel();
+        if (currentMaxLevel > Config::NOISE_THRESHOLD) {
+            // Analyze samples for note detection
+            noteDetector.analyzeSamples(raw_samples, samples_read);
+            
+            // Send results to visualization task via queue
+            if (xQueueSend(audioQueue, &samples_read, 0) != pdTRUE) {
+                // Queue full, just skip this update
+            }
+        }
+        
+        // Small delay to prevent watchdog trigger
+        vTaskDelay(1);
+    }
+}
+
+// Visualization and network task running on Core 0
+void visualizationTask(void *parameter) {
+    while (true) {
+        int samples_read;
+        
+        // Check if there's new audio data to process
+        if (xQueueReceive(audioQueue, &samples_read, 0) == pdTRUE) {
+            uint32_t currentTime = millis();
+            const auto& detectedNotes = noteDetector.getDetectedNotes();
+
+            // Update web clients with spectrum data
+            if (currentTime - lastWebUpdateTime >= 50) {
+                sendSpectrumData();
+                lastWebUpdateTime = currentTime;
+            }
+
+            // Show spectrum if enabled
+            if (showSpectrum && 
+                !noteDetector.isCalibrating() && 
+                currentTime - lastSpectrumPrintTime >= Config::DEBUG_INTERVAL_MS) {
+                SpectrumVisualizer::visualizeSpectrum(noteDetector.getSpectrum(), Config::FFT_SIZE);
+                lastSpectrumPrintTime = currentTime;
+            }
+
+            // Print detected notes at specified interval
+            if (currentTime - lastNotePrintTime >= Config::NOTE_PRINT_INTERVAL_MS) {
+                if (!detectedNotes.empty() && !noteDetector.isCalibrating()) {
+                    SpectrumVisualizer::visualizeNotes(detectedNotes);
+                }
+                lastNotePrintTime = currentTime;
+            }
+        }
+
+        ws.cleanupClients();
+        handleSerialCommands();
+        
+        // Small delay to prevent watchdog trigger
+        vTaskDelay(1);
+    }
+}
+
+void setup() {
+    Serial.begin(Config::SERIAL_BAUD_RATE);
+    while(!Serial) {
+        delay(10);
+    }
+    
+    initWiFi();
+    initWebServer();
+    initI2S();
+    
+    Serial.println("Piano Note Detection Ready (C2-C6)");
+    Serial.println("Press 'h' for help");
+    noteDetector.beginCalibration();
+
+    // Create queue for inter-core communication
+    audioQueue = xQueueCreate(4, sizeof(int));
+    if (audioQueue == nullptr) {
+        Serial.println("Failed to create queue!");
+        return;
+    }
+
+    // Create audio processing task on Core 1
+    xTaskCreatePinnedToCore(
+        audioProcessingTask,
+        "AudioTask",
+        Config::TASK_STACK_SIZE,
+        nullptr,
+        Config::TASK_PRIORITY,
+        &audioTaskHandle,
+        1  // Run on Core 1
+    );
+
+    // Create visualization task on Core 0
+    xTaskCreatePinnedToCore(
+        visualizationTask,
+        "VisualTask",
+        Config::TASK_STACK_SIZE,
+        nullptr,
+        1,  // Lower priority than audio task
+        &visualizationTaskHandle,
+        0  // Run on Core 0
+    );
+}
+
+void loop() {
+    // Main loop is now empty as all work is done in tasks
+    vTaskDelete(nullptr);  // Delete the main loop task
 }