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Voice Denoising Model for Alif E7 NPU - Complete Development Guide
Overview
This guide provides step-by-step instructions for developing and deploying a voice denoising model on the Alif E7's Ethos-U55 NPU using the DTLN (Dual-signal Transformation LSTM Network) architecture.
Alif E7 NPU Specifications
- NPU: Dual Arm Ethos-U55 (128 MAC and 256 MAC units)
- CPU: Dual Cortex-M55 (High-Performance @ 400MHz, High-Efficiency @ 160MHz)
- Memory: 256KB ITCM, 1MB DTCM (tightly coupled)
- Supported: 8-bit and 16-bit quantized CNNs/RNNs
- Performance: 250+ GOPS
- Framework: TensorFlow Lite for Microcontrollers
Why DTLN Model?
- Real-time capable: < 1 million parameters
- TF-Lite support: Native quantization and optimization
- Proven edge performance: Runs on Raspberry Pi 3B+
- Low latency: < 8ms execution time required for real-time
- Efficient architecture: Stacked dual-signal transformation LSTM
Step 1: Environment Setup
Prerequisites
# Create conda environment
conda create -n alif_audio python=3.8
conda activate alif_audio
# Install core dependencies
pip install tensorflow==2.12.0
pip install tensorflow-model-optimization
pip install soundfile librosa numpy
pip install tflite-runtime
Step 2: Clone and Setup DTLN Repository
git clone https://github.com/breizhn/DTLN.git
cd DTLN
Step 3: Model Architecture Modifications for Ethos-U55
Create a new file dtln_ethos_u55.py:
import tensorflow as tf
from tensorflow import keras
from tensorflow.keras.layers import LSTM, Dense, Input, Multiply
from tensorflow.keras.models import Model
import numpy as np
class DTLN_Ethos_U55:
"""
DTLN model optimized for Ethos-U55 NPU deployment
- Reduced LSTM units for memory constraints
- 8-bit quantization aware training
- Optimized for < 1MB model size
"""
def __init__(self,
frame_len=512,
frame_shift=128,
lstm_units=128, # Reduced from 256 for NPU
):
self.frame_len = frame_len
self.frame_shift = frame_shift
self.lstm_units = lstm_units
def build_model(self):
"""Build DTLN model with Ethos-U55 optimizations"""
# First stage - Magnitude processing
mag_input = Input(shape=(None, self.frame_len//2 + 1))
# LSTM layer - optimized size for NPU
lstm_1 = LSTM(self.lstm_units,
return_sequences=True,
unroll=False)(mag_input)
# Fully connected layer for mask estimation
mask = Dense(self.frame_len//2 + 1,
activation='sigmoid',
name='magnitude_mask')(lstm_1)
# Apply mask to magnitude spectrum
estimated_mag = Multiply()([mag_input, mask])
# Second stage - Enhanced magnitude processing
lstm_2 = LSTM(self.lstm_units,
return_sequences=True,
unroll=False)(estimated_mag)
# Final mask estimation
final_mask = Dense(self.frame_len//2 + 1,
activation='sigmoid',
name='final_mask')(lstm_2)
# Final enhanced magnitude
enhanced_mag = Multiply()([estimated_mag, final_mask])
model = Model(inputs=mag_input, outputs=enhanced_mag)
return model
def create_stateful_inference_models(self, model):
"""
Create stateful models for real-time inference
This splits the model for frame-by-frame processing
"""
# Model 1: First stage
mag_input = Input(batch_shape=(1, 1, self.frame_len//2 + 1))
lstm_1_state_in_h = Input(batch_shape=(1, self.lstm_units))
lstm_1_state_in_c = Input(batch_shape=(1, self.lstm_units))
lstm_1 = LSTM(self.lstm_units,
return_sequences=True,
stateful=True,
return_state=True)
lstm_out, state_h, state_c = lstm_1(
mag_input,
initial_state=[lstm_1_state_in_h, lstm_1_state_in_c]
)
mask = Dense(self.frame_len//2 + 1, activation='sigmoid')(lstm_out)
estimated_mag = Multiply()([mag_input, mask])
model_1 = Model(
inputs=[mag_input, lstm_1_state_in_h, lstm_1_state_in_c],
outputs=[estimated_mag, state_h, state_c]
)
# Model 2: Second stage (similar structure)
# ... (implement similar structure for stage 2)
return model_1 # Return both models in practice
Step 4: Quantization-Aware Training
Create train_quantized.py:
import tensorflow as tf
import tensorflow_model_optimization as tfmot
from dtln_ethos_u55 import DTLN_Ethos_U55
def apply_quantization_aware_training(model):
"""
Apply quantization-aware training for 8-bit deployment on Ethos-U55
"""
quantize_model = tfmot.quantization.keras.quantize_model
# Quantize the entire model
q_aware_model = quantize_model(model)
return q_aware_model
def train_model():
"""Training loop with QAT"""
# Build model
dtln = DTLN_Ethos_U55(lstm_units=128)
model = dtln.build_model()
# Apply quantization-aware training
q_aware_model = apply_quantization_aware_training(model)
# Compile with appropriate loss
q_aware_model.compile(
loss='mean_squared_error',
optimizer=tf.keras.optimizers.Adam(learning_rate=0.001),
metrics=['mae']
)
# Load your training data
# X_train, y_train = load_training_data()
# Train
# history = q_aware_model.fit(
# X_train, y_train,
# batch_size=32,
# epochs=50,
# validation_split=0.2
# )
return q_aware_model
if __name__ == "__main__":
model = train_model()
model.save('dtln_ethos_u55_qat.h5')
Step 5: Convert to TensorFlow Lite with 8-bit Quantization
Create convert_to_tflite.py:
import tensorflow as tf
import numpy as np
def representative_dataset_gen():
"""
Generator for representative dataset
Used for full integer quantization
"""
# Load sample audio data
# This should be representative of your inference data
for _ in range(100):
# Generate or load real audio frames
data = np.random.randn(1, 10, 257).astype(np.float32)
yield [data]
def convert_to_tflite_int8(model_path, output_path):
"""
Convert model to TFLite with INT8 quantization for Ethos-U55
"""
# Load the quantization-aware trained model
model = tf.keras.models.load_model(model_path)
# Create TFLite converter
converter = tf.lite.TFLiteConverter.from_keras_model(model)
# Enable full integer quantization
converter.optimizations = [tf.lite.Optimize.DEFAULT]
# Set representative dataset for calibration
converter.representative_dataset = representative_dataset_gen
# Ensure input/output are int8
converter.target_spec.supported_ops = [
tf.lite.OpsSet.TFLITE_BUILTINS_INT8
]
converter.inference_input_type = tf.int8
converter.inference_output_type = tf.int8
# Convert
tflite_model = converter.convert()
# Save
with open(output_path, 'wb') as f:
f.write(tflite_model)
# Print model size
print(f"Model size: {len(tflite_model) / 1024:.2f} KB")
return tflite_model
def analyze_model(tflite_path):
"""Analyze the converted model"""
interpreter = tf.lite.Interpreter(model_path=tflite_path)
interpreter.allocate_tensors()
input_details = interpreter.get_input_details()
output_details = interpreter.get_output_details()
print("\nInput Details:")
for detail in input_details:
print(f" Name: {detail['name']}")
print(f" Shape: {detail['shape']}")
print(f" Type: {detail['dtype']}")
print(f" Quantization: {detail['quantization']}")
print("\nOutput Details:")
for detail in output_details:
print(f" Name: {detail['name']}")
print(f" Shape: {detail['shape']}")
print(f" Type: {detail['dtype']}")
print(f" Quantization: {detail['quantization']}")
if __name__ == "__main__":
# Convert model
tflite_model = convert_to_tflite_int8(
'dtln_ethos_u55_qat.h5',
'dtln_ethos_u55_int8.tflite'
)
# Analyze the converted model
analyze_model('dtln_ethos_u55_int8.tflite')
Step 6: Convert TFLite to Ethos-U55 Format
Create convert_for_ethos_u55.sh:
#!/bin/bash
# Install Vela compiler for Ethos-U55
pip install ethos-u-vela
# Convert TFLite model to Ethos-U55 format
vela \
--accelerator-config ethos-u55-256 \
--system-config Ethos_U55_High_End_Embedded \
--memory-mode Shared_Sram \
--output-dir ./ethos_u55_output \
dtln_ethos_u55_int8.tflite
echo "Conversion complete! Output in ./ethos_u55_output/"
Step 7: Deployment on Alif E7
Create alif_e7_inference.c:
#include "tensorflow/lite/micro/all_ops_resolver.h"
#include "tensorflow/lite/micro/micro_interpreter.h"
#include "tensorflow/lite/micro/micro_error_reporter.h"
#include "tensorflow/lite/schema/schema_generated.h"
// Model data (generated from xxd or similar)
extern const unsigned char dtln_model_data[];
extern const int dtln_model_data_len;
// Audio buffer parameters
#define FRAME_LEN 512
#define FRAME_SHIFT 128
#define SAMPLE_RATE 16000
namespace {
tflite::ErrorReporter* error_reporter = nullptr;
const tflite::Model* model = nullptr;
tflite::MicroInterpreter* interpreter = nullptr;
TfLiteTensor* input = nullptr;
TfLiteTensor* output = nullptr;
// Arena size - adjust based on model requirements
constexpr int kTensorArenaSize = 100 * 1024; // 100KB
uint8_t tensor_arena[kTensorArenaSize];
}
void setup_model() {
// Set up error reporter
static tflite::MicroErrorReporter micro_error_reporter;
error_reporter = µ_error_reporter;
// Load model
model = tflite::GetModel(dtln_model_data);
if (model->version() != TFLITE_SCHEMA_VERSION) {
TF_LITE_REPORT_ERROR(error_reporter,
"Model version mismatch!");
return;
}
// Set up ops resolver
static tflite::AllOpsResolver resolver;
// Build interpreter
static tflite::MicroInterpreter static_interpreter(
model, resolver, tensor_arena, kTensorArenaSize, error_reporter);
interpreter = &static_interpreter;
// Allocate tensors
TfLiteStatus allocate_status = interpreter->AllocateTensors();
if (allocate_status != kTfLiteOk) {
TF_LITE_REPORT_ERROR(error_reporter, "AllocateTensors() failed");
return;
}
// Get input and output tensors
input = interpreter->input(0);
output = interpreter->output(0);
}
void process_audio_frame(float* audio_frame, float* enhanced_frame) {
// Preprocessing: compute magnitude spectrum
// (Use CMSIS-DSP for FFT on Cortex-M55)
// Copy to input tensor (with quantization if needed)
for (int i = 0; i < input->dims->data[1]; i++) {
input->data.int8[i] = /* quantize audio_frame[i] */;
}
// Run inference
TfLiteStatus invoke_status = interpreter->Invoke();
if (invoke_status != kTfLiteOk) {
TF_LITE_REPORT_ERROR(error_reporter, "Invoke failed");
return;
}
// Get output and dequantize
for (int i = 0; i < output->dims->data[1]; i++) {
enhanced_frame[i] = /* dequantize output->data.int8[i] */;
}
// Post-processing: inverse FFT to get time-domain signal
}
// Real-time processing loop
void real_time_denoising() {
float audio_buffer[FRAME_LEN];
float enhanced_buffer[FRAME_LEN];
while (1) {
// Read audio frame from microphone (I2S/PDM)
// read_audio_frame(audio_buffer);
// Process frame
process_audio_frame(audio_buffer, enhanced_buffer);
// Output enhanced audio
// write_audio_frame(enhanced_buffer);
}
}
int main() {
// Initialize Alif E7 peripherals
// init_audio_peripherals();
// Setup model
setup_model();
// Start real-time processing
real_time_denoising();
return 0;
}
Step 8: Build Configuration for Alif E7
Create CMakeLists.txt:
cmake_minimum_required(VERSION 3.15)
project(alif_e7_voice_denoising)
# Set toolchain for Arm Cortex-M55
set(CMAKE_TOOLCHAIN_FILE arm-none-eabi.cmake)
# TensorFlow Lite Micro
add_subdirectory(tensorflow/lite/micro)
# CMSIS-DSP for FFT
find_package(CMSISDSP REQUIRED)
# Source files
set(SOURCES
alif_e7_inference.c
model_data.c
)
# Create executable
add_executable(voice_denoising ${SOURCES})
# Link libraries
target_link_libraries(voice_denoising
tensorflow-lite-micro
CMSISDSP
)
# Compiler flags for Cortex-M55
target_compile_options(voice_denoising PRIVATE
-mcpu=cortex-m55
-mfpu=fp-dp-d16
-mfloat-abi=hard
-mthumb
-O3
)
Performance Optimization Tips
1. Memory Optimization
- Use stateful LSTM for streaming inference
- Minimize arena size by quantizing all layers
- Use DTCM for frequently accessed data
2. Computation Optimization
- Leverage Ethos-U55's 256 MACs for convolutions
- Use CMSIS-DSP for FFT operations on Cortex-M55
- Pipeline audio processing with NPU inference
3. Latency Reduction
- Target < 8ms frame processing time
- Use frame_shift = 128 samples (8ms @ 16kHz)
- Optimize I/O with DMA
Testing and Validation
# test_model.py
import tensorflow as tf
import numpy as np
import soundfile as sf
import time
def test_latency(tflite_path, audio_file):
"""Test inference latency"""
interpreter = tf.lite.Interpreter(model_path=tflite_path)
interpreter.allocate_tensors()
audio, sr = sf.read(audio_file)
# Process frames
frame_len = 512
frame_shift = 128
latencies = []
for i in range(0, len(audio) - frame_len, frame_shift):
frame = audio[i:i+frame_len]
# Compute magnitude spectrum
spec = np.abs(np.fft.rfft(frame))
spec = spec[np.newaxis, np.newaxis, :]
# Set input
interpreter.set_tensor(
interpreter.get_input_details()[0]['index'],
spec.astype(np.float32)
)
# Time inference
start = time.time()
interpreter.invoke()
latency = (time.time() - start) * 1000 # ms
latencies.append(latency)
print(f"Average latency: {np.mean(latencies):.2f} ms")
print(f"Max latency: {np.max(latencies):.2f} ms")
print(f"Real-time factor: {np.mean(latencies) / 8:.2f}x")
if __name__ == "__main__":
test_latency('dtln_ethos_u55_int8.tflite', 'test_audio.wav')
Alternative Models to Consider
If DTLN doesn't meet requirements, consider:
- RNNoise - Very lightweight, excellent for embedded
- Conv-TasNet - Better quality but more compute
- U-Net based - Good quality/performance balance
- CleanUNet - State-of-art but may need more optimization
Resources
- Alif Semiconductor Documentation: https://alifsemi.com/
- Arm Ethos-U55 Guide: https://developer.arm.com/
- TFLite Micro: https://www.tensorflow.org/lite/microcontrollers
- DTLN Repository: https://github.com/breizhn/DTLN
- Vela Compiler: https://pypi.org/project/ethos-u-vela/
Troubleshooting
Common Issues
- Model too large: Reduce LSTM units, use more aggressive quantization
- Latency too high: Reduce frame length, optimize FFT
- Poor quality: Increase training data, adjust quantization calibration
- Memory overflow: Reduce tensor arena size, use external PSRAM
Next Steps
- Train on your specific noise conditions
- Benchmark on actual hardware
- Optimize based on profiling results
- Fine-tune hyperparameters for your use case