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This document outlines project pipelines for two audio-based machine learning applications implemented in C: Environmental Noise Cancellation (ENC) and Audio Source Separation. It details the steps, tools, and libraries required for each project, including audio I/O, digital signal processing, and machine learning techniques. Additionally, it provides guidance on project setup, feature extraction, and model inference, emphasizing the importance of various C libraries and mathematical operations.

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0% found this document useful (0 votes)
15 views10 pages

Project

This document outlines project pipelines for two audio-based machine learning applications implemented in C: Environmental Noise Cancellation (ENC) and Audio Source Separation. It details the steps, tools, and libraries required for each project, including audio I/O, digital signal processing, and machine learning techniques. Additionally, it provides guidance on project setup, feature extraction, and model inference, emphasizing the importance of various C libraries and mathematical operations.

Uploaded by

manimaran
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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project.

md 2025-07-25

Detailed Project Pipelines for Audio ML in C


This document outlines the project pipelines for two audio-based machine learning applications
implemented in C: Audio Source Separation and Audio Event Classi cation.

Project 1: Environmental Noise Cancellation (ENC) Project Pipeline in C

This pipeline outlines the steps to implement a basic Spectral Subtraction algorithm for Environmental Noise
Cancellation in C, along with relevant open-source C libraries and GitHub repositories.

Project Goal:

To create a C application that takes a noisy audio input, estimates the noise, and produces a cleaner audio
output by applying the Spectral Subtraction algorithm.

1. Project Setup & Version Control

Description: Initialize your project repository and set up a basic build system.
Tools:
Git: For version control.
CMake / Make: For building your C project.
GitHub Reference: Your own new repository (e.g., my-c-enc-project).

2. Audio Input/Output (I/O)

Description: Read audio data from a le (e.g., WAV) or a microphone and write processed audio to a
le or speaker.
Recommended C Libraries:
libsnd le: For reading and writing common audio le formats like WAV, AIFF, FLAC. It's robust
and widely used.
GitHub: https://github.com/libsnd le/libsnd le
miniaudio: A single- le, cross-platform audio playback and capture library. Excellent for real-
time microphone input and speaker output.
GitHub: https://github.com/mackron/miniaudio
PortAudio: Another popular cross-platform audio I/O library.
GitHub: https://github.com/PortAudio/portaudio

3. Core Digital Signal Processing (DSP) Libraries

Description: Essential for performing Fast Fourier Transform (FFT), Inverse FFT (IFFT), and potentially
other signal manipulations.
Recommended C Libraries:
KissFFT: A fast, small, and self-contained mixed-radix FFT library. It's often preferred for
embedded systems due to its simplicity and low memory footprint.
GitHub: https://github.com/mborgerding/kiss t
FFTW (Fastest Fourier Transform in the West): Highly optimized and very fast, but can be
more complex to integrate than KissFFT. Best for desktop/server applications where maximum
performance is critical.

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Website (main source): http://www. tw.org/ (Source code typically downloaded from
here, not a direct GitHub repo).

4. Framing and Windowing

Description: Divide the continuous audio stream into small, overlapping frames. Apply a window
function (e.g., Hann, Hamming) to each frame to reduce spectral leakage before FFT.
Implementation: These are typically implemented directly in your C code using math.h functions.
Hann Window Formula: w[n]=0.5−0.5cos(N−12πn)
Relevant C Concepts: Array manipulation, loops, math.h for cos().

5. Noise Estimation

Description: During an initial "noise-only" period (e.g., the rst few seconds of audio before speech
begins), estimate the average noise spectrum. This average spectrum will be subtracted from
subsequent noisy frames.
Implementation:
1. Collect several frames of pure noise.
2. For each noise frame:
Apply windowing.
Perform FFT.
Calculate the magnitude spectrum.
3. Average the magnitude spectra of all noise frames to get the
estimated_noise_spectrum_magnitude.
Relevant C Concepts: Loops, array averaging, magnitude calculation (sqrt(real^2 + imag^2)).

6. Spectral Subtraction Implementation (Core ENC Logic)

Description: For each incoming noisy audio frame (containing both speech and noise):
1. Apply windowing.
2. Perform FFT to get the noisy signal's complex spectrum.
3. Calculate the magnitude and phase of the noisy spectrum.
4. Apply the spectral subtraction formula to the magnitude:
Clean Magnitude[k]=max(Noisy Magnitude[k]2−α⋅Noise Magnitude[k]2,β⋅Noisy
Magnitude[k]2)
Where α is the over-subtraction factor and β is the spectral oor.
5. Reconstruct the complex spectrum using the Clean Magnitude and the Original Phase (phase is
usually preserved in simple spectral subtraction).
6. Perform IFFT to convert the cleaned spectrum back to the time domain.
Relevant C Concepts: Loops, array manipulation, math.h (sqrt, pow, fmax, atan2, cos, sin).
MAC Operations Focus: FFT/IFFT, magnitude calculations, and the subtraction/reconstruction steps
are all highly MAC-intensive.

7. Overlap-Add (OLA)

Description: Since frames are processed with overlap, the IFFT output of each frame needs to be
correctly added to overlapping sections of the output bu er to reconstruct a continuous, smooth
audio signal.

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Implementation: Requires careful management of input and output bu ers, adding overlapping
portions of processed frames.
Relevant C Concepts: Bu er management, array indexing, additions.

8. Testing and Evaluation

Description: Test your ENC application with various types of noisy audio.
Tools:
Audacity / Praat: For visualizing waveforms and spectrograms to qualitatively assess noise
reduction.
Objective Metrics (C Implementation): You could implement simple metrics like Signal-to-
Noise Ratio (SNR) improvement in C to quantitatively evaluate performance.
Test Audio: Use publicly available noisy speech datasets (e.g., from research papers) or create
your own.

This pipeline provides a structured approach to building an ENC application in C, highlighting the key stages
and the role of various C libraries and mathematical operations. Remember that a full- edged, production-
grade ENC system would involve more advanced algorithms (e.g., Wiener ltering, deep learning-based
methods) and more sophisticated real-time audio management.

Project 2: Audio Source Separation (e.g., for Karaoke or Speech Enhancement)

Project Goal: To separate di erent sound sources (e.g., vocals from music, or speech from noise) from a
mixed audio track. This is a complex task, and a "simple" C implementation will likely focus on a traditional
DSP approach or the inference of a very lightweight, pre-trained model.

Key Concepts/Algorithms:

Short-Time Fourier Transform (STFT) / Inverse STFT (ISTFT): Decomposing audio into overlapping
time-frequency frames.
Magnitude and Phase: Separating the amplitude and phase information in the frequency domain.
Masking: Creating a "mask" (e.g., binary or soft mask) in the frequency domain to isolate desired
components.
Wiener Filtering (Traditional DSP approach): An adaptive lter that estimates the original signal by
minimizing the mean square error between the estimated and original signals. This is a common non-
AI method for speech enhancement that can be implemented in C.
Non-Negative Matrix Factorization (NMF): A more advanced traditional ML technique for source
separation that can be implemented in C.
Deep Learning Inference (Advanced): For state-of-the-art separation, deep learning models (e.g., U-
Net architectures like Demucs) are used. A C project would focus on implementing the inference of a
pre-trained, lightweight version of such a model, typically by porting it via frameworks like ONNX
Runtime or TensorFlow Lite.

Detailed Pipeline Steps:

1. Project Setup & Dependencies:


Initialize Git repository.
Set up CMake/Make build system.

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Audio I/O: Integrate libsnd le for WAV le reading/writing. For real-time, consider miniaudio or
PortAudio.
FFT/IFFT Library: Integrate KissFFT (recommended for simplicity and embedded focus) or
FFTW (for high performance).
2. Audio Pre-processing (Framing & Windowing):
Read the mixed audio signal.
Divide the audio into short, overlapping frames (e.g., 20-40ms frames with 50% overlap).
Apply a window function (e.g., Hann window: w[n]=0.5−0.5cos(frac2pinN−1)) to each frame.
Relevant C Concepts: Array manipulation, loops, math.h for cos().
3. STFT (Time-to-Frequency Domain):
For each windowed frame:
Perform FFT to transform the time-domain signal into a complex frequency spectrum.
Calculate the magnitude spectrum (∣X[k]∣=sqrttextReal[k]2+textImag[k]2) and phase
spectrum (textPhase[k]=textatan2(textImag[k],textReal[k])).
Relevant C Concepts: KissFFT/FFTW usage, complex number arithmetic (structs), math.h for
sqrt(), atan2().
MAC Operations Focus: FFT is highly MAC-intensive.
4. Source Separation Logic (Core Algorithm):
Option A: Spectral Subtraction (for Speech Enhancement/Noise Reduction):
Estimate the noise spectrum (e.g., during silent periods or using a VAD).
For each noisy frame, subtract the estimated noise power (magnitude squared) from the
noisy signal's power spectrum. Apply a spectral oor.
Relevant C Concepts: Loops, array operations, math.h (sqrt, fmax).
MAC Operations Focus: Magnitude calculations, multiplications, subtractions.
Option B: Simple Masking (e.g., Ideal Binary Mask):
(Requires prior knowledge or estimation of source characteristics). For example, if you
know target speech is in certain frequency bands, you could create a binary mask.
Clean_Magnitude[k] = Mask[k] * Noisy_Magnitude[k]
Relevant C Concepts: Array element-wise multiplication.
Option C: Inference of a Pre-trained Deep Learning Model (Advanced):
Load a pre-trained model (e.g., ONNX format) into a C/C++ inference runtime (like ONNX
Runtime or TFLite).
Feed the magnitude (and potentially phase) spectrograms as input to the model.
The model outputs a mask or directly the separated sources' spectrograms.
Relevant C Concepts: Integration with external ML inference libraries.
MAC Operations Focus: Extremely high due to neural network computations
(convolutions, matrix multiplications).
5. ISTFT (Frequency-to-Time Domain):
Combine the cleaned magnitude spectrum with the original phase spectrum to reconstruct the
complex spectrum of the separated source.
Perform IFFT to transform the complex spectrum back to the time domain.
Relevant C Concepts: KissFFT/FFTW usage, math.h for cos(), sin().
MAC Operations Focus: IFFT is highly MAC-intensive.
6. Overlap-Add (OLA):
Correctly combine the overlapping, processed time-domain frames to reconstruct the
continuous separated audio signal.

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Relevant C Concepts: Bu er management, array indexing, additions.


7. Audio Output:
Write the separated audio signal to a new WAV le (libsnd le) or play it in real-time
(miniaudio/PortAudio).

GitHub Reference:

You would create your own repository for this project.


Refer to the GitHub repositories of the chosen libraries (e.g., libsnd le, miniaudio, kiss t).
For advanced deep learning inference in C, explore:
ONNX Runtime: https://github.com/microsoft/onnxruntime (Provides C API)
TensorFlow Lite Micro:
https://github.com/tensor ow/tensor ow/tree/master/tensor ow/lite/micro (For embedded
systems)

Project 3: Audio Event Classi er (using MLP or KNN)

Project Goal: To classify short segments of audio into prede ned categories (e.g., "Speech", "Music",
"Silence", "Clap", "Whistle"). This involves extracting relevant features and feeding them into a simple
machine learning model.

Key Concepts/Algorithms:

Feature Extraction: Converting raw audio into numerical features that represent its characteristics
(e.g., Short-Time Energy, Zero-Crossing Rate, Spectral Centroid, MFCCs).
Multi-Layer Perceptron (MLP): A simple feedforward neural network with an input layer, one or more
hidden layers, and an output layer. It learns non-linear relationships between features and classes.
K-Nearest Neighbors (KNN): A non-parametric, instance-based learning algorithm that classi es a
new data point based on the majority class among its 'k' nearest neighbors in the feature space.
Supervised Learning: The model is "trained" on labeled audio data (e.g., examples of "speech" and
"noise" with their corresponding labels). For a C project, this training is typically done o ine in
Python, and only the trained model's parameters are loaded into the C application for inference.

Detailed Pipeline Steps:

1. Project Setup & Dependencies:


Initialize Git repository.
Set up CMake/Make build system.
Audio I/O: Integrate libsnd le for WAV le reading. For real-time, miniaudio or PortAudio.
FFT Library: Integrate KissFFT if using spectral features like Spectral Centroid or MFCCs.
2. Audio Pre-processing (Framing & Windowing):
Read the audio signal.
Divide the audio into short, overlapping frames.
Apply a window function (e.g., Hann window) to each frame.
Relevant C Concepts: Array manipulation, loops, math.h.
3. Feature Extraction:
For each windowed frame, compute a set of numerical features.
Common Features (implementable in C):

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Short-Time Energy (STE): sumx[n]2. (Sum of squares).


Zero-Crossing Rate (ZCR): Number of sign changes.
Spectral Centroid: (Requires FFT) fracsumkcdot∣X[k]∣sum∣X[k]∣.
Mel-Frequency Cepstral Coe cients (MFCCs): (More complex, requires Mel lter banks
and DCT).
Relevant C Concepts: Loops, arithmetic, KissFFT for spectral features, math.h (sqrt, log, cos, sin
for MFCCs).
MAC Operations Focus: STE, Spectral Centroid, and MFCCs involve many multiplications and
additions.
4. Model Loading (Inference Phase):
O ine Training: This is a crucial step typically done outside the C project. You would use
Python with libraries like librosa (for features) and scikit-learn (for KNN/MLP) or
TensorFlow/PyTorch (for MLP) to train your model on a labeled dataset.
Parameter Export: After training, save the learned parameters (e.g., MLP weights/biases, or
KNN training data) to a simple text or binary le.
C Loading: In your C application, implement functions to read these parameters from the le
into C arrays/structs.
Relevant C Concepts: File I/O, parsing data into arrays.
5. Classi cation (Core ML Logic):
For each set of extracted features:
Option A: Multi-Layer Perceptron (MLP) Inference:
Implement the forward pass of the MLP:
Input Layer -> Hidden Layer: Perform weighted sums (sumw_ix_i+b) and apply
activation functions (e.g., sigmoid: \frac{1}{1+e^{-x}}).
Hidden Layer -> Output Layer: Repeat weighted sums and activation.
The output layer provides probabilities or scores for each class.
Relevant C Concepts: Array/matrix multiplication (loops), math.h (exp for sigmoid).
MAC Operations Focus: Matrix multiplications are highly MAC-intensive.
Option B: K-Nearest Neighbors (KNN) Inference:
For the current input feature vector, calculate its Euclidean distance to all stored training
examples.
Sort distances and identify the 'k' nearest neighbors.
Determine the class by majority vote among these 'k' neighbors.
Relevant C Concepts: Loops, array manipulation, math.h (sqrt, pow).
MAC Operations Focus: Distance calculations involve many subtractions, multiplications
(squaring), and additions.
6. Decision & Output:
Based on the classi er's output, determine the most likely audio event category.
Print the classi cation result to the console or trigger an action (e.g., light an LED, play a sound).

GitHub Reference:

You would create your own repository for this project.


Refer to the GitHub repositories of the chosen libraries (e.g., libsnd le, miniaudio, kiss t).
For simple neural network implementation in C:
Genann: https://github.com/codeplea/genann (A very simple C ANN library).
For o ine training in Python:

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scikit-learn (for KNN/MLP): https://github.com/scikit-learn/scikit-learn


librosa (for audio features): https://github.com/librosa/librosa

K-Nearest Neighbors (KNN) Audio Event Classi er Project Pipeline in C

This pipeline details the implementation of a real-time audio event classi er in C, speci cally using the K-
Nearest Neighbors (KNN) algorithm. This project is well-suited for DSP devices due to KNN's relatively
straightforward implementation and its reliance on fundamental arithmetic operations (which map well to
DSP's MAC units).

Project Goal: To develop a C application that continuously processes live audio, extracts relevant features,
and classi es detected sound events into prede ned categories (e.g., "Clap", "Whistle", "Background Noise")
in real-time using a pre-trained KNN model.

Key Concepts/Algorithms:

Real-time Audio Capture: Obtaining audio samples from a microphone continuously.


Short-Time Audio Analysis: Processing audio in small, overlapping frames.
Feature Extraction: Deriving numerical features from each audio frame that are discriminative for the
target events. Common choices include:
Short-Time Energy (STE): Good for detecting the presence of sound.
Zero-Crossing Rate (ZCR): Useful for distinguishing noisy/unvoiced sounds from more periodic
ones.
Spectral Centroid: Indicates the "brightness" or "darkness" of a sound.
K-Nearest Neighbors (KNN): A non-parametric, instance-based learning algorithm. It classi es a new
data point by nding the 'k' closest points (neighbors) in the training data and assigning the majority
class among them.
Euclidean Distance: The most common metric for measuring "closeness" between feature vectors.
Supervised Learning (O ine Training): The KNN model "learns" by simply remembering all its
training data points and their labels. For a C project, this training is performed o ine (typically in
Python), and the entire labeled dataset is loaded into the C application for inference.

Detailed Pipeline Steps:

1. Project Setup & Dependencies:


Initialize a new Git repository for your project.
Set up a build system (e.g., CMake or Make le) to compile your C code and link necessary
libraries.
Audio I/O Library:
miniaudio: Highly recommended for real-time, cross-platform microphone input and
speaker output due to its single- le nature and simplicity.
GitHub: https://github.com/mackron/miniaudio
Alternatively, PortAudio is a robust choice.
GitHub: https://github.com/PortAudio/portaudio
FFT Library (if using spectral features like Spectral Centroid):
KissFFT: Small, fast, and easy to integrate for Fast Fourier Transform operations.
GitHub: https://github.com/mborgerding/kiss t
Standard C Libraries: math.h for mathematical functions (sqrt, pow, abs), stdlib.h, stdio.h,
stdbool.h.
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2. Audio Capture & Framing:


Real-time Input: Con gure the chosen audio I/O library (miniaudio or PortAudio) to capture
audio from the default microphone. This typically involves setting up a callback function that
receives chunks (frames) of audio data at regular intervals.
Frame Size & Overlap: De ne parameters for the audio frames (e.g., FRAME_SIZE = 512
samples, OVERLAP_SIZE = 256 samples for 50% overlap). This determines the granularity of
your analysis.
Bu ering: Implement circular bu ers or similar mechanisms to manage the continuous stream
of incoming audio frames, crucial for real-time processing.
Relevant C Concepts: Pointers, arrays, volatile keyword (for shared memory with DMA/ISR),
interrupt service routines (ISRs) for bu er completion.
3. Audio Pre-processing (Windowing):
For each captured audio frame, apply a window function (e.g., Hann window). This helps to
reduce spectral leakage when performing FFT and prepares the frame for feature extraction.
Hann Window Formula: w[n]=0.5−0.5cos(N−12πn)
Relevant C Concepts: Array iteration, oating-point or xed-point arithmetic (if using xed-
point DSP). math.h for cos().
4. Feature Extraction:
For each windowed audio frame, compute a set of numerical features. These features should be
chosen to e ectively distinguish between your target audio events.
Short-Time Energy (STE): Calculate the sum of the squares of the samples within the frame.
Formula: STE=∑n=0N−1x[n]2
MAC Operations Focus: Direct multiplications and accumulations.
Zero-Crossing Rate (ZCR): Count the number of times the audio signal changes sign within the
frame.
Formula: Count n where x[n]⋅x[n−1]<0.
Spectral Centroid: (Requires FFT) Calculate the weighted average of the frequencies in the
magnitude spectrum.
Steps:
1. Perform FFT on the windowed frame using KissFFT.
2. Calculate the magnitude of each frequency bin: ∣X[k]∣=Real[k]2+Imag[k]2.
3. Apply the Spectral Centroid formula: SC=∑k=0N−1∣X[k]∣∑k=0N−1k⋅∣X[k]∣.
MAC Operations Focus: FFT is highly MAC-intensive. The centroid calculation involves
more multiplications and additions.
Normalization: Normalize the extracted features (e.g., scale to a 0-1 range or standardize) to
ensure all features contribute equally to distance calculations. The normalization parameters
(min/max or mean/std dev for each feature) must be learned during o ine training and loaded
into the C application.
Relevant C Concepts: Loops, array operations, KissFFT API usage, math.h (sqrt).
5. Model Training (O ine - Typically in Python):
This step is performed outside your C application.
Data Collection: Record a diverse dataset of your target audio events (e.g., hundreds of claps,
hundreds of whistles, hundreds of segments of background noise). Collect enough data to
represent variations within each class.
Labeling: Manually label each recording (e.g., "clap", "whistle", "noise").

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Feature Extraction: Use a Python script (with librosa for audio loading/features) to extract the
same features (STE, ZCR, Spectral Centroid) from your labeled dataset.
Model Training (KNN): Use scikit-learn.neighbors.KNeighborsClassi er to train a KNN model.
The "training" for KNN is simply storing the labeled feature vectors.
Parameter Export: Save the entire labeled training dataset (feature vectors and their
corresponding labels) to a simple text le (e.g., CSV) or a custom binary format that your C
application can easily parse and load. Also, save the normalization parameters.
Python Libraries:
librosa: https://github.com/librosa/librosa
scikit-learn: https://github.com/scikit-learn/scikit-learn
6. Model Loading (Inference in C):
In your C application, implement functions to read the pre-trained KNN training data (feature
vectors and labels) and normalization parameters from the le(s) you exported in the previous
step.
Store this data in appropriate C data structures (e.g., a 2D array for features, a 1D array for
labels).
Relevant C Concepts: File I/O (fopen, fread, fscanf), dynamic memory allocation (malloc, free) if
the training dataset size isn't xed.
7. Classi cation Logic (Core KNN Inference in C):
For each new, normalized feature vector extracted from a real-time audio frame:
Distance Calculation: Iterate through all stored training examples. For each training
example, calculate the Euclidean distance between the current input feature vector and
the training example's feature vector.
Formula: distance=∑i=0NUM_FEATURES−1(input_featurei−train_featurei)2
MAC Operations Focus: Distance calculations involve many subtractions,
multiplications (squaring), and additions.
Find K-Nearest Neighbors: Keep track of the 'k' training examples that have the smallest
distances to the input feature vector. You'll need to maintain a sorted list or use a min-
heap for e ciency.
Majority Vote: Among the 'k' nearest neighbors, count the occurrences of each class
label. The class with the highest count is the predicted class for the input audio frame.
Relevant C Concepts: Loops, array manipulation, sorting (or partial sorting), math.h (sqrt, pow).
8. Decision & Output:
Based on the KNN classi er's output, determine the most likely audio event category.
Implement logic to handle sequences of classi cations (e.g., only declare an event if a class is
detected for several consecutive frames to reduce false positives).
Trigger actions based on the classi cation:
Print "Clap Detected!" or "Whistle Sound!" to the console.
Light up an LED (if on an embedded system).
Play a con rmation sound.
Relevant C Concepts: Conditional statements (if/else), printf, state machines for event
detection.
9. Testing and Optimization:
Real-time Testing: Run your C application with live microphone input. Test with various claps,
whistles, and background noise.

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Performance Pro ling: Use tools like gprof (Linux) or platform-speci c pro lers (e.g., in your
DSP IDE) to identify bottlenecks. KNN's distance calculation can be computationally intensive if
the training dataset is very large.
Memory Optimization: For DSP devices, minimize dynamic memory allocation. Use static arrays
or pre-allocated bu ers. Consider xed-point arithmetic if the DSP supports it well and oating-
point is slow.
Parameter Tuning (K value): Experiment with di erent values of 'k' (e.g., 3, 5, 7) to nd the
best balance between accuracy and robustness.

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