DSP & Signal Processing
Signals/systems theory, digital filter design, spectral analysis, adaptive DSP, FPGA implementation, and ML for signals.
280 topics · 0 with articles
Intern Engineer· 40
- Aliasing Effect and Anti-Aliasing Filter Need
- Analytic Signal and Hilbert Transform
- Autocorrelation and Power Spectral Density
- BIBO Stability from Pole Locations
- Continuous vs. Discrete Time Signals
- Correlation Receiver for Signal Detection
- Cross-Correlation and Cross-PSD
- DCT Properties and Applications (JPEG, MP3)
- Discrete Cosine Transform (DCT) Definition
- Discrete-Time Convolution and Correlation
- Energy and Power Signals
- Envelope Detection via Hilbert Transform
- Fourier Series: Trigonometric and Exponential
- Fourier Transform: Definition and Properties
- Frequency Response H(ejw) of DT System
- Group Delay and Phase Delay
- Ideal Sampling and Reconstruction
- Impulse Response and Step Response
- Instantaneous Frequency Estimation
- Inverse Fourier Transform and Synthesis
- Inverse Z-Transform via Partial Fraction
- Laplace Transform and Transfer Function H(s)
- Linear Convolution vs. Circular Convolution
- Magnitude and Phase Response Plotting
- Matched Filter Theory and Derivation
- Modulation: AM, FM, PM in DSP
- Neyman-Pearson Lemma and LRT
- Number Theoretic Transform (NTT)
- Optimal Bayesian Detection Theory
- Optimal Transport and Wasserstein Distance for DSP
- Parseval Theorem for Energy Calculation
- Periodic and Aperiodic Signals
- Phase Distortion and Linear Phase Filter
- Practical Sample-and-Hold Effects
- Sampling Theorem: Nyquist Rate and Condition
- Signal Detection: Binary Hypothesis Test
- Signal Energy Spectral Density
- System Properties: Linearity, Causality, BIBO Stability
- Unit Impulse, Unit Step, Ramp Functions
- Z-Transform: Definition, ROC, Properties
Junior Engineer· 40
- Adaptive Equalizer FIR Filter Design
- All-Pass Filter for Phase Equalization
- Bessel Filter: Maximally Flat Group Delay
- Bilinear Transform (BLT) and Frequency Warping
- Butterworth IIR: Maximally Flat Magnitude
- CIC (Cascaded Integrator-Comb) Filter
- Cascade Biquad Section (SOS) Implementation
- Chebyshev Type I and Type II IIR Design
- Comb Filter and Resonator Design
- Constrained Least-Squares FIR Design
- Cosine Modulated Filter Bank (CMFB)
- Digital Differentiator Design
- Direct Form I and II IIR Realization
- Elliptic (Cauer) Filter Design
- Equiripple FIR with Chebyshev Approximation
- FIR Filter Order Estimation (Harris Rule)
- FIR Filter: Definition and Properties
- FIR Linear Phase Conditions (Type I-IV)
- FIR Windowing: Rectangular Window
- Finite Word-Length Effects: Overflow, Rounding
- Fixed-Point IIR Implementation Considerations
- Frequency Sampling FIR Design Method
- Half-Band Filter for Efficient Decimation
- Hann, Hamming, Blackman Windows
- Hilbert Transformer FIR Design
- IIR Design via Analog Prototype Method
- IIR Filter Stability and Limit Cycles
- Impulse Invariance Method for IIR
- Kaiser Window and Beta Parameter Selection
- Lattice Filter Structure
- Maximum Flat FIR (Maxflat) Design
- Multirate: Decimation by Integer M
- Multirate: Interpolation by Integer L
- Notch Filter for Power Line Noise (50/60 Hz)
- Parks-McClellan / Remez Exchange Algorithm
- Polyphase FIR Structure for Efficiency
- Quantization Effects in IIR: Coefficient Sensitivity
- Sample Rate Conversion: Fractional Rate L/M
- Transposed Direct Form II (Numerically Stable)
- Variable-Length FIR for Run-Time Tuning
Middle Engineer· 80
- 2D FFT for Image and Radar Processing
- AR Model Spectral Estimation (Yule-Walker)
- ARMA Model Spectral Estimation
- Acoustic Echo Cancellation (AEC) with LMS
- Adaptive Noise Cancellation (ANC) Architecture
- Bartlett Method and Periodogram Averaging
- Bit-Reversal Permutation in FFT
- Blind Equalization: CMA Algorithm
- Blind Source Separation (BSS)
- Chirp Z-Transform (CZT) for Zoom FFT
- Cohen Class of Time-Frequency Distributions
- Compressed Sensing: Restricted Isometry Property
- Cooley-Tukey FFT Butterfly Diagram
- DFT Properties: Linearity, Shift, Circular Convolution
- Discrete Fourier Transform (DFT): Definition and Matrix
- Discrete Wavelet Transform (DWT): Mallat
- Double-Talk Detection in AEC
- EM for Gaussian Mixture Separation
- ESPRIT DOA Algorithm
- Empirical Mode Decomposition (EMD)
- Expectation Maximization (EM) for DSP
- Extended Kalman Filter (EKF) for Nonlinear
- FFT of Real-Valued Signal: Half-Spectrum Trick
- Factor Graphs and Belief Propagation
- FastICA Algorithm
- Fractional Fourier Transform (FrFT) Application
- Goertzel Algorithm for Single Frequency
- Hilbert-Huang Transform (HHT)
- In-Place FFT: Memory-Efficient Implementation
- Independent Component Analysis (ICA)
- Joint Diagonalization for BSS
- Kalman Filter Prediction and Update Steps
- Kalman Filter: State-Space Model and Equations
- L1-Norm Minimization (LASSO, Basis Pursuit)
- LMS Algorithm: Derivation and Convergence
- LMS Step Size Selection and Misadjustment
- Leaky LMS for Drift Prevention
- M-Estimation for Robust Statistics
- MUSIC Algorithm for DOA Estimation
- Maximum Entropy Method (MEM) Spectral Est.
- Mixed-Radix FFT Implementation
- Multitaper PSD Estimation
- Non-Stationary Analysis: VMD
- Normalized LMS (NLMS) Algorithm
- Online Learning and Recursive Estimation
- Optimum Linear Predictor (FIR Wiener Filter)
- Orthogonal Matching Pursuit (OMP)
- Particle Filter for Non-Gaussian Noise
- Power Spectral Density Non-Parametric Estimation
- RLS Algorithm: Exponential Forgetting Factor
- Radix-2 Decimation-in-Frequency (DIF) FFT
- Radix-2 Decimation-in-Time (DIT) FFT
- Radix-4 FFT for Hardware Efficiency
- Robust Estimation and Outlier Rejection
- S-Transform (Stockwell Transform)
- SOBI (Second-Order Blind Identification)
- STFT Time-Frequency Resolution Trade-off
- Short-Time Fourier Transform (STFT)
- Sparse FFT for Compressive Sensing
- Sparse Signal Recovery and Dictionary Learning
- Spectrogram: STFT Magnitude Squared
- Split-Radix FFT Algorithm
- Statistical Signal Detection: LRT
- Subspace Methods: MUSIC, ESPRIT, Matrix Pencil
- Synchrosqueezing Wavelet Transform
- Tensor Decomposition for Multi-Way Signal
- Turbo Decoding as Message Passing
- Unscented Kalman Filter (UKF)
- Variational Autoencoder for Signal Generation
- Variational Inference for Bayesian DSP
- Wavelet Packet Decomposition
- Wavelet Transform: Continuous (CWT)
- Welch Method for Power Spectral Density
- Wiener Filter: MMSE Estimation
- Wiener-Hopf Equation Solution
- Wigner-Ville Distribution (WVD)
- Window Selection: Resolution vs. Leakage
- Windowed FFT and Spectral Leakage Analysis
- Winograd Small-N FFT Algorithms
- Zero-Padding and Frequency Resolution
Senior Engineer· 40
- AI-DSP Co-Design Methodology
- ARM Cortex-M DSP Intrinsics (CMSIS-DSP)
- ARM Cortex-M55 Helium MVE for DSP
- Analog Devices SHARC Processor Architecture
- Block RAM (BRAM) for Coefficient Storage
- CORDIC Algorithm for Trigonometric on FPGA
- Custom DSP ASIC RTL to GDS Flow
- DDC: Digital Down Converter on FPGA
- DUC: Digital Up Converter on FPGA
- Dedicated FFT Accelerator IP Core
- Edge DSP Inference with SNPE / OpenVINO
- FPGA AI Inference: Vitis AI, OpenVINO
- FPGA DSP48E2 Slice: Multiply-Accumulate
- FPGA Floating-Point DSP (IEEE 754 Core)
- FPGA-Based Real-Time Radar DSP
- Fixed-Point Overflow and Saturation on FPGA
- GNU Radio: Software-Defined Radio Framework
- GPU-Accelerated DSP: CUDA cuFFT
- High-Level Synthesis (HLS) for DSP
- LibROSA: Audio and Music Analysis
- MATLAB Signal Processing Toolbox
- MFCC Extraction Optimized on DSP MCU
- Mixed-Precision DSP on Heterogeneous SoC
- NCO (Numerically Controlled Oscillator)
- Neuromorphic DSP Architectures (Intel Loihi)
- OpenCL for Heterogeneous DSP
- Photonic DSP for Ultra-High-Speed
- Pipeline Architecture for FFT on FPGA
- Polyphase FIR Decimator on FPGA DSP48
- PySDR for SDR and DSP Prototyping
- Python SciPy.signal for DSP Prototyping
- Quantum Signal Processing Algorithms
- Radix-2 FFT FPGA Implementation
- SNPE (Qualcomm DSP) for Edge Inference
- Silicon Labs DSP Peripherals (LESENSE)
- Systolic Array Architecture for FIR FPGA
- TMS320C28x Fixed-Point DSP for Control
- Texas Instruments C6000 DSP Architecture
- Xilinx Vitis AI for FPGA Inference
- torchaudio: PyTorch Audio Processing
Staff Engineer· 40
- 5G NR PHY DSP Chain: NR-PDSCH Processing
- AAC: Filterbank, TNS, Prediction
- Acoustic Echo Cancellation (AEC)
- Active Noise Cancellation (ANC) for Headphones
- Audio Equalization: Parametric, Graphic, Shelving
- Audio Sampling Rates: 44.1kHz, 48kHz, 96kHz, 192kHz
- Beamforming: Delay-and-Sum Algorithm
- CRC Generation and Verification
- Channel Coding: LDPC Decode Algorithm
- Channel Estimation: LS, MMSE for OFDM
- Convolution Reverb and Room Impulse Response
- DTMF Encoder/Decoder with Goertzel Algorithm
- FMCW Radar Range and Doppler Processing
- LC3 (LE Audio) Codec for Bluetooth
- LCMV Beamformer for Multi-Constraint
- MIMO Detection: ZF, MMSE, Sphere Decoder
- MP3 Encoding: Subband Filter + MDCT
- MVDR Minimum Variance Beamformer
- Noise Reduction: Wiener Filter, Spectral Subtraction
- OFDM Channel Estimation and Equalization
- OFDM PAPR Reduction: Clipping, SLM, PTS
- OFDM: Subcarrier Spacing, Cyclic Prefix, Pilot
- OTFS (Orthogonal Time-Frequency Space) Waveform
- Opus Codec: SILK + CELT Architecture
- PCM and ADPCM Audio Coding (G.711, G.726)
- Perceptual Audio Coding Principles
- Pitch Detection Algorithms (AMDF, CREPE)
- Polar Code Successive Cancellation Decoder
- Psychoacoustic Model: ISO 226, Masking
- Radar Ambiguity Function Analysis
- Radar Pulse Compression: Matched Filter
- Radar Signal Processing: CFAR Detector
- Reed-Solomon Decoder for Burst Error Correction
- Room Acoustics: Reverberation Time (RT60)
- Software Defined Radio: RTL-SDR, USRP
- Superdirectional Microphone Array Design
- Synchronization: Timing Offset, CFO Correction
- Turbo Decoder: MAP, Log-MAP, Max-Log-MAP
- Viterbi Algorithm for Convolutional Codes
- Voice Activity Detection (VAD) Algorithms
Distinguished Engineer· 40
- Audio Event Detection with PANNs
- CNN for 1D Signal Classification
- Continual Learning for Non-Stationary Signals
- DNN-Based Channel Estimation for 5G
- DeepReceiver: ML Receiver for OFDM
- Domain Adaptation for Signal Recognition
- End-to-End ASR: CTC Loss and Beam Search
- Explainable AI for DSP Model Diagnostics
- Few-Shot Signal Classification
- Foundation Models for Time-Series (Moirai)
- GNN for Sensor Network DSP
- Generative AI for Synthetic Signal Augmentation
- HiFi-GAN Vocoder Architecture
- HuBERT and data2vec for Audio Pre-Training
- Keyword Spotting (KWS) with TFLite Micro
- LSTM / GRU for Time Series Modeling
- MAML Meta-Learning for Rapid Signal Adaptation
- ML-Based AEC and Noise Suppression (RNNoise)
- MusicGen: LM for Music Generation
- Neural Audio Codec: EnCodec, DAC
- Neural Beamforming with Attended Speech
- Neuromorphic Auditory Processing with SNN
- Physics-Informed Neural Network for Wave
- SEGAN: Speech Enhancement with GAN
- Self-Supervised Contrastive Learning for DSP
- Semi-Supervised Learning for Signal Labeling
- Signal Classification with 2D CNN (Spectrogram)
- Sound Event Localization and Detection (SELD)
- SoundStream: Neural Audio Codec
- Speech Enhancement with U-Net
- TFLite Micro MFCC + CNN on Cortex-M
- TTS: Tacotron 2, FastSpeech 2, VITS
- Temporal Convolutional Network (TCN)
- Transfer Learning: ImageNet to Spectrogram
- Transformer for Time-Series (TFT)
- Uncertainty Estimation in ML for Safety DSP
- Wake-Word Detection on MCU (Edge Impulse)
- Wav2Vec 2.0: Self-Supervised Speech
- WaveNet: Dilated Causal CNN for Audio
- Whisper: Large-Scale ASR Encoder-Decoder