0 views collected around this technical thread.
Fourier series, introduced by Joseph Fourier to solve the heat equation, uses sine and cosine functions to represent complex phenomena, enabling the decomposition of problems into simpler parts and finding applications across wave motion, physics, finance, and modern digital signal processing.
This article explains the mathematical definition of convolution, walks through discrete and continuous examples, demonstrates its use in signal smoothing with moving averages, and surveys its wide-ranging applications in signal processing, communications, computer vision, seismology, medical imaging, and statistics.
This article introduces Fourier series, explains how any musical note can be decomposed into its constituent frequencies, demonstrates generating a composite waveform with Python, computes its Fourier coefficients, visualizes the results, and discusses practical applications such as audio analysis, instrument classification, and digital music compression.
At ICASSP 2023, Kuaishou’s audio team presented a low‑latency hierarchical fusion network for full‑band acoustic echo cancellation, detailing its multi‑stage design, asymmetric windowing, loss functions, training strategy, and achieving second place in the non‑personalized AEC Challenge, with real‑world deployment results.
The article demonstrates how to boost iOS chat session page performance by refactoring heavy data‑source and business‑data processing into ReactiveObjC signals, using combineLatest and flattenMap to shift intensive tasks off the main thread, improve readability, and eliminate frame drops.
The article provides a comprehensive overview of digital watermarking, covering its fundamental concepts, security features, embedding/detection/extraction processes, major algorithm families, practical applications such as copyright protection and anti‑counterfeiting, and future research directions in multimedia information security.
This article explains Fourier series theory, demonstrates how to remove linear trends from monthly CO₂ data, and shows step‑by‑step Python code using SciPy's curve_fit to fit and predict the data with a 100‑term Fourier expansion.
Originating from three largely unknown Indian engineers in the early 1970s, the Discrete Cosine Transform (DCT) became the cornerstone of modern image and video compression standards such as JPEG and MPEG, enabling efficient lossy compression by converting spatial data to frequency components and exploiting human visual perception.
Adaptive filtering predicts time series by iteratively adjusting weight coefficients based on prediction errors, using a learning constant to control adaptation speed, making it a simple yet powerful method that leverages all historical data for continuously improving forecasts.
This article presents a non‑reference audio quality assessment method that leverages a bidirectional LSTM network to predict perceptual scores from spectral features extracted via FFT, describing the system workflow, technical advantages, data preparation, loss design, and TensorFlow implementation details.
This article presents a no‑reference audio quality assessment system that leverages a bidirectional LSTM network to extract spectral features via FFT and predict perceptual scores, describing the architecture, technical advantages, data preparation, loss design, and TensorFlow implementation.
The inaugural L3DAS22 competition, co‑hosted by Kuaishou Audio and Sapienza University, gathered nearly 50 academic and industry teams to benchmark deep‑learning‑based 3D audio signal processing, featuring tasks on multi‑channel speech enhancement and source detection, with results presented at ICASSP 2022.
The article explains Android audio visualization by decoding PCM data, applying Fourier and Fast Fourier transforms to obtain frequency spectra, comparing the built‑in Visualizer API with a custom JNI‑based implementation, and detailing FFT processing, smoothing, buffering, and Canvas rendering techniques for smooth, performant visual effects.
Tencent Cloud Gaming Multimedia Engine (GME) tackles the intense, non‑stationary background noise of internet cafés by employing a specialized voice‑activity detection algorithm combined with mean removal, low‑pass and numerical filtering, enabling it to isolate a primary speaker’s voice and suppress surrounding chatter, clicks, and music for clear in‑game communication.
This article describes a C# program that continuously captures microphone audio, uses volume thresholds derived from Fourier analysis to automatically start and stop recording, and provides the full configuration and source code needed to build a smart recorder for capturing dream speech.