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Related Concept Videos

Upsampling01:22

Upsampling

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Managing signal sampling rates is essential in digital signal processing to maintain signal integrity. A decimated signal, characterized by a reduced frequency range due to its lower sampling rate, can be upsampled by inserting zeros between each sample. This upsampling process expands the original spectrum and introduces repeated spectral replicas at intervals dictated by the new Nyquist frequency. To refine this zero-inserted sequence, it is passed through a lowpass filter with a cutoff...
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When considering a sampled sequence with zero values between sampling instants, one can replace it by taking every N-th value of the sequence. At these integer multiples of N, the original and sampled sequences coincide. This process, known as decimation, involves extracting every N-th sample from a sequence, thereby creating a more efficient sequence.
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Sampling Continuous Time Signal01:11

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In signal processing, a continuous-time signal can be sampled using an impulse-train sampling technique, followed by the zero-order hold method. Impulse-train sampling involves the use of a periodic impulse train, which consists of a series of delta functions spaced at regular intervals determined by the sampling period. When a continuous-time signal is multiplied by this impulse train, it generates impulses with amplitudes corresponding to the signal's values at the sampling points.
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Reconstruction of Signal using Interpolation01:10

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Signal processing techniques are essential for accurately converting continuous signals to digital formats and vice versa. When a continuous signal is sampled with a period T, the resulting sampled signal exhibits replicas of the original spectrum in the frequency domain, spaced at intervals equal to the sampling frequency. To handle this sampled signal, a zero-order hold method can be applied, which creates a piecewise constant signal by retaining each sample's value until the next...
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Aliasing01:18

Aliasing

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Accurate signal sampling and reconstruction are crucial in various signal-processing applications. A time-domain signal's spectrum can be revealed using its Fourier transform. When this signal is sampled at a specific frequency, it results in multiple scaled replicas of the original spectrum in the frequency domain. The spacing of these replicas is determined by the sampling frequency.
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Linear Approximation in Frequency Domain01:26

Linear Approximation in Frequency Domain

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Linear systems are characterized by two main properties: superposition and homogeneity. Superposition allows the response to multiple inputs to be the sum of the responses to each individual input. Homogeneity ensures that scaling an input by a scalar results in the response being scaled by the same scalar.
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Zero-bit self-adaptive spectral noise shaping for speech and audio coding.

Niloofar Omidi Piralideh1, Philippe Gournay1, Roch Lefebvre1

  • 1Speech and audio research group (GRPA), Department of Electrical and Computer Engineering, Université de Sherbrooke, Sherbrooke, QC J1K 2R1 Canada.

Journal on Audio, Speech, and Music Processing
|March 30, 2026
PubMed
Summary
This summary is machine-generated.

This study introduces a novel "zero-bit" constrained adaptation method for noise shaping in speech and audio coding. It effectively reduces quantization noise without increasing bitrate or introducing temporal lag.

Keywords:
Coding noiseConstrained adaptationNoise shapingPre- and post-processing filtersSpeech and audio coding

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Area of Science:

  • Signal Processing
  • Audio Engineering
  • Information Theory

Background:

  • Quantization noise in audio coding is often modeled as white noise, causing audible distortions.
  • Noise shaping techniques redistribute noise to less perceptible frequencies, improving perceived audio quality.
  • Existing methods for adaptive noise shaping involve forward or backward coefficient estimation, which can introduce lag or require additional bitrate.

Purpose of the Study:

  • To introduce a novel, bitrate-efficient method for perceptually shaping coding noise in speech and audio codecs.
  • To develop a new approach for estimating adaptive linear filter coefficients for noise shaping.
  • To implement and evaluate a "zero-bit" constrained adaptation technique.

Main Methods:

  • The proposed method utilizes pre- and post-processing adaptive linear filters.
  • Filter coefficients are estimated directly from the current input signals for both pre- and post-processors.
  • A "constrained adaptation" approach is implemented, imposing specific constraints on coefficient calculation.

Main Results:

  • The "zero-bit" constrained adaptation method effectively shapes coding noise.
  • Performance is comparable to traditional forward and backward adaptation techniques.
  • The method introduces no additional bitrate cost or temporal lag.

Conclusions:

  • Constrained adaptation offers an efficient and effective solution for noise shaping in audio coding.
  • This bitrate-free, lag-free approach enhances perceived audio quality.
  • The technique holds promise for improving speech and audio codec performance.