Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.
Common applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques.
The techniques employed in speech coding are similar to those used in audio data compression and audio coding where appreciation of psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but the reconstructed signal retains adequate intelligibility.
Speech coding differs from other forms of audio coding in that speech is a simpler signal than other audio signals, and statistical information is available about the properties of speech. As a result, some auditory information that is relevant in general audio coding can be unnecessary in the speech coding context. Speech coding stresses the preservation of intelligibility and pleasantness of speech while using a constrained amount of transmitted data. In addition, most speech applications require low coding delay, as latency interferes with speech interaction.
Speech coders are of two classes:
Waveform coders
Time-domain: PCM, ADPCM
Frequency-domain: sub-band coding, ATRAC
Vocoders
Linear predictive coding (LPC)
Formant coding
Machine learning, i.e. neural vocoder
The A-law and μ-law algorithms used in G.711 PCM digital telephony can be seen as an earlier precursor of speech encoding, requiring only 8 bits per sample but giving effectively 12 bits of resolution.
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Explore les systèmes LID réalisables, les fonctions de transfert rationnelles, la stabilité, la causalité et les paramètres de mise en œuvre dans le calcul.
The goal of this course is to provide the students with the main formalisms, models and algorithms required for the implementation of advanced speech processing applications (involving, among others,
The goal of this course is to introduce the engineering students state-of-the-art speech and audio coding techniques with an emphasis on the integration of knowledge about sound production and auditor
Study of the essential components and implementation technologies of digital signal processing and communication systems from the theoretical, algorithmic and system implementation point of view.
The course provides a comprehensive overview of digital signal processing theory, covering discrete time, Fourier analysis, filter design, sampling, interpolation and quantization; it also includes a
Adaptive signal processing, A/D and D/A. This module provides the basic
tools for adaptive filtering and a solid mathematical framework for sampling and
quantization
La voix sur IP, ou « VoIP » pour « Voice over IP », est une technologie informatique qui permet de transmettre la voix sur des réseaux compatibles IP, via Internet ou des réseaux privés (intranets) ou publics, qu'ils soient filaires (câble/ADSL/fibre optique) ou non (satellite, Wi-Fi et réseaux mobiles). Des logiciels de VoIP tels que Skype, Signal, Discord, WhatsApp gèrent aujourd'hui tous les flux multimédia (téléphonie, appels vidéo, messagerie instantanée et transferts de fichiers).
An audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
La transformée en cosinus discrète ou TCD (de l'anglais : DCT ou Discrete Cosine Transform) est une transformation proche de la transformée de Fourier discrète (DFT). Le noyau de projection est un cosinus et crée donc des coefficients réels, contrairement à la DFT, dont le noyau est une exponentielle complexe et qui crée donc des coefficients complexes. On peut cependant exprimer la DCT en fonction de la DFT, qui est alors appliquée sur le signal symétrisé.
Atypical aspects in speech concern speech that deviates from what is commonly considered normal or healthy. In this thesis, we propose novel methods for detection and analysis of these aspects, e.g. to monitor the temporary state of a speaker, diseases tha ...
Speech recognition-based applications upon the advancements in artificial intelligence play an essential role to transform most aspects of modern life. However, speech recognition in real-life conditions (e.g., in the presence of overlapping speech, varyin ...
EPFL2023
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Self-supervised learning (SSL) models use only the intrinsic structure of a given signal, independent of its acoustic domain, to extract essential information from the input to an embedding space. This implies that the utility of such representations is no ...