Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.

Audio watermarking for copyright protection of digital audio data

Abstract: A novel audio watermarking algorithm is presented as a means of protecting unauthorised copying of digital audio data. The proposed watermarking scheme includes a psychoacoustic model to achieve perceptual transparency after carrying out an embedding and whitening procedure before correlation in detection to extract the copyright information without requiring access to the original signal. It is shown that the proposed scheme is robust and introduces no audible distortion after watermark insertion.

The JSRU channel vocoder

Abstract: During the period from 1956 to 1966 the UK Government's Joint Speech Research Unit was conducting research into channel vocoders, culminating in a laboratory-built design suitable for evaluation by potential users over digital transmission networks at 2400 bit/s. The success of the basic vocoder design was such that it has since been engineered in various forms for widespread operational use, using different technologies as they have evolved. In view of the JSRU vocoder's continued competitiveness with other narrow-band speech coding techniques, such as linear predictive coding, this paper has been written to give a full specification of all the essential features of the vocoder configuration, with comments on the reasons for the design decisions and reference to supporting research where appropriate. The two most important factors contributing to this vocoder's successful performance are the use of narrow-band single-resonant circuits for the synthesis filters and the use of differential coding between channels in the digitisation process.

Audio encoder for encoding an audio signal having an impulse-like portion and stationary portion, encoding methods, decoder, decoding method; and encoded audio signal

Abstract: An audio encoder for encoding an audio signal includes an impulse extractor (10) for extracting an impulse-like portion from the audio signal. This impulse-like portion is encoded and forwarded to an output interface (22). Furthermore, the audio encoder includes a signal encoder (16) which encodes a residual signal derived from the original audio signal so that the impulse-like portion is reduced or eliminated in the residual audio signal. The output interface (22) forwards both, the encoded signals, i.e., the encoded impulse signal (12) and the encoded residual signal (20) for transmission or storage. On the decoder-side, both signal portions are separately decoded and then combined to obtain a decoded audio signal.

Apparatus and methods for proportional audio compression and frequency shifting

Abstract: Apparatus and methods for audio compression and frequency shifting retain the spectral shape of an audio input signal while compressing and shifting its frequency. The fast Fourier transform of the input signal is generated, to allow processing in the frequency domain. The input audio signal is divided into small time segments, and each is subjected to frequency analysis. Frequency processing includes compression and optional frequency shifting. The inverse fast Fourier transform function is performed on the compressed and frequency shifted spectrum, to compose an output audio signal, equal in duration to the original signal. The output signal is then provided to the listener with appropriate amplification to insure audible speech across the usable frequency range.

A wideband speech and audio codec at 16/24/32 kbit/s using hybrid ACELP/TCX techniques

Abstract: A hybrid ACELP/TCX algorithm for coding speech and music signals at 16, 24, and 32 kbit/s is presented. The algorithm switches between algebraic code excited linear prediction (ACELP) and transform coded excitation (TCX) modes on a 20-ms frame basis. Applying TCX on 20 ms frames improved the quality for music signals. Special care was taken to alleviate the switching artifacts between the two modes resulting in a transparent switching process. Subjective test results showed that for speech signals, the performance at 16, 24, and 32 kbit/s, is equivalent to G.722 at 48, 56, and 64 kbit/s, respectively. For music signals, the quality at 24 kbit/s was found equivalent to G.722 at 56 kbit/s. However, at 16 kbit/s, the quality for music was slightly lower than G.722 at 48 kbit/s.