Digital coding of waveforms: Principles and applications to speech and video

An aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy, and for each block a gain is determined from the factor and is applied to the transform coefficients of the block.

Methods and devices for low-frequency emphasis during audio compression based on acelp/tcx

An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter Additionally, a signal analyzer for analyzing the audio signal is provided The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter

Multi-resolution switched audio encoding/decoding scheme

The present invention provides a transform coding method efficient for music signals that is suitable for use in a hybrid codec, whereby a common Linear Predictive (LP) synthesis filter is employed for both speech and music signals. The LP synthesis filter switches between a speech excitation generator and a transform excitation generator, in accordance with the coding of a speech or music signal, respectively. For coding speech signals, the conventional CELP technique may be used, while a novel asymmetrical overlap-add transform technique is applied for coding music signals. In performing the common LP synthesis filtering, interpolation of the LP coefficients is conducted for signals in overlap-add operation regions. The invention enables smooth transitions when the decoder switches between speech and music decoding modes.

Method for coding speech and music signals

An efficient and accurate classification method for classifying speech and music signals, or other diverse signal types, is provided. The method and system are especially, although not exclusively, suited for use in real-time applications. Long-term and short-term features are extracted relative to each frame, whereby short-term features are used to detect a potential switching point at which to switch a coder operating mode, and long-term features are used to classify each frame and validate the potential switch at the potential switch point according to the classification and a predefined criterion.

Real-time speech and music classifier

This paper describes a combined adaptive transform codec (ATC) and code-excited linear prediction (CELP) algorithm, called ATCELP, for the compression of wideband (7 kHz) signals. The CELP algorithm applies mainly to speech, whereas the ATC mode is selected for music and noise signals. We propose a switching scheme between CELP and ATC mode and describe a frame erasure concealment technique. Subjective listening tests have shown that the ATCELP codec at bit rates of 16, 24 and 32 kbit/s achieved performances close to those of the CCITT G.722 at 48, 56 and 64 kbit/s, respectively, at most operating conditions.

A 16, 24, 32 kbit/s wideband speech codec based on ATCELP

Embedded algebraic CELP/VSELP coders for wideband speech coding

Digital techniques are more and more often used in telecommunications networks. How would these techniques affect the quality of transmitted speech? The use of subjective testing is generally necessary to answer this question. In this paper a degradation category rating test, which is a speech-adapted procedure of the one used for image testing, is proposed to distinguish among good telephone quality codecs. A comparative study of the sensitivity of the method and that of a classic opinion rating test is reported. It shows that the resolution of the degradation category rating test is twice that of a classic opinion rating test within good telephone quality domain. Subjective performances of various 32 kbit/s A.D.P.C.M codecs, including the effects of transmission errors and tandem connections, using degradation category rating tests are also presented.

Quality evaluation of 32 kbit/s coded speech by means of degradation category ratings

The need of variable bit rate actually exists for an increasing number of applications such as videoconferencing, audioconferencing and packet circuit multiplication. In this contribution we extend the concept of variable bit rate to multi-stage VSELP coding. First we begin with an unified approach for multi stage CELP and VSELP coding and we show that codebooks orthogonalization leads to optimum embedded coders for variable bit rates. Then we derive a new algorithm for the off-line optimization of the basis vectors in the multi-stage case. Finally we give some subjective test results obtained with 24 and 32 kbitts embedded wideband speech coders.

http://ieeexplore.ieee.org/iel5/6181/16503/00762316.pdf

Embedded Wideband Vselp Speech Coding With Optimized Codebooks

This article presents a bitrate- and bandwidth-scalable coder submitted as a candidate to the qualification phase of ITU-T embedded G.711 wideband (G.711-WB) standardization. The encoder operates on 5 ms frames and bitrate can be set on a frame basis at either 64, 80 or 96 kbit/s. The input signal, which is by default sampled at 16 kHz, is split in two bands. The low band (0-4000 Hz) is coded by a low-complexity embedded pulse code modulation (PCM) coder at 8-10 bits/sample, including noise feedback at the encoder and PCM-specific post-filtering at the decoder. The high-band (4000- 8000 Hz) is coded by a time-domain aliasing cancellation (TDAC) coder derived from ITU-T G.729.1. Frame erasures are concealed using a low-complexity split-band algorithm derived from ITU-T G.722 Appendix IV. The proposed coder passed all requirements of the G.711WB qualification phase. We summarize and discuss formal ITU-T test results.

A. Le Guyader

Papers

A 16, 24, 32 kbit/s wideband speech codec based on ATCELP

Embedded algebraic CELP/VSELP coders for wideband speech coding

Quality evaluation of 32 kbit/s coded speech by means of degradation category ratings

Embedded Wideband Vselp Speech Coding With Optimized Codebooks

An 64-80-96 kbit/s scalable wideband speech coding candidate for ITU-T G.711-WB standardization