International Business Machines Corporation

Jean E Menez

Claude R Galand

Michele Rosso

Gerald Lebizay

Gerard Richter

Daniel J Esteban

Emmanuel Lancon

Paolo Scotton

Bernard Y Pucci

Daniel Mauduit

Gilbert Furlan

Francoise Bottau

Catherine Raimondo

Eric Goubault

Process for transmitting compressed voice circuits over a packet switching network of the type comprising a plurality of switching nodes (SW-1 to SW-7) interconnected by connection lines and including at least an end switching node (SW-1) connected to a source exchange telephone device (PABX A) and another end switching node (SW-3) connected to a destination exchange telephone device (CX), and comprising the steps of receiving from the source exchange telephone device a sequence of uncompressed frames wherein each frame contains n slots containing each J bytes respectively associated to J voice circuits, compressing the data bits of each voice circuit in order to build a packet containing J compressed voice circuits, and providing this packet to the end switching node for transmission to the destination exchange telephone device.

Transmission system of telephony circuits over a packet switching network

A system for coding voice signal to optimize bandwidth occupation in a High Speed Packet Switching network while ensuring best voice transmission quality. The voice signal is first encoded using a conventional GSM like RPE/LTP coder providing first sub-frames of coded signal and tagging these first sub-frames as being non-discardable. In addition, a convenient difference between an RPE/LTP provided signal and a corresponding synthesized image is performed (see 36) and is also block encoded into second sub-frames which second sub-frames are tagged as being discardable sub-frames. Said second sub-frames when concatenated to corresponding first sub-frames provide so-called multirate frames. Then, when transmitting said multirate frames over the High Speed packet switching network, dropping discardable tagged data enables solution network congestion situations in any network node and at random with no significant disturbing effect over the voice communication operation.

System for coding voice signals to optimize bandwidth occupation in high

This technique for coding images involves filtering the original input image into subband images using successive high-pass and low-pass filterings over a tree-shaped filter with each tree filtering stage involving re-scaling operations adjusted to keep the subband image signal dynamics substantially stable throughout the filtering tree.

Method and apparatus for subband coding images

Technique for coding video images by filtering the original input image into subband images using successive high-pass and low-pass filterings over a tree-shaped filter bank. Each subband image is then submitted to activity detection and coded with a different number of levels depending on the activity detected and on the subband involved.

Method and apparatus for subband coding video signals

A vector quantization encoder comprises a codebook built with the so-called Linde-Buzo-Gray (LBG) algorithm, an intermediary codebook built from the initial codebook (LBG) and a Hash-table containing subsets of the initial codebook (LBG). Vector quantization is then performed in two steps. First, a multistep prequantization of an input vector gives an index in the Hash-table which points to a subset of vectors of the initial codebook (LBG). Then a full search is performed within the pointed subset of vectors of the initial codebook (LBG).

Method and apparatus for coding digital data using vector quantizing

The echo cancellation method includes starting with an arbitrary set of echo filter coefficients. The coefficients are first sequentially re-estimated using a fast converging method. These operations are repeated a predefined number of times, and then the coefficients estimation is switched to a low converging but more precise method. In addition, an adaptive threshold is being defined and readjusted and all filter coefficients lower than said threshold are being discarded. Finally, sampling time boundaries are adaptively defined based on non-discarded filter coefficients with reference to said threshold, and in such a way the final echo cancellation algorithm is adaptive to echo variations (backtracking algorithm).

Adaptive echo cancellation method and device for implementing said method

A vector quantizing speech coder includes a Short-Term-Predictive filter which receives originally sampled speech signal s(n) and decorrelates it into a residual signal r(n). A device is provided to quantize the residual signal r(n) at a low bit rate. The device also generates a reconstructed signal r(n) from which coefficients for adjusting the Short-Term-Predictive filter are dynamically derived.

Low-delay low-bit-rate speech coder

A signal vector quantizing coder (CELP) is provided with an adaptive codebook originally loaded with preselected codewords. The codebook is split into a fixed contents portion and a fixed length adaptive contents portion. During coding operations, the codewords dynamically selected for coding the coder input signal are shifted into the fixed length adaptive codebook section for codebook contents updating purposes.

Method and apparatus for dynamically adapting a vector-quantizing coder

The voice signal is analyzed to derive therefrom a low frequency base band signal, linear prediction coefficients and high frequency (HF) descriptors. Said HF descriptors include HF energy indications as well as indications relative to the phase shift between the low frequency and the high frequency band. Said HF descriptors are used during the voice synthesis operation to provide an inphase HF bandwidth component to be added to the base band prior to be used for driving a linear prediction synthesis filter tuned using said linear prediction parameters.

Voice coding process and device for implementing said process

This low bit rate voice encoding involves short-term predictive filtering the voice signal s(n) using partial correlation related coefficients derived from pre-emphasized s(n), and deriving a short term signal r(n); then deriving a long-term residual signal e(n) by subtracting a delayed synthesized short term b.r'(n-31 M) from said r(n); and code excited encoding e(n) into codeword references k's and associated gains G's.

Low bit rate voice coding method and system

A voice signal is transmitted digitally at reduced bit rate by use of data compression. The original frequency components of a telephone band width of the voice signal sampled at 8 KHz and quantized with 12 bits are transformed into three parameters: SIGNAL data representing adaptive quantization for lower frequency (300-800 Hz) sub-bands; COEF data representing pre-emphasized parcor type coefficients for the higher frequency (800-3000 Hz) band; and, ENERGY data representing higher frequency short term energy level. The three parameters are multiplexed for transmission in binary-code form, thereby representing a recoding of the original binary-coded voice signals.

Process for compressing data relative to voice signals and device

This is a block quantizer which converts a given block of samples into an optimum sequence of numbers describing the sampled signal with minimum distortion due to the quantizing process. The quantizer self-adjusts the basic parameters, i.e., quantizing step Q, d.c. level C and the sequence of numbers fn, describing the signal for each block of samples for a minimal Mean Squared Error.

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Nice, France

Jean E Menez