Nortel Networks Corporation

Northern Telecom Limited

Paul Mermelstein

Rafi Rabipour

Ahmad Jalali

Vishwa N Gupta

Paul Vincent Coverdale

Matthew Lennig

William Navarro

Brian R Shelton

Vasu Iyengar

Michael G Beyrouti

An audio signal encoding device is provided including an input for receiving a sub-frame of an audio signal to be encoded, an adaptive codebook and a processing unit. The adaptive codebook stores at least one prior knowledge entry which includes a data element representative of characteristics of at least a portion of a previously generated audio signal sub-frame. The processing unit generates a set of parameters allowing for synthesization of the audio signal sub-frame received at the input on the basis of at least the sub-frame of the audio signal received at the input and the data element stored in the adaptive codebook. A corresponding decoding device for synthesizing an audio signal on the basis of a set of parameters is also provided.

Apparatus and method for coding speech signals by making use of an adaptive codebook

An audio signal encoding device is provided comprising an input for receiving a sub-frame of an audio signal, a voiced audio signal synthesis stage, an unvoiced audio signal synthesis stage, and a processing unit. The voiced audio signal synthesis stage is operative for producing a first synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a first set of parameters. The unvoiced audio signal synthesis stage is operative for producing a second synthetic audio signal approximating the sub-frame of an audio signal received at the input on the basis of a second set of parameters. The processing unit is operative for releasing a set of parameters allowing to generate a selected one of the first synthetic audio signal and the second synthetic audio signal.

Apparatus and method for coding speech signals by making use of voice/unvoiced characteristics of the speech signals

The invention relates to a linear prediction audio signal processing apparatus, such as a vocoder, including a nonlinear filter to attenuate the residual signal used to excite a linear prediction synthesis filter. The nonlinear filter is capable of reducing the noise component in the signal while keeping only the periodic component of the speech signal. This feature enhances speech quality. The invention also extends to a novel method for processing a residual signal used to excite a linear prediction synthesis filter in order to attenuate wide band additive noise in the speech signal as constructed by the synthesis filter.

Nonlinear filter for noise suppression in linear prediction speech

In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.

Method and apparatus for improving the voice quality of tandemed vocoders

A synchronous discontinuous transmission medium access control (SDTX-MAC) method and apparatus for more efficiently using existing uplink channels by sharing these uplink channels between multiple terminals engaged in bursty data transmission. This is accomplished by assigning each mobile terminal an individual time slot and by not requiring each mobile terminal to broadcast its identity. This results in a reduction in the number of receivers on each base station and a reduction in the length of the synchronization message.

Medium access control scheme for data transmission on code division

A code-excited linear prediction (CELP) coding method and code divide the residual signal into frequency bands. Codebooks provided for each band decrease in size with increasing band frequency. Reduction in codebook size with increasing frequency together with reduction in sampling rate with decreasing frequency provide reductions in codebook search complexity that allow real time implementation on digital signal processor chips.

Reducing search complexity for code-excited linear prediction (CELP)

A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.

Speech bandwidth extension method and apparatus

In a speech recognizer, for recognizing unknown utterances in isolated-word speech or continuous speech, improved recognition accuracy is obtained by augmenting the usual spectral representation of the unknown utterance with a dynamic component. A corresponding dynamic component is provided in the templates with which the spectral representation of the utterance is compared. In preferred embodiments, the representation is mel-based cepstral and the dynamic components comprise vector differences between pairs of primary cepstra. Preferably the time interval between each pair is about 50 milliseconds. It is also preferable to compute a dynamic perceptual loudness component along with the dynamic parameters.

Speech recognition

A dual tone multi-frequency (DTMF) receiver for use in detecting digitally transmitted signals in the telephone industry. The receiver derives linear predictive coefficients for the digital signals in the data frame. This information is used to compute frequency response magnitudes at the DTMF frequencies. A plurality of magnitude comparisons are then performed to verify the presence of true DTMF signals and concurrently the frequency of these signals is obtained.

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Cote St-Luc, Canada

Paul Mermelstein