A headset terminal for speech applications includes a headband assembly, an earcup assembly and a power source assembly. Processing circuitry is positioned in at least one of the earcup assembly and the power source assembly and includes speech processing circuitry for recognizing and synthesizing speech. A radio communicates with a central system to process the activity information of the headset terminal user. A rotatable microphone boom assembly includes controls mounted on opposite sides of a rotation axis to maintain a consistent orientation on the boom assembly with respect to the head of a user. The boom assembly snaps together with the earcup assembly to rotate. The headband assembly includes at least one transverse band and a sliding arm coupled to the earcup assembly for dynamically adjusting the position of the earcup assembly. A latch of the power source assembly snaps into position to secure it with the assembly and slides between latched and unlatched positions to secure a battery.

Headset terminal with speech functionality

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

Scalable and embedded codec for speech and audio signals

A method and system use an alternative sensor signal received from a sensor other than an air conduction microphone to estimate a clean speech value. The estimation uses either the alternative sensor signal alone, or in conjunction with the air conduction microphone signal. The clean speech value is estimated without using a model trained from noisy training data collected from an air conduction microphone. Under one embodiment, correction vectors are added to a vector formed from the alternative sensor signal in order to form a filter, which is applied to the air conductive microphone signal to produce the clean speech estimate. In other embodiments, the pitch of a speech signal is determined from the alternative sensor signal and is used to decompose an air conduction microphone signal. The decomposed signal is then used to determine a clean signal estimate.

Method and apparatus for multi-sensory speech enhancement

In a synthesis unit generator, a plurality of synthesis speech segments are generated by synthesizing training speech segments labeled with phonetic contexts and input speech segments while altering the pitch/duration of the input speech segments in accordance with the pitch/duration of the training speech segments. Typical speech segments are selected from the input speech segments on the basis of a distance between the synthesis speech segments and the training speech segments, and are stored in a storage. In addition, a plurality of phonetic context clusters corresponding to the synthesis units are generated on the basis of the distance, and are stored in a storage. A synthesis speech signal is generated by reading out, from the storage, those of the synthesis units, which correspond to the phonetic context clusters including phonetic contexts of input phonemes, and connecting the selected synthesis units in a speech synthesizer.

Speech synthesis method

An inband signaling modem (28) communicates digital data (29) over a voice channel (34) of a wireless telecommunications network (12). An input receives digital data. An encoder (18) converts the digital data (29) into audio tones that synthesize frequency characteristics of human speech (22). The digital data (29) is also encoded to prevent voice encoding circuitry in the telecommunications network (12) from corrupting the synthesized audio tones representing the digital data (29). An output then outputs the synthesized audio tones to a voice channel (34) of a digital wireless telecommunications network (12).

In-band signaling for data communications over digital wireless telecommunications networks

A modular system and method is provided for encoding and decoding of speech signals using voicing probability determination. The continuous input speech is divided into time segments of a predetermined length. For each segment the encoder of the system computes the signal pitch and a parameter which is related to the relative content of voiced and unvoiced portions in the spectrum of the signal, which is expressed as a ratio Pv, defined as a voicing probability. The voiced portion of the signal spectrum, as determined by the parameter Pv, is encoded using a set of harmonically related amplitudes corresponding to the estimated pitch. The unvoiced portion of the signal is processed in a separate processing branch which uses a modified linear predictive coding algorithm. Parameters representing both the voiced and the unvoiced portions of a speech segment are combined in data packets for transmission. In the decoder, speech is synthesized from the transmitted parameters representing voiced and unvoiced portions of the speech in a reverse order. Boundary conditions between voiced and unvoiced segments are established to ensure amplitude and phase continuity for improved output speech quality. Perceptually smooth transition between frames is ensured by using an overlap and add method of synthesis. Also disclosed is the use of the system in the generation of a variety of voice effects.

Speech coding system and method using voicing probability determination

A modular system and method is provided for low bit rate encoding and decoding of speech signals using voicing probability determination. The continuous input speech is divided into time segments of a predetermined length. For each segment the encoder of the system computes a model signal and subtracts the model signal from the original signal in the segment to obtain a residual excitation signal. Using the excitation signal the system computes the signal pitch and a parameter which is related to the relative content of voiced and unvoiced portions in the spectrum of the excitation signal, which is expressed as a ratio Pv, defined as a voicing probability. The voiced and the unvoiced portions of the excitation spectrum, as determined by the parameter Pv, are encoded using one or more parameters related to the energy of the excitation signal in a predetermined set of frequency bands. In the decoder, speech is synthesized from the transmitted parameters representing the model speech, the signal pitch, voicing probability and excitation levels in a reverse order. Boundary conditions between voiced and unvoiced segments are established to ensure amplitude and phase continuity for improved output speech quality. Perceptually smooth transition between frames is ensured by using an overlap and add method of synthesis. LPC interpolation and post-filtering is used to obtain output speech with improved perceptual quality.

Low bit-rate speech coding system and method using voicing probability determination

A voicing probability determination method is provided for estimating a percentage of unvoiced and voiced energy for each harmonic within each of a plurality of bands of a speech signal spectrum. Initially, a synthetic speech spectrum is generated based on the assumption that speech is purely voiced. The original and synthetic speech spectra are then divided into plurality of bands. The synthetic and original speech spectra are compared harmonic by harmonic, and a voicing determination is made based on this comparison. In one embodiment, each harmonic of the original speech spectrum is assigned a voicing decision as either completely voiced or unvoiced by comparing the difference with an adaptive threshold. If the difference for each harmonic is less than the adaptive threshold, the corresponding harmonic is declared as voiced; otherwise the harmonic is declared as unvoiced. The voicing probability for each band is then computed based on the amount of energy in the voiced harmonics in that decision band. Alternatively, the voicing probability for each band is determined based on a signal to noise ratio for each of the bands which is determined based on the collective differences between the original and synthetic speech spectra within the band.

Suat Yeldener

Papers

Speech coding system and method using voicing probability determination

Low bit-rate speech coding system and method using voicing probability determination

Method of determining the voicing probability of speech signals

Verfahren zur feststellung der wahrscheinlichkeit, dass ein sprachsignal stimmhaft ist