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 system and method for indexing an audio stream for subsequent information retrieval and for skimming, gisting, and summarizing the audio stream includes using special audio prefiltering such that only relevant speech segments that are generated by a speech recognition engine are indexed. Specific indexing features are disclosed that improve the precision and recall of an information retrieval system used after indexing for word spotting. The invention includes rendering the audio stream into intervals, with each interval including one or more segments. For each segment of an interval it is determined whether the segment exhibits one or more predetermined audio features such as a particular range of zero crossing rates, a particular range of energy, and a particular range of spectral energy concentration. The audio features are heuristically determined to represent respective audio events including silence, music, speech, and speech on music. Also, it is determined whether a group of intervals matches a heuristically predefined meta pattern such as continuous uninterrupted speech, concluding ideas, hesitations and emphasis in speech, and so on, and the audio stream is then indexed based on the interval classification and meta pattern matching, with only relevant features being indexed to improve subsequent precision of information retrieval. Also, alternatives for longer terms generated by the speech recognition engine are indexed along with respective weights, to improve subsequent recall.

System and method for automatic audio content analysis for word spotting, indexing, classification and retrieval

The method and preprocessor enhances the intelligibility of narrowband speech without essentially lengthening the overall time duration of the signal. Both spectral enhancements and variable-rate time-scaling procedures are implemented to improve the salience of initial consonants, particularly the perceptually important formant transitions. Emphasis is transferred from the dominating vowel to the preceding consonant through adaptation of the phoneme timing structure. In a further embodiment, the technique is applied as a preprocessor to a speech coder.

Enhancing speech intelligibility using variable-rate time-scale modification

A subscriber or other user is provided with access to a subscription service of a signal distribution system at a roaming location remote from a home location of the user. The subscription service comprises a plurality of selectable channels available to the user at the home location. In one aspect of the invention, a request for initiation of a roaming session is received, and responsive to initiation of the roaming session, at least one user interface is generated for presentation on a device at the remote location. The generated user interface is configured to provide access at the remote location to one or more of the channels available to the user at the home location, and represents an at least partial alteration of a normal user interface that would otherwise be presented on the device at the remote location in the absence of the roaming session.

User interface methods and apparatus for roaming access to subscription services

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 method and system is provided for encoding and decoding of speech signals at a low bit rate. The continuous input speech is divided into voiced and unvoiced time segments of a predetermined length. The encoder of the system uses a linear predictive coding model for the unvoiced speech segments and harmonic frequencies decomposition for the voiced speech segments. Only the magnitudes of the harmonic frequencies are determined using the discrete Fourier transform of the voiced speech segments. The decoder synthesizes voiced speech segments using the magnitudes of the transmitted harmonics and estimates the phase of each harmonic from the signal in the preceding speech segments. Unvoiced speech segments are synthesized using linear prediction coding (LPC) coefficients obtained from codebook entries for the poles of the LPC coefficient polynomial. Boundary conditions between voiced and unvoiced segments are established to insure amplitude and phase continuity for improved output speech quality.

Harmonic adaptive speech coding method and system

A modular system and method is for displaying and controlling a host computer application interface using a real-time video and control stream to a remote digital media receiver client is disclosed. The continuous graphical display pixels and audio of a host computer application interface window are divided into time segments of a predetermined length. These segments are then subsequently coded as video and audio streams. These video and audio streams are then sent to a remote digital media receiver client where they are decoded for display on a television device or played back on an audio system. The host computer application interface is controlled by user commands that originate at the remote digital media receiver client. The commands are sent over a control channel that is established between the host computer and the digital media receiver.

Joseph Gerard Aguilar

Papers

Speech coding system and method using voicing probability determination

Harmonic adaptive speech coding method and system

Systems and methods for remotely controlling computer applications