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

The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal

In one disclosed aspect, dynamic gain modification s are applied to an audio signal at least partly in response to auditory events and/or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Audio gain control using specific-loudness-based auditory event detection

One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

Method, apparatus and computer program for calculating and adjusting the perceived loudness of an audio signal

Methods and apparatus for embedding a watermark are disclosed. An example method disclosed herein to embed a watermark in a compressed data stream comprises obtaining a set of transform coefficients included in the compressed data stream, the set of transform coefficients having a respective first set of mantissa codes and a respective set of exponents, the first set of mantissa codes associated with a respective set of mantissa step sizes, identifying a first transform coefficient from the set of transform coefficients having a smallest magnitude among the set of transform coefficients, determining a second set of mantissa codes based on the first transform coefficient and the set of step sizes, and replacing the first set of mantissa codes included in the compressed data stream with the second set of mantissa codes to embed the watermark without uncompressing the compressed data stream.

Methods and apparatus for embedding watermarks

A method of concealing transmission error in a digital audio signal in which a signal that has been decoded after transmission is received, the samples decoded while the transmitted data is valid are stored, at least one short-term prediction operator and one long-term prediction operator are estimated as a function of stored valid samples, and any missing or erroneous samples in the decoder signal are generated using the operators estimated in this way, the method being characterized in that the energy of the synthesized signal as generated in this way is controlled by means of a gain that is computed and adapted sample by sample.

Transmission error concealment in an audio signal

Dispositif de traitement de signal audio aopte a recevoir une pluralite de flux audio d'entree et a fournir au moins un flux audio de sortie a partir desdits flux audio d'entree, l'un au moins des flux audio d'entree, dit premier flux, pouvant presenter des donnees manquantes, le dispositif comportant des moyens pour decider ou non de la regeneration des donnees manquantes dudit premier flux a partir d'au moins une caracteristique d'au moins un second flux audio d'entree.

Methode et dispositif de traitement d'une pluralite de flux binaires audio

The digital data packets received contain data describing successive time segments of a signal stream to be restored, for example audio. Elements obtained from the packets received are recorded in a buffer memory organized in first-in first-out mode. A burst of packets fulfilling a certain burst size condition is detected a priori, on the basis of a reception delay evaluated in response to the reception of a first packet of the burst. The content of the buffer memory is controlled by accounting for the detection of the burst fulfilling the size condition.

Method of processing data packets received on asynchronous networks, and device for implementing the method

The invention relates to processing signal for improving a sound pick-up in telephone and/ or videophone units. The inventive method consists in comparing (13) a signal transmitted by a unit microphone (Mic) with a threshold low value (THR_LOW) and in amplifying (14,15) said microphone-transmitted signal when it is less than the threshold low value. However, said steps depend on the test (10) carried out on a signal used for feeding the loud speaker (HP), in particular consisting in controlling said last signal (crete_HP) of a background noise order (BDFd) in such a way that the background noise is not amplified by an acoustical coupling between the microphone (Mic) and the loud speaker (HP).

Sound pick-up method and device for, in particular in handsfree telephone terminals

The invention relates to a real time processing and management method for canceling out the echo between a loudspeaker (HP) and a microphone (M) of a computer terminal. Said method consists of establishing the loudspeaker signal or the microphone signal as a reference signal and synchronizing (A) the other signal with said reference signal, and (B) synchronizing the acquisition and audio reestablishment tasks. The delay that exists (rej) between the other signal and the reference signal is measured (C) and the existing delay value (rej) is validated (D) in order to control the delay applied to the reference signal starting from the current delay (rej). The invention can be used for computer terminals which are equipped with multi-tasks operating systems.

Real time processing and management method for canceling out the echo between a loudspeaker and a microphone of a computer terminal

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