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 present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Audio encoder and decoder

An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits.

Quality improvement techniques in an audio encoder

An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Multi-channel audio encoding and decoding

Efficient and Scalable Parametric Stereo Coding for Low Bitrate Applications

The linear predictive analysis method is used in order to determine the spectral parameters representing the spectral envelope of the audiofrequency signal. This method comprises q successive prediction stages, q being an integer greater than 1. At each prediction stage p(1≦p≦q), parameters are determined representing a predefined number Mp of linear prediction coefficients a 1 p , . . . , a Mp p of an input signal of the said stage. The audiofrequency signal to be analysed constitutes the input signal of the first stage. The input signal of a stage p+1 consists of the input signal of the stage p filtered with a filter with transfer function ##EQU1##

Method for linear predictive analysis of an audiofrequency signal, and method for coding and decoding an audiofrequency signal including application thereof

A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Coding with noise shaping in a hierarchical coder

A Scalable Three Bit Rate (8, 14.2, and 24 kbit/s) Audio Coder

This paper presents a scalable three bit- rates (8, 14.1 and 24 kbit/s) coder. For the two embedded lowest ones, operating in the telephone bandwidth, celp coding techniques are used. For the highest rate, that both improves narrowband quality and extends the band to [50-7000Hz], transform coding techniques are used. The main applications deal with transmission over network with no guaranteed QoS.

A Scalable Three Bit-rates 8–14.1-24 kbit/s Audio Coder

The invention relates to a method for generating mutually orthogonal signals having a controlled spectrum, comprising the generation of a plurality of mututally orthogonal, controlled-power discrete spectra s(i) having dimension Q, i designating spectrum number. The aforementioned spectra represent time signals in the spectral range and have a modulus μ that is constant in a spectral line designation set G and zero everywhere else. The inventive method comprises the following steps: determining (20) at least part of a complex Hadamard matrix H of order dR = w in the case of spectra of real signals and dR = 2.w in the case of spectra of complex signals; determining (22) the extension P of matrix H from G and dimension Q; and obtaining (22) controlled-power spectra s(i) = μ.P(H[.][i]), wherein H[.][i] designates the i-th column of matrix H. In addition, a plurality of mutually orthogonal time signals s(i) is also generated from said discrete complex signals.

Alain Le Guyader

Papers

Method for linear predictive analysis of an audiofrequency signal, and method for coding and decoding an audiofrequency signal including application thereof

Coding with noise shaping in a hierarchical coder

A Scalable Three Bit Rate (8, 14.2, and 24 kbit/s) Audio Coder

A Scalable Three Bit-rates 8–14.1-24 kbit/s Audio Coder

Method for generating mutually orthogonal signals having a controlled spectrum