Geotagged environmental audio for enhanced speech recognition accuracy

A user terminal (16) includes an RF transceiver (30) for transmitting modulated RF signals to a gateway (14) and for receiving modulated RF signals from the gateway via at least one satellite (112). The satellite may be a GEO or a non-GEO satellite. The user terminal further includes a signal processor (32) coupled to the RF transceiver for demodulating received RF signals to digital signals, an audio unit (90,92,94) for transponding speech signals with a user, and a plurality of encoders/decoders (2A,2B) and/or vocoders (82,84,86) between the signal processor and the audio unit for encoding and decoding speech signals passing to and from the audio unit. A first vocoder can be a variable rate vocoder, a second vocoder can be a fixed rate vocoder, where the fixed rate vocoder operates at an average data rate that differs from, e.g., is less than, an average data rate of the variable rate vocoder. The encoder/decoder or vocoder selection is made in the gateway based on one or more criterion such as a presence of an RF signal impairment or RF interference, a predicted and/or actual communications system, a user-selected voice quality, and/or a signal quality received from the user terminal. In one embodiment the vocoder selection command is generated in the terrestrial gateway in response to a request from the user terminal, and in this case the criterion is based on a quality of a signal received from the terrestrial gateway at the user terminal. The selection can be made on the fly on a frame or multi-frame basis.

Multiple vocoder mobile satellite telephone system

An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Encoding device and decoding device

The paper introduces a modification of the commonly used postfilter that improves performance when acoustic background noise is present. The modification consists of replacing the nonadaptive postfilter parameters that govern the degree of spectral emphasis (commonly denoted as /spl gamma//sub 1/ and /spl gamma//sub 2/) with parameters that adapt to the noise statistics. We describe an effective mapping from the noise statistics to the emphasis parameters and provide a low complexity noise estimation algorithm that is sufficient for this application. The resulting noise-adaptive postfilter successfully attenuates the background noise and naturally converges to the conventional postfilter at high SNR conditions. Thus, the speech enhancement problem is solved with minimal modification of legacy codecs, since the existing structure of the speech codec is used. Test results indicate that the presented algorithm significantly outperforms the standard postfilter with non-adaptive parameters.

Noise-dependent postfiltering

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Method for encoding and decoding a speech signal including background noise

Adjusting the shape of a spectrum of a speech signal includes the steps of using a first filter with pole-zero transfer function A(z)/B(z) for subjecting a speech signal to a spectrum envelop emphasis and a second filter cascade-connected with the first filter, for compensating for a spectral tilt due to the first filter, independently deriving two filter coefficients used in the second filter for compensating for the spectral tilt from the pole-zero transfer function, and compensating for the spectral tilt corresponding to the pole-zero transfer function according to the derived filter coefficients.

Method and apparatus for adjusting a spectrum shape of a speech signal

An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal

Audio signal processing method and apparatus for the same

A noise estimation unit estimates a noise signal in an input signal. A section decision unit distinguishes a target signal section from a noise signal section in the input signal. A noise suppression unit suppresses the noise signal based on a first suppression coefficient from the input signal. A noise excess suppression unit suppresses the noise signal based on a second suppression coefficient from the input signal. The second suppression coefficient is larger than the first suppression coefficient. A switching unit switches between an output signal from the noise suppression unit and an output signal from the noise excess suppression unit based on a decision result of the section decision unit.

Noise suppression apparatus and method

According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.

Tadashi Amada

Papers

Method for encoding and decoding a speech signal including background noise

Method and apparatus for adjusting a spectrum shape of a speech signal

Audio signal processing method and apparatus for the same

Noise suppression apparatus and method

Pickup signal processing apparatus, method, and program product