Digital Voice Systems, Inc.

John C Hardwick

Jae S Lim

Daniel W Griffin

Thomas Clark

William S McKinney

Brian D Clooney

Timothy E Kalvaitis

This disclosure provides a method of decoding a digital speech signal, a speech decoder, a handset or mobile radio, and a base station or console. The method includes receiving a voice bit stream including at least one frame of bits that includes block codes, determining least confident bits in a first block code, generating candidates for the first block code based on the least confident bits, determining a first distance between each candidate and the first block code, and demodulating at least one other block code to obtain at least one demodulated vector. For each demodulated vector, a second distance between the demodulated vector and possible transmitted vectors is determined, and from the possible transmitted vectors, a vector corresponding to a minimum second distance is selected as a corrected demodulated vector. A minimum total distance is determined, and a candidate is selected as a corrected first block code.

Bit error correction in digital speech

Tone data embedded in a voice bit stream that includes frames of non-tone bits and frames of tone bits is detected and extracted by selecting a frame of bits, analyzing the selected frame of bits to determine whether it is a frame of tone bits, and, when it is a frame of tone bits, extracting tone data from it. Analyzing the selected frame includes comparing bits of the selected frame to sets of tone data to produce error criteria representative of differences between the selected frame and each of multiple sets of tone data. Based on the error criteria, a set of tone data that most closely corresponds to the bits of the selected frame is selected. When the error criteria corresponding to the selected set of tone data satisfies a set of thresholds, the selected frame is designated as a frame of tone bits.

Tone frame detector for digital speech

Compensating a speech signal for the presence of a speaker mask includes receiving a speech signal, dividing the speech signal into subframes, generating speech parameters for a subframe, and determining whether the subframe is suitable for use in detecting a mask. If the subframe is suitable for use in detecting a mask, the speech parameters for the subframe are used in determining whether a mask is present. If a mask is present, the speech parameters for the subframe are modified to produce modified speech parameters that compensate for the presence of the mask.

Detecting and compensating for the presence of a speaker mask in a speech signal

Non-voice data is embedded in a voice bit stream that includes frames of voice bits by selecting a frame of voice bits to carry the non-voice data, placing non-voice identifier bits in a first portion of the voice bits in the selected frame, and placing the non-voice data in a second portion of the voice bits in the selected frame. The non-voice identifier bits are employed to reduce a perceived effect of the non-voice data on audible speech produced from the voice bit stream.

Reducing perceived effects of non-voice data in digital speech

Quantizing speech model parameters includes, for each of multiple vectors of quantized excitation strength parameters, determining first and second errors between first and second elements of a vector of excitation strength parameters and, respectively, first and second elements of the vector of quantized excitation strength parameters, and determining a first energy and a second energy associated with, respectively, the first and second errors. First and second weights for, respectively, the first error and the second error, are determined and are used to produce first and second weighted errors, which are combined to produce a total error. The total errors of each of the multiple vectors of quantized excitation strength parameters are compared and the vector of quantized excitation strength parameters that produces the smallest total error is selected to represent the vector of excitation strength parameters.

Speech model parameter estimation and quantization

An audio watermarking system conveys information using an audio channel by modulating an audio signal to produce a modulated signal by embedding additional information into the audio signal. Modulating the audio signal includes segmenting the audio signal into overlapping time segments using a non-rectangular analysis window function produce a windowed audio signal, processing the windowed audio signal for a time segment to produce frequency coefficients representing the windowed time segment and having phase values and magnitude values, selecting one or more of the frequency coefficients, modifying phase values of the selected frequency coefficients using the additional information to map the phase values onto a known phase constellation, and processing the frequency coefficients including the modified phase values to produce the modulated signal.

Audio watermarking via phase modification

Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames, computing model parameters for a frame, and quantizing the model parameters to produce pitch bits conveying pitch information, voicing bits conveying voicing information, and gain bits conveying signal level information. One or more of the pitch bits are combined with one or more of the voicing bits and one or more of the gain bits to create a first parameter codeword that is encoded with an error control code to produce a first FEC codeword that is included in a bit stream for the frame. The process may be reversed to decode the bit stream.

Half-rate vocoder

Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Interoperable vocoder

Speech enhancement in a breathing apparatus is provided using a primary sensor mounted near a breathing mask user's mouth, at least one reference sensor mounted near a noise source, and a processor that combines the signals from these sensors to produce an output signal with an enhanced speech component. The reference sensor signal may be filtered and the result may be subtracted from the primary sensor signal to produce the output signal with an enhanced speech component. A method for detecting the exclusive presence of a low air alarm noise may be used to determine when to update the filter. A triple filter adaptive noise cancellation method may provide improved performance through reduction of filter maladaptation. The speech enhancement techniques may be employed as part of a communication system or a speech recognition system.

Breathing apparatus speech enhancement using reference sensor

Synthesizing a set of digital speech samples corresponding to a selected voicing state includes dividing speech model parameters into frames, with a frame of speech model parameters including pitch information, voicing information determining the voicing state in one or more frequency regions, and spectral information. First and second digital filters are computed using, respectively, first and second frames of speech model parameters, with the frequency responses of the digital filters corresponding to the spectral information in frequency regions for which the voicing state equals the selected voicing state. A set of pulse locations are determined, and sets of first and second signal samples are produced using the pulse locations and, respectively, the first and second digital filters. Finally, the sets of first and second signal samples are combined to produce a set of digital speech samples corresponding to the selected voicing state.

Synthesizing/decoding speech samples corresponding to a voicing state

A digital audio server may be used to automatically download music from a collection of audio media, such as CDs or DVDs. The server also may automatically identify the media using track offset information.

Digital audio server

First encoded voice bits are transcoded into second encoded voice bits by dividing the first encoded voice bits into one or more received frames, with each received frame containing multiple ones of the first encoded voice bits. First parameter bits for at least one of the received frames are generated by applying error control decoding to one or more of the encoded voice bits contained in the received frame, speech parameters are computed from the first parameter bits, and the speech parameters are quantized to produce second parameter bits. Finally, a transmission frame is formed by applying error control encoding to one or more of the second parameter bits, and the transmission frame is included in the second encoded voice bits.

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Sudbury, MA, United States of America

John C Hardwick