Abstract: 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.

Abstract: 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.

Abstract: An improved speech model and methods for estimating the model parameters, synthesizing speech from the parameters, and quantizing the parameters are disclosed. The improved speech model allows a time and frequency dependent mixture of quasi-periodic, noise-like, and pulse-like signals. For pulsed parameter estimation, an error criterion with reduced sensitivity to time shifts is used to reduce computation and improve performance. Pulsed parameter estimation performance is further improved using the estimated voiced strength parameter to reduce the weighting of frequency bands which are strongly voiced when estimating the pulsed parameters. The voiced, unvoiced, and pulsed strength parameters are quantized using a weighted vector quantization method using a novel error criterion for obtaining high quality quantization. The fundamental frequency and pulse position parameters are efficiently quantized based on the quantized strength parameters.

Abstract: 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.

Abstract: 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.

Abstract: A lossless coding method may be used to compress information, such as audio data, without introducing any artifacts. This lossless coding method may be used to compress audio signals for use in storage and/or transmission of audio data. The audio data may be compressed by first dividing digital samples taken from the audio data into frames. A predictor is then used on the frames to generate prediction coefficients that can then be quantized to form predictor bits. The frames may then be subdivided into subsets. Another predictor can be used on the subsets to produce error samples that can be entropy coded into codeword bits. The predictor bits and codeword bits can be included in the compressed audio output for use in decoding.

Abstract: 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.

Abstract: An improved speech model and methods for estimating the model parameters, synthesizing speech from the parameters, and quantizing the parameters are disclosed. The improved speech model allows a time and frequency dependent mixture of quasi-periodic, noise-like, and pulse-like signals. For pulsed parameter estimation, an error criterion with reduced sensitivity to time shifts is used to reduce computation and improve performance. Pulsed parameter estimation performance is further improved using the estimated voiced strength parameter to reduce the weighting of frequency bands which are strongly voiced when estimating the pulsed parameters. The voiced, unvoiced, and pulsed strength parameters are quantized using a weighted vector quantization method using a novel error criterion for obtaining high quality quantization. The fundamental frequency and pulse position parameters are efficiently quantized based on the quantized strength parameters.

Abstract: A lossless coding method may be used to compress information, such as audio data, without introducing any artifacts. This lossless coding method may be used to compress audio signals for use in storage and/or transmission of audio data. The audio data may be compressed by first dividing digital samples taken from the audio data into frames. A predictor is then used on the frames to generate prediction coefficients that can then be quantized to form predictor bits. The frames may then be subdivided into subsets. Another predictor can be used on the subsets to produce error samples that can be entropy coded into codeword bits. The predictor bits and codeword bits can be included in the compressed audio output for use in decoding.

Abstract: A speech signal is encoded into a set of encoded bits by digitizing the speech signal to produce a sequence of digital speech samples that are divided into a sequence of frames, each of which spans multiple digital speech samples. A set of speech model parameters are estimated for a frame. The speech model parameters include voicing parameters dividing the frame into voiced and unvoiced regions, at least one pitch parameter representing pitch for at least the voiced regions of the frame, and spectral parameters representing spectral information for at least the voiced regions of the frame. The speech model parameters are quantized to produce parameter bits. The frame is also divided into one or more subframes for which transform coefficients are computed. The transform coefficients for unvoiced regions of the frame are quantized to produce transform bits. The parameter bits and the transform bits are included in the set of encoded bits.

Abstract: Speech is encoded into a frame of bits. A speech signal is digitized into a sequence of digital speech samples that are then divided into a sequence of subframes. A set of model parameters is estimated for each subframe. The model parameters include a set of voicing metrics that represent voicing information for the subframe. Two or more subframes from the sequence of subframes are designated as corresponding to a frame. The voicing metrics from the subframes within the frame are jointly quantized. The joint quantization includes forming predicted voicing information from the quantized voicing information from the previous frame, computing the residual parameters as the difference between the voicing information and the predicted voicing information, combining the residual parameters from both of the subframes within the frame, and quantizing the combined residual parameters into a set of encoded voicing information bits which are included in the frame of bits.

Abstract: Speech is encoded into a frame of bits. A speech signal is digitized into a sequence of digital speech samples that are then divided into a sequence of subframes. A set of model parameters is estimated for each subframe. The model parameters include a set of spectral magnitude parameters that represent spectral information for the subframe. Two or more consecutive subframes from the sequence of subframes may be combined into a frame. The spectral magnitude parameters from both of the subframes within the frame may be jointly quantized.

Abstract: Speech is encoded into a 90 millisecond frame of bits for transmission across a satellite communication channel. A speech signal is digitized into digital speech samples that are then divided into subframes. Model parameters that include a set of spectral magnitude parameters that represent spectral information for the subframe are estimated for each subframe. Two consecutive subframes from the sequence of subframes are combined into a block and their spectral magnitude parameters are jointly quantized. The joint quantization includes forming predicted spectral magnitude parameters from the quantized spectral magnitude parameters from the previous block, computing the residual parameters as the difference between the spectral magnitude parameters and the predicted spectral magnitude parameters, combining the residual parameters from both of the subframes within the block, and using vector quantizers to quantize the combined residual parameters into a set of encoded spectral bits.

Abstract: The performance of digital communication over a noisy communication channel is improved. An encoder combines bit modulation with error control encoding to allow the decoder to use the redundancy in the error control codes to detect uncorrectable bit errors. This method improves the efficiency of the communication system since fewer bits are required for error control, leaving more bits available for data. In the context of a speech coding system, speech quality is improved without sacrificing robustness to bit errors. A bit prioritization method further improves performance over noisy channels. Individual bits in a set of quantizer values are arranged according to their sensitivity to bit errors. Error control codes having higher levels of redundancy are used to protect the most sensitive (highest priority) bits, while lower levels of redundancy are used to protest less sensitive bits.

Abstract: A method for encoding a speech signal into digital bits including the steps of dividing the speech signal into speech frames representing time intervals of the speech signal, determining voicing information for frequency bands of the speech frames, and determining spectral magnitudes representative of the magnitudes of the spectrum at determined frequencies across the frequency bands. The method further includes quantizing and encoding the spectral magnitudes and the voicing information. The steps of determining, quantizing and encoding the spectral magnitudes is done is such a manner that the spectral magnitudes independent of voicing information are available for later synthesizing.

Abstract: A method for decoding and synthesizing a synthetic digital speech signal from digital bits of the type produced by dividing a speech signal into frames and encoding the speech signal by an MBE based encoder. The method includes the steps of decoding the bits to provide spectral envelope and voicing information for each of the frames, processing the spectral envelope information to determine regenerated spectral phase information for each of the frames based on local envelope smoothness determining from the voicing information whether frequency bands for a particular frame are voiced or unvoiced. The method further includes synthesizing speech components for voiced frequency bands using the regenerated spectral phase information, synthesizing a speech component representing the speech signal in at least one unvoiced frequency band, and synthesizing the speech signal by combining the synthesized speech components for voiced and unvoiced frequency bands.

Abstract: A speech decoder apparatus for synthesizing a speech signal from a digitized speech bit stream of the type produced by processing speech with a speech encoder. The apparatus includes an analyzer for processing the digitized speech bit stream to generate an angular frequency and magnitude for each of a plurality of sinusoidal components representing the speech processed by the speech encoder, the analyzer generating the angular frequencies and magnitudes over a sequence of times; a random signal generator for generating a time sequence of random phase components; a phase synthesizer for generating a time sequence of synthesized phases for at least some of the sinusoidal components, the synthesized phases being generated from the angular frequencies and random phase components; and a synthesizer for synthesizing speech from the time sequences of angular frequencies, magnitudes, and synthesized phases.

Abstract: The effects of mismatch between the data rate states of at least first and second transceiver components in a signal transmission line for transmitting an original data signal are minimized by an apparatus that includes buffer means located between first and second transceivers for storing signal components, and data rate matching means for receiving a signal at a data rate that matches the data rate state of the first transceiver and transmitting a signal at a data rate that matches the data rate state of the second transceiver.

Abstract: In a speech coding and decoding system, in which a timewise segment of an acoustic speech signal is represented by a frame of a data signal characterized by a fundamental frequency and spectral harmonics, a current frame is reconstructed using a set of prediction signals based on the number of spectral harmonics for the current frame and a preceding frame and reconstructed signal parameters characterizing the preceding frame. The number of spectral harmonics for the current and preceding frames are reconstructed from at least a pair of digitally encoded signals that are generated using error protection codes for all of their bits.

Abstract: The pitch estimation method is improved. Sub-integer resolution pitch values are estimated in making the initial pitch estimate; the sub-integer pitch values are preferably estimated by interpolating intermediate variables between integer values. Pitch regions are used to reduce the amount of computation required in making the initial pitch estimate. Pitch-dependent resolution is used in making the initial pitch estimate, with higher resolution being used for smaller values of pitch. The accuracy of the voiced/unvoiced decision is improved by making the decision dependent on the energy of the current segment relative to the energy of recent prior segments; if the relative energy is low, the current segment favors an unvoiced decision; if high, it favors a voiced decision.