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 of encoding speech by analyzing a digitized speech signal to determine excitation parameters for the digitized speech signal is disclosed. The method includes dividing the digitized speech signal into at least two frequency bands, determining a first preliminary excitation parameter by performing a nonlinear operation on at least one of the frequency band signals to produce a modified frequency band signal and determining the first preliminary excitation parameter using the modified frequency band signal, determining a second preliminary excitation parameter using a method different from the first method, and using the first and second preliminary excitation parameters to determine an excitation parameter for the digitized speech signal. The method is useful in encoding speech. Speech synthesized using the parameters estimated based on the invention generates high quality speech at various bit rates useful for applications such as satellite voice communication.

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 of encoding speech analyzes a digitized speech signal to determine excitation parameters for the digitized speech signal. The method includes dividing the digitized speech signal into at least two frequency bands, performing a nonlinear operation on at least one of the frequency bands to produce a modified frequency band, and determining whether the modified frequency band is voiced or unvoiced. The nonlinear operation is an operation that emphasizes a fundamental frequency of the digitized speech signal so that the modified frequency band signal includes a component corresponding to the fundamental frequency even when the at least one frequency band signal does not include such a component.

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.

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: The quantized parameter bits are grouped into several categories according to their sensitivity to bit errors. More effective error correction codes are used to encode the most sensitive parameter bits, while less effective error correction codes are used to encode the less sensitive parameter bits. This method improves the efficiency of the error correction and improves the performance if the total bit rate is limited. The perceived quality of coded speech is improved. A smoothed spectral envelope is created in the frequency domain. The ratio between the actual spectral envelope and the smoothed spectral envelope is used to enhance the spectral envelope. This reduces distortion which is contained in the spectral envelope.

Abstract: The performance of speech coding in the presence of bit errors is improved. The quantized parameter bits are grouped into several categories according to their sensitivity to bit errors. More effective error correction codes are used to encode the most sensitive parameter bits, while less effective error correction codes are used to encode the less sensitive parameter bits. This method improves the efficiency of the error correction and improves the performance if the total bit rate is limited. The perceived quality of coded speech is improved. A smoothed spectral envelope is created in the frequency domain. The ratio between the actual spectral envelope and the smoothed spectral envelope is used to enhance the spectral envelope. This reduces distortion which is contained in the spectral envelope.

Abstract: The redundancy contained within the spectral amplitudes is reduced, and as a result the quantization of the spectral amplitudes is improved. The prediction of the spectral amplitudes of the current segment from the spectral amplitudes of the previous is adjusted to account for any change in the fundamental frequency between the two segments. The spectral amplitudes prediction residuals are divided into a fixed number of blocks each containing approximately the same number of elements. A prediction residual block average (PRBA) vector is formed; each element of the PRBA is equal to the average of the prediction residuals within one of the blocks. The PRBA vector is vector quantized, or it is transformed with a Discrete Cosine Transform (DCT) and scalar quantized. The perceived effect of bit errors is reduced by smoothing the voiced/unvoiced decisions. An estimate of the error rate is made by locally averaging the number of correctable bit errors within each segment.

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.