摘要:
A technique is used in a speech encoder (107) that reduces non-speech activity of a low bit rate digital voice message. Speech model parameters that include quantized speech spectral parameter vectors are generated in a sequence of frames. A determination is made as to which frames of the sequence of frames are voiced frames and which frames are unvoiced frames. A consecutive sequence of frames of unvoiced frames is identified (2330) as an unvoiced burst when a length, NUV, of the consecutive sequence of frames exceeds a predetermined length, Ns. A non-speech activity portion of the unvoiced burst is identified (2335-2365) and removed.
摘要:
A system controller (106) is for transferring a low bit rate digital voice message. The system controller generates from an analog voice signal representing the voice message a set of speech model parameters, and generates a first derived set of speech model parameters from a first subset of the set of speech model parameters, the first derived set encoding the voice signal at a second voice quality and second vocoder rate that are less, respectively, than a first voice quality and vocoder rate. The system controller transmits (3610) the low bit rate-digital voice message comprising the first derived set of speech model parameters to a communication receiver (114). The communication receiver requests (3640) an incremental message when the quality of the voice message is unsatisfactory. The system controller generates and transmits (3555, 3650) an incremental message-and the communication receiver uses (3660) the incremental message to generate a higher quality voice message.
摘要:
A system controller (106) includes a speech encoder (107) that dynamically segments frames of a low bit rate digital voice message. Speech model parameters have been generated in a sequence of frames. The speech model parameters include quantized speech spectral parameter vectors. The speech encoder selects (1820) a first quantized speech spectral parameter vector as a current anchor vector, selects (1820, 1830) a second quantized speech spectral parameter vector located a predetermined number of frames (LMAX) from the current anchor vector as a target speech parameter vector, and perturbs (1840) the target speech parameter vector to derive a plurality (K) of perturbed speech parameter vectors.
摘要:
A system controller (106) includes a speech encoder (107) that encodes a low bit rate digital voice message. The speech encoder sets values of words of a header of the encoded message. The values of the words define a quantity of frames in the voice message, N, and define a vocoder rate used for the encoded message. The speech encoder sets a state of each indicator in each frame status field of N frame status fields that are transmitted after the header of the encoded message. The speech encoder assembles N frame data fields, wherein each of the frame data fields comprises a set of data words. The N frame data fields follow the N frame status fields. Each set of data words conforms to at least one of the vocoder rate and the states of the indicators. A decoder (3310) decodes the encoded low bit rate digital message.
摘要:
A pitch determiner (931) of a system controller (106) that generates a smoothed pitch value for a current frame of a low bit rate voice message includes a pitch function generator (955) that generates a pitch detection function (PDF) for each frame of digital samples of a voice signal, a pitch candidate selector (960) that selects a future frame pitch candidate from a pitch detection function (PDF), and a pitch adjuster (978) that generates the smoothed pitch value. The pitch adjuster includes a subharmonic pitch corrector (965) that determines a future frame pitch value by performing pitch subharmonic correction of a future frame pitch candidate using a roughness factor of the frequency transformed window.
摘要:
A method and apparatus is provided for a low bit rate speech transmission. Speech spectral parameter vectors are generated from a voice message and stored in a sequence of speech spectral parameter vectors within a speech spectral parameter matrix. A first index identifying a first speech parameter template corresponding to a first speech spectral parameter vector of the sequence of speech spectral parameter vectors is transmitted. A subsequent speech spectral parameter vector of the sequence is selected and a subsequent speech parameter template is determined having a subsequent index. One or more intervening interpolated speech parameter templates are interpolated between the first speech parameter template and the subsequent speech parameter template. The one or more intervening speech spectral parameter vectors are compared to the corresponding one or more intervening interpolated speech parameter templates to derive a distance. The subsequent index is transmitted when the distance derived is less than or equal to a predetermined distance.
摘要:
A pitch determiner (414) for use with a speech analyzer includes a pitch function generator (414) which generates a plurality of pitch components representing a pitch function for one or more sequential segments of speech. which are represented by a predetermined number of digitized speech samples. A pitch enhancer (1116) enhances the pitch function of a current segment of speech utilizing the pitch function of one or more sequential segments of speech to generate a plurality of enhanced pitch components. A pitch detector (1118) detects the pitch of the current segment of speech by determining the pitch of an enhanced pitch component having a largest amplitude of the plurality of enhanced pitch components.
摘要:
Error detection and correction of a received message, such as a digitized voice message is achieved by generating (318) interpolated vectors for each error vector corresponding to a codebook index in a sequence of codebook indexes representing parameters of portions of the message. A plurality of error corrected candidate vectors for the vector corresponding to the codebook index in error, are generated (322,324,326) by flipping one bit in a sequence of bits representing the codebook index in error. The error corrected candidate vector which has a minimal difference from its corresponding interpolated vector is used (338) to replace the error vector. In the case of digital voice, the vectors are spectral vectors which represent spectral information for a time sample of a voice message. An ordering property of vector components is exploited to detect errors in a received codebook index without parity bits.
摘要:
An MBE synthesizer (116) for generating a segment of speech from compressed speech data received by a receiver (2004). The compressed speech data includes one or more indexes (2240, 2242) and pitch data (2248). The MBE synthesizer (116) includes the following: an excitation generator (2222) utilizing a transform function for generating transformed excitation components responsive to the pitch data (2248). A memory (3006) for storing a table of predetermined spectral vectors (2205) and associated predetermined voicing vectors (2203). A harmonic amplitude estimator (2209) that is responsive to the one or more predetermined spectra/vectors identified by the indexes (2240, 2242) received, that generates harmonic amplitude control signals. The harmonic amplitude estimator (2209) which includes a peak detector (2503), a peak enhancer (2505), a valley detector (2507), a valley enhancer (2509). A multi-band voicing controller (2214), responsive to the predetermined voicing vectors which are associated with the one or more predetermined spectral vectors identified, for controlling a selection of the excitation components.
摘要:
An apparatus codes excitation parameters for very low bit rate voice messaging using a method that processes a voice message to generating speech parameters. The speech parameters are separated (316) to produce a first group of energy parameters and a second group of pitch and voicing parameters. Subsequently, the first group of energy parameters are encoded and compressed using a non-uniform root-mean-square scalar process (318) to create a first plurality of encoded data. Additionally, the second group of pitch and voicing parameters are compressed, encoded, and combined into a single parameter using a three slope vector encoding process (320) that creates a second plurality of encoded data. Finally, the first and second plurality of encoded data are multiplexed (322) to create a multiplexed signal for transmission, the multiplexed signal representing the voice message.