Patent classifications
G10L21/057
Voice signal processing apparatus and voice signal processing method
A voice signal processing apparatus and a voice signal processing method are provided. A last sampling point of an m.sup.th original frequency-lowered signal frame is determined according to a phase reference sampling point number of the m.sup.th original frequency-lowered signal frame. Here, the phase reference sampling point number corresponds to a middle sampling point of an m.sup.th renovating frequency-lowered signal frame, and the last sampling point is phase-matched with a sampling point corresponding to the phase reference sampling point number in the m.sup.th original frequency-lowered signal frame. P consecutive sampling points starting from the last sampling point are applied as sampling points of an (m+1).sup.th renovating frequency-lowered signal frame.
SOUND ENHANCEMENT THROUGH REVERBERATION MATCHING
Embodiments of the present invention relate to enhancing sound through reverberation matching. In sonic implementations, a first sound recording recorded in a first environment is received. The first sound recording is decomposed to a first clean signal and a first reverb kernel. A second reverb kernel corresponding with a second sound recording recorded in a second environment is accessed, for example, based on a user indication to enhance the first sound recording to sound as though recorded in the second environment. An enhanced sound recording is generated based on the first clean signal and the second reverb kernel. The enhanced sound recording is a modification of the first sound recording to sound as though recorded in the second environment.
SOUND ENHANCEMENT THROUGH REVERBERATION MATCHING
Embodiments of the present invention relate to enhancing sound through reverberation matching. In sonic implementations, a first sound recording recorded in a first environment is received. The first sound recording is decomposed to a first clean signal and a first reverb kernel. A second reverb kernel corresponding with a second sound recording recorded in a second environment is accessed, for example, based on a user indication to enhance the first sound recording to sound as though recorded in the second environment. An enhanced sound recording is generated based on the first clean signal and the second reverb kernel. The enhanced sound recording is a modification of the first sound recording to sound as though recorded in the second environment.
Speech processing device and method
A speech processing device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: obtaining input speech, detecting a vowel segment contained in the input speech, estimating an accent segment contained in the input speech, calculating a first vowel segment length containing the accent segment and a second vowel segment length excluding the accent segment, and controlling at least one of the first vowel segment length and the second vowel segment length.
Speech processing device and method
A speech processing device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: obtaining input speech, detecting a vowel segment contained in the input speech, estimating an accent segment contained in the input speech, calculating a first vowel segment length containing the accent segment and a second vowel segment length excluding the accent segment, and controlling at least one of the first vowel segment length and the second vowel segment length.
SYSTEMS, METHODS AND DEVICES FOR INTELLIGENT SPEECH RECOGNITION AND PROCESSING
Systems, methods, and devices for intelligent speech recognition and processing are disclosed. According to one embodiment, a method for improving intelligibility of a speech signal may include (1) at least one processor receiving an incoming speech signal comprising a plurality of sound elements; (2) the at least one processor recognizing a sound element in the incoming speech signal to improve the intelligibility thereof; (3) the at least one processor processing the sound element by at least one of modifying and replacing the sound element; and (4) the at least one processor outputting the processed speech signal comprising the processed sound element.
Method and device for processing audio signals
Method and device of processing audio signals are disclosed. The method includes: obtaining a set of data, the set of data comprising LSP parameters for an audio signal; determining a set of sampling data points from the set of LSP parameters using a predetermined sampling rule, the set of sampling data points including spectrum amplitude values for a plurality of sampled frequency values; identifying one or more local maxima among the set of sampling data points, and a respective preceding local minimum and a respective succeeding local minimum for each of the identified local maxima; for each of the identified local maxima, shifting one or more of the set of data comprising LSP parameters located between the respective preceding local minimum and the respective succeeding local minimum of an identified local maximum towards the identified local maximum; and adjusting the set of data comprising LSP parameters using an energy coefficient.
Method and device for processing audio signals
Method and device of processing audio signals are disclosed. The method includes: obtaining a set of data, the set of data comprising LSP parameters for an audio signal; determining a set of sampling data points from the set of LSP parameters using a predetermined sampling rule, the set of sampling data points including spectrum amplitude values for a plurality of sampled frequency values; identifying one or more local maxima among the set of sampling data points, and a respective preceding local minimum and a respective succeeding local minimum for each of the identified local maxima; for each of the identified local maxima, shifting one or more of the set of data comprising LSP parameters located between the respective preceding local minimum and the respective succeeding local minimum of an identified local maximum towards the identified local maximum; and adjusting the set of data comprising LSP parameters using an energy coefficient.
HEARING DEVICE COMPRISING AN ADAPTIVE FILTER BANK
A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h (n), where m=0, 1, . . . , M1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.
HEARING DEVICE COMPRISING AN ADAPTIVE FILTER BANK
A hearing device comprises a) at least one input transducer configured to pick up sound from an acoustic environment around the user when the user is wearing the hearing device, the at least one input transducer providing at least one electric input signal representative of said sound, b) at least one analysis filter bank configured to provide said at least one electric input signal as a multitude of frequency sub-band signals, the at least one analysis filter bank comprising b1) a plurality of M first filters h.sub.m(n), whose impulse responses are modulated from a first prototype filter h (n), where m=0, 1, . . . , M1 is a frequency band index, and n is a time index, c) a processor for processing said at least one electric input signal provided by said at least one analysis filter bank, or a signal originating therefrom, and providing a processed signal, d) an output transducer configured to provide stimuli perceivable as sound to the user in dependence of said processed signal, and e) a controller for controlling said analysis filter bank by applying a different first prototype filter to said at least one filter bank in dependence of said current acoustic environment. A method of operating a hearing device is further disclosed.