Class / Patent application number | Description | Number of patent applications / Date published |
704268000 | Frequency element | 20 |
20080228487 | SPEECH SYNTHESIS APPARATUS AND METHOD - A language processing unit identifies a word by performing language analysis on a text supplied from a text holding unit. A synthesis selection unit selects speech synthesis processing performed by a rule-based synthesis unit or speech synthesis processing performed by a pre-recorded-speech-based synthesis unit for a word of interest extracted from the language analysis result. The selected rule-based synthesis unit or pre-recorded-speech-based synthesis unit executes speech synthesis processing for the word of interest. | 09-18-2008 |
20090076822 | Audio signal transforming - A sequence is received of time domain digital audio samples representing sound (e.g., a sound generated by a human voice or a musical instrument). The time domain digital audio samples are processed to derive a corresponding sequence of audio pulses in the time domain. Each of the audio pulses is associated with a characteristic frequency. Frequency domain information is derived about each of at least some of the audio pulses. The sound represented by the time domain digital audio samples is transformed by processing the audio pulses using the frequency domain information. | 03-19-2009 |
20090204405 | METHOD, APPARATUS AND PROGRAM FOR SPEECH SYNTHESIS - Apparatus and method for generating high quality synthesized speech having smooth waveform concatenation. The apparatus includes a pitch frequency calculation section, a pitch synchronization position calculation section, a unit waveform storage, a unit waveform selection section, a unit waveform generation section, and a waveform synthesis section. The unit waveform generation section includes a conversion ratio calculation section, a sampling rate conversion section, and a unit waveform re-selection section. The conversion ratio calculation section calculates a sampling rate conversion ratio from the pitch information and the position of pitch synchronization, and the sampling rate conversion section converts the sampling rate of the unit waveform, delivered as input, based on the sampling rate conversion ratio. The unit waveform re-selection section selects, from the sampling-rate-converted unit waveform, the unit waveform having a phase necessary to obtain a synthesized speech waveform which will exhibit smooth waveform concatenation. | 08-13-2009 |
20090265173 | TONE DETECTION FOR SIGNALS SENT THROUGH A VOCODER - A tone detector and associated method for use with EVRC-B and GSM vocoders to enable reliable detection of system connect tones over a wireless communication system. The tone detection method examines a number of sequential data frames of the signal received from the vocoder and determines that the tone is present if the spectral energy at frequencies around the tone is much higher than that at neighboring frequencies and if the calculated center frequency of the data frames is at or near the frequency of the tone. | 10-22-2009 |
20090326951 | SPEECH SYNTHESIZING APPARATUS AND METHOD THEREOF - Ratios of powers at the peaks of respective formants of the spectrum of a pitch-cycle waveform and powers at boundaries between the formants are obtained and, when the ratios are large, bandwidth of window functions are widened and the formant waveforms are generated by multiplying generated sinusoidal waveforms from the formant parameter sets on the basis of pitch-cycle waveform generating data by the window functions of the widened bandwidth, whereby a pitch-cycle waveform is generated by the sum of these formant waveforms. | 12-31-2009 |
20100324907 | ATTENUATION OF OVERVOICING, IN PARTICULAR FOR THE GENERATION OF AN EXCITATION AT A DECODER WHEN DATA IS MISSING - The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A′,B′,C′,D′) of at least two samples and inverting positions of samples in the groups, randomly (B′,C′) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated. | 12-23-2010 |
20110046958 | METHOD AND APPARATUS FOR EXTRACTING PROSODIC FEATURE OF SPEECH SIGNAL - The present invention discloses a method and an apparatus for extracting a prosodic feature of a speech signal, the method including: dividing the speech signal into speech frames; transforming the speech frames from time domain to frequency domain; and extracting respective prosodic features for different frequency ranges. According to the above technical solution of the present invention, it is possible to effectively extract the prosodic feature which can combine with a traditional acoustics feature without any obstacle. | 02-24-2011 |
20110196680 | SPEECH SYNTHESIS SYSTEM - When a system ( | 08-11-2011 |
20110218810 | System for Controlling Digital Effects in Live Performances with Vocal Improvisation - A system for controlling digital effects in live performances with vocal improvisation is described. The system features a complex controller that in one embodiment utilizes several magnetically activated electronic switches attached to a glove that is worn by an artist during a live performance. The switches are activated by a permanent magnet that is also attached to the switch bearing glove and a second magnet attached to a glove worn on the opposite hand. Furthermore, the switches are wirelessly connected by a miniature, battery-operated wireless data communications unit to a digital vocal processor unit that provides a dual mode, multi-channel phrase looping capability wherein individual channels can be selected for re-recording and selected banks of channels can be deleted during the performance. This combination of features allows a complex sequence of digital effects to be controlled by the artist during a performance while maintaining the freedom of movement desired to enhance the performance. | 09-08-2011 |
20120209611 | SPEECH SIGNAL RESTORATION DEVICE AND SPEECH SIGNAL RESTORATION METHOD - A synthesis filter | 08-16-2012 |
20130151255 | METHOD AND DEVICE FOR EXTENDING BANDWIDTH OF SPEECH SIGNAL - A method for extending a bandwidth of a speech signal received, according to an embodiment of the present invention, includes: transforming the received speech signal into a frequency domain by decoding the received speech signal; normalizing the transformed speech signal; differentiating a voiced sound period or unvoiced sound period from the received speech signal; extracting, from the normalized speech signal, a first period including a harmonic component of the voiced sound period on the basis of the voiced sound period; extracting, from the normalized speech signal, a second period on the basis of correlation between the unvoiced sound period and the normalized speech signal; generating a high-band speech signal on the basis of the first period and the second period; and synthesizing the generated high-band speech signal and the transformed speech signal to output a wideband speech signal. | 06-13-2013 |
20130151256 | SYSTEM AND METHOD FOR SINGING SYNTHESIS CAPABLE OF REFLECTING TIMBRE CHANGES - Herein provided is a system for singing synthesis capable of reflecting not only pitch and dynamics changes but also timbre changes of a user's singing. A spectral transform surface generating section | 06-13-2013 |
20130262122 | SPEECH RECEIVING APPARATUS, AND SPEECH RECEIVING METHOD - Disclosed is a speech receiving apparatus. A low-band PLC module and a synthesis filter reconstructs a low-band speech signal of a lost frame from a previous good frame. A high-band PLC module reconstructs a high-band speech signal of the lost frame from the previous good frame. A transforming part transforms the low-band speech signal into a frequency range. A bandwidth extending part generates at least an extended MDCT coefficient as information for the high-band speech signal from the low-band speech signal transformed by the transforming part. A smoothing part smoothes the extended MDCT coefficient. An inverse transforming part inversely transforms the extended MDCT coefficient smoothed by the smoothing part to a time domain. A synthesizing part synthesizes the low-band speech signal, and the high-band speech signal which is inverse-transformed by the inverse transforming part and reconstructed, to output a wideband speech signal. | 10-03-2013 |
20130311189 | VOICE PROCESSING APPARATUS - In a voice processing apparatus, a processor performs generating a converted feature by applying a source feature of source voice to a conversion function, generating an estimated feature based on a probability that the source feature belongs to each element distribution of a mixture distribution model that approximates distribution of features of voices having different characteristics, generating a first conversion filter based on a difference between a first spectrum corresponding to the converted feature and an estimated spectrum corresponding to the estimated feature, generating a second spectrum by applying the first conversion filter to a source spectrum corresponding to the source feature, generating a second conversion filter based on a difference between the first spectrum and the second spectrum, and generating target voice by applying the first conversion filter and the second conversion filter to the source spectrum. | 11-21-2013 |
20130339023 | Enhancing Performance of Spectral Band Replication and Related High Frequency Reconstruction Coding - The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems. | 12-19-2013 |
20140222434 | AUDIO SIGNAL SYNTHESIZER AND AUDIO SIGNAL ENCODER - An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation. | 08-07-2014 |
20140288938 | SYSTEMS AND METHODS FOR ENHANCING PLACE-OF-ARTICULATION FEATURES IN FREQUENCY-LOWERED SPEECH - To improve the intelligibility of speech for users with high-frequency hearing loss, the present systems and methods provide an improved frequency lowering system with enhancement of spectral features responsive to place-of-articulation of the input speech. High frequency components of speech, such as fricatives, may be classified based on one or more features that distinguish place of articulation, including spectral slope, peak location, relative amplitudes in various frequency bands, or a combination of these or other such features. Responsive to the classification of the input speech, a signal or signals may be added to the input speech in a frequency band audible to the hearing-impaired listener, said signal or signals having predetermined distinct spectral features corresponding to the classification, and allowing a listener to easily distinguish various consonants in the input. | 09-25-2014 |
20150325232 | SPEECH SYNTHESIZER, AUDIO WATERMARKING INFORMATION DETECTION APPARATUS, SPEECH SYNTHESIZING METHOD, AUDIO WATERMARKING INFORMATION DETECTION METHOD, AND COMPUTER PROGRAM PRODUCT - According to an embodiment, a speech synthesizer includes a source generator, a phase modulator, and a vocal tract filter unit. The source generator generates a source signal by using a fundamental frequency sequence and a pulse signal. The phase modulator modulates, with respect to the source signal generated by the source generator, a phase of the pulse signal at each pitch mark based on audio watermarking information. The vocal tract filter unit generates a speech signal by using a spectrum parameter sequence with respect to the source signal in which the phase of the pulse signal is modulated by the phase modulator. | 11-12-2015 |
20150371641 | ENHANCED AUDIO FRAME LOSS CONCEALMENT - A method is provided for concealing a lost audio frame of a received audio signal by performing a sinusoidal analysis of a part of a previously received or reconstructed audio signal. The sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, and applying a sinusoidal model on a segment of the previously received or reconstructed audio signal. The segment is used as a prototype frame in order to create a substitution frame for a lost audio frame. The method includes creating the substitution frame for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies. The method further includes performing at least one of an enhanced frequency estimation and an adaptation of the creating of the substitution frame in response to the tonality of the audio signal. | 12-24-2015 |
20150371642 | AUDIO FRAME LOSS CONCEALMENT - Concealing a lost audio frame of a received audio signal by performing a sinusoidal analysis of a part of a previously received or reconstructed audio signal, wherein the sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, applying a sinusoidal model on a segment of the previously received or reconstructed audio signal, wherein said segment is used as a prototype frame in order to create a substitution frame for a lost audio frame, and creating the substitution frame for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies. | 12-24-2015 |