592 results about "Timbre" patented technology

An analysis and synthesis system for sound is provided that can independently modify characteristics of audio signals such as pitch, duration, and timbre. High-quality pitch-scaling and time-scaling are achieved by using a technique for sinusoidal phase compensation adapted to a sinusoidal representation. Such signal modification systems can avoid the usual problems associated with interpolation-based re-sampling so that the pitch-scaling factor and the time-scaling factor can be varied independently, arbitrarily, and continuously. In the context of voice modification, the sinusoidal representation provides a means with which to separate the acoustic contributions of the vocal excitation and the vocal tract, which can enable independent timbre modification of the voice by altering only the vocal tract contributions. The system can be applied to efficiently encode the pitch in sinusoidal models by compensating for pitch quantization errors. The system can also be applied to non-speech signals such as music.

A music information retrieval system of the present invention can retrieve unknown songs including singing voices having similar voice timbres. Voice timbre features of the songs and identifiers for the respective songs are stored in voice timbre feature storage section 2. When one of the songs is selected, similarity calculation section 3 calculates voice timbre similarities between the selected song and the respective remaining songs, based on voice timbre features of the selected song and the other songs. Similar song retrieval and display section 5 displays on a display 10 a plurality of identifiers for songs which are similar to the selected song in voice timbre. Song data reproduction section 6 reproduces song data corresponding to one or more identifiers selected from among the plurality of identifiers displayed on the display 10.

A concept, method and apparatus for detecting and correcting an accent by means of sound morphing is provided. The input audio signal is analyzed for finding pre-specified unwanted speech patterns, i.e. phonemes or groups of phonemes that are to be corrected, for instance because they represent a foreign accent. These unwanted sounds are then modified or completely replaced by the pre-stored replacement audio patterns, adjusted to the current pitch and voice timbre of the user. The degree of speech modification, i.e. the set of phonemes to be modified, can be set at a desired level. The system works in two modes: first, learning, i.e. storing the unwanted and the replacement phoneme patterns, and second, the correction mode which performs phoneme modification based on the stored information. The implementation is both in software and hardware. The hardware apparatus is based on parallel signal processing and therefore allows for real-time accent correcting of variable complexity, up to multiple-user multiple-accent super-complex systems based on mesh architecture of multiple chips and boards, possibly as a part of a telephone or another networking system.

A method and apparatus for classifying mood of music at high speed. The method includes: extracting a Modified Discrete Cosine Transformation-based timbre feature from a compressed domain of a music file; extracting a Modified Discrete Cosine Transformation-based tempo feature from the compressed domain of the music file; and classifying the mood of the music file based on the extracted timbre feature and the extracted tempo feature.

A tone control device applied to an electronic wind instrument realizes an octave-changeover-blowing technique in which the same note is produced with different octaves respectively by use of the same fingering state, thus increasing controllable ranges with regard to the tone volume, tone color, and tone pitch. A plurality of flow sensors are arranged in proximity to an edge with which a jet flow caused by blowing air into a blow hole of a lip plate collides within a tube of a wind instrument controller simulating an air-reed instrument. The flow sensors are horizontally arranged to detect a jet width, thus controlling the tone volume; and the flow sensors are vertically arranged to detect a jet eccentricity or a jet thickness, thus controlling the tone color. Ascending or descending of the tone pitch by octaves is controlled by use of the flow sensor and a jet length sensor.

Sound reproduction according to the present disclosure integrates an additional sound field, direct surround sound, to existing diffuse or ambient surround sound systems. Direct surround sound permits point source sounds to be located in an ambient or diffuse surround field and or to be moved within the surround field with control of the sound timbre. Sound mixing equipment may include two or more direct surround channels to permit sound designers and engineers to add, locate and move point source sounds within the surround field. The inclusion of two or more point source loudspeakers improves the surround array by providing direct sources that may be localizable and timbre matched to the screen channel speakers. This in turn provides filmmakers and other content creators a new tool they can use during the creation of content such as movie soundtracks. In another aspect of the present disclosure, a hybrid speaker may receive diffuse surround sound input and direct surround sound input and reproduce both sound fields from a single apparatus.

A method and a system for implementing interconversion between a ring back tone and a ringing tone are disclosed. The method for converting the ring back tone into the ringing tone includes: configuring ring back tone information for a user according to a ring back tone configuration request; obtaining relevant data of corresponding ringtone information according to the ring back tone information configured, and sending the relevant data to a user terminal. The method for converting the ringing tone into the ring back tone includes: generating a ring back tone configuration request according to ringing tone configuration information, where a user terminal configures the ringing tone with the ringing tone configuration information; and configuring the ring back tone corresponding to the ringing tone to be a ring back tone information of the user terminal according to the ringing tone configuration information in the ring back tone configuration request.

The invention relates to a music voice tone changing method for stabilizing tone quality, and the method comprises the steps: exporting a spectrum envelope through employing a voice signal, which can be divided into a glottis stimulation component and a channel stimulation response component, and the cepstrum sequence of the voice signal; separating the stimulation component of the voice signal through employing the spectrum envelope; carrying out the processing of the stimulation components of the voice signal through a tone changing algorithm, and changing the pitch of the voice signal; finally enabling the stimulation components to be synthesized again after the spectrum envelope and the pitch are changed, and obtaining a music voice signal with the changed pitch and the stable tone quality. The method provided by the invention is simple, is flexible in implementation, and is higher in practicality.

The invention relates to an electronic musical instrument simulation learning tool. A bottom circuit 2, a baffle 3 and a top circuit 4 are made of three-layered fpc plate flexible circuit material; through pressing keys on the bottom circuit 2 and the top circuit 4, circuit connectors are contacted with a switch through baffle key holes 20, signals are processed to digital signals via a usb interface 6 to be inputted to a multimedia computer, the digital signals are converted to sounds corresponding to a digital musical instrument music score, sounds including those of a violin, a piano, a harmonica, an erhu and the like can be played, 88keys, the three-layered fpc keyboard circuit boards 2 and 4, and the baffle 3 are pressed downwardly and fixed on the surface of a finger board 1 through bolts 8 and bolt holes 23, the keys are distributed corresponding to the sound positions of the bowstring musical instrument, and three parts of software, terminal simulation hardware and a simulation blanket are included.

A performance apparatus 11 is provided with a first acceleration sensor 22 at its head portion and a second acceleration sensor 23 at its base portion. CPU 21 determines operation of the performance apparatus 11 during a period from the first timing to the second timing, based on a first acceleration-sensor value and a second acceleration-sensor value, and determines operation mode corresponding to the operation of the performance apparatus. Referring to a tone-color table stored in RAM 26, CPU 21 determines a tone color of musical tones to be generated, based on the determined operation mode.

A guitar-family-characteristic-sound selector switch (63) is incorporated into an electric guitar and used in conjunction with the normal guitar pickup selector switch (62) and the volume and tone controls. This guitar-family-characteristic-sound selector switch invokes different pickup (61) combinations and resistance/capacitance/inductance circuitry (68) in order to generate sounds characteristic of at least two other guitars in addition to the normal sound of the host guitar. In one implementation, the family selector switch is a single user controllable switch that makes the system is intuitive in that a single guitar-family-characteristic-sound selector switch is all that needs to be operated to put the guitar in the mode of sounding like one of three or more styles of guitars following such selection; the pickup selector, volume and tone controls are used in the usual manner.

The present invention is a method and system to convert speech signal into a parametric representation in terms of timbre vectors, and to recover the speech signal thereof. The speech signal is first segmented into non-overlapping frames using the glottal closure instant information, each frame is converted into an amplitude spectrum using a Fourier analyzer, and then using Laguerre functions to generate a set of coefficients which constitute a timbre vector. A sequence of timbre vectors can be subject to a variety of manipulations. The new timbre vectors are converted back into voice signals by first transforming into amplitude spectra using Laguerre functions, then generating phase spectra from the amplitude spectra using Kramers-Knonig relations. A Fourier transformer converts the amplitude spectra and phase spectra into elementary acoustic waves, then superposed to become the output voice. The method and system can be used for voice transformation, speech synthesis, and automatic speech recognition.

A Plurality of key switches for tone-generating are arranged two-dimensionally. Mode setting section sets a tone adjusting mode in which the key switches are caused to function as tone-adjusting operators. In the tone adjusting mode, adjustment of a predetermined tone factor (e.g., tone pitch, tone length, tone volume or tone color) is permitted in response to operation of the key switch. For example, once a user moves a finger to change a Y-coordinate position of the key switch in the tone adjusting mode, an amount of the movement, i.e. a difference between Y-coordinates of two or more successively-operated key switches, is detected, and the thus-detected movement amount is set as a value for adjusting a tone volume or the like. All of light-emitting elements located at Y-coordinate positions of the key switches may be illuminated in a line, to allow the user to visually confirm the adjustment and operation.

The invention discloses a voice changing method and device. The method comprises the following steps: receiving a source speaker statement; extracting voice recognition acoustic features and voice synthesis acoustic features from the source speaker statement; utilizing the voice recognition acoustic features to obtain voice recognition hidden layer features; utilizing the voice synthesis acousticfeatures to obtain voice synthesis coding features; inputting the voice recognition hidden layer features and the voice synthesis coding features into a pre-constructed tone conversion model corresponding to a specific target speaker to obtain voice synthesis acoustic features of the specific target speaker; and generating an audio signal of the specific target speaker by using the voice synthesisacoustic feature of the specific target speaker. According to the scheme of the invention, the voice changing method and device can achieve the conversion from any source speaker voice to the targetspeaker voice, and is better in voice changing effect.

A data sonification system for representing a plurality of channels of numerical information is described. The data sonification system includes a plurality of audio waveform generator elements. Each of the audio waveform generator elements generates an associated audio frequency waveform. Each audio frequency waveform has an audio frequency parameter and at least one timbre modulation parameter having a settable value. The timbre modulation parameter affects the timbre of the audio waveform. The data sonification system includes a mapping element for associating aspects of multidimensional numerical data with the timbre modulation parameter of each audio frequency waveform. The mapping element sets the value of the timbre modulation parameter in response to multidimensional numerical data.

The invention provides a method for singing imitation. The method comprises the steps that corresponding audio materials of a source singer and a target singer are prepared; voice features of the source singer and voice features of the target singer are analyzed by using a STRAIGHT model; a joint GMM model of the source singer and the target singer is obtained through training by using a Gaussian mixture model; Gaussian transform functions based on inter-frame correlation are used in the tone conversion process; the tone of the source singer and the tone of the target singer are mixed in proportion; voice having the tone of the target singer is reconstructed by using the STRAIGHT model. The invention further provides a device for implementing the method. According to the method and device, the voice of the source singer can be converted into the voice having the tone of the target singer, converted tone quality is good, the tone is close to the tone of the target singer, and the tone features of the target singer can be added in proportion; especially, when the target singer is a music star, the self-confidence and interestingness of singing for the user at a digital audio and video place can be improved greatly, and meanwhile the level of the user for imitating the tone of the music star is improved.

The embodiment of the invention discloses an individualized song recommending system based on vocal music characteristics. The method includes the following steps of extracting characteristics, wherein the voice register characteristics, speed characteristics and tone characteristics of singing data are extracted, the voice register characteristics include absolute voice register and relative voice register, the speed characteristics include the number of beats per minute, and the tone characteristics include a Gaussian mixture model of Mel frequency cepstrum coefficient training; recommending songs systematically, wherein a corresponding song in a music library is found through a key sound matching algorithm according to a snatch sung by a user, and voice register conformity detection, song conformity detection and singer conformity detection are conducted. Singer recommending and song recommending are conducted according to extracted user characteristics. By means of the method, whether a current song is suitable for being sung by the user or not can be evaluated, and singers matched with the user vocal music ability and songs suitable for being sung by the user are further recommended. From the perspective of user singing, the traditional music recommending range is expanded, and higher practical value is achieved.

Typically, changes in timbre in an audio reproduction system use analogue or digital filters, either shelving filters (as in the case of bass, treble and loudness controllers) and/or peaking filters (as in the case of speed-and noise-compensation in automotive systems). A problem associated with this implementation lies in the audibility of the phase shift caused by these filters. According to the invention, this and further problems are solved by utilising the crossover network of the system itself—with level adjustable output signals—to attain desired timbre and/or dynamic range adjustments and hence avoiding undesirable phase effects of the traditional bass and treble control filters. An input signal (18) is according to an embodiment of the invention divided by a cross over network (20) into a number of frequency bands and applied via level control means (21, 22, 23) and separate power amplifiers (24, 25, 26) to loudspeaker drivers (27, 28, 29). In this manner, the mentioned phase shifts can be avoided and by proper control of the level adjustment means by a frequency-dependent analysis and weighting network (30), desired adjustment of timbre and dynamic range of the audio reproduction can be achieved. Also control dependent on vehicle speed and/or background noise can be incorporated into the system..

The invention discloses a speech synthesis method and device, a model training method and device, equipment and a storage medium, and relates to the technical field of artificial intelligence such asmachine learning and intelligent speech. The specific implementation scheme is as follows: based on text information, timbre information and rhythm information of to-be-synthesized speech, generatingacoustic feature information of the to-be-synthesized speech by adopting a pre-trained speech synthesis model; based on the acoustic feature information of the to-be-synthesized voice, synthesizing the corresponding voice by adopting a pre-trained vocoder. By adopting the technical scheme of the invention, when the voice is synthesized, any text information, tone information and rhythm informationcan be randomly combined and synthesized into the desired voice, and the voice synthesis mode is very flexible and convenient.