1075 results about "Voice pitch" patented technology

An audio privacy system reduces the intelligibility of speech in an audio signal while preserving prosodic information, such as pitch, relative energy and intonation so that a listener has the ability to recognize environmental sounds but not the speech itself. An audio signal is processed to separate non-vocalic information, such as pitch and relative energy of speech, from vocalic regions, after which syllables are identified within the vocalic regions. Representations of the vocalic regions are computed to produce a vocal tract transfer function and an excitation. The vocal tract transfer function for each syllable is then replaced with the vocal tract transfer function from another prerecorded vocalic sound. In one aspect, the identity of the replacement vocalic sound is independent of the identity of the syllable being replaced. A modified audio signal is then synthesized with the original prosodic information and the modified vocal tract transfer function to produce unintelligible speech that preserves the pitch and energy of the speech as well as environmental sounds.

An acoustic control system for an electronic musical instrument, which can obtain optimal acoustic characteristics irrespective of whether or not a low range speaker is attached to a body thereof. A piano body has high range and mid range speakers. A low range speaker is removable from the body. A stand-excluding switch designates a first speaker-use mode for using the high range and mid range speakers alone, and a stand-including switch designates a both speaker-use mode for using both the high range and mid range speakers and the low range speaker. A ROM stores stand-excluding and stand-including factors for setting acoustic characteristics in the two modes, respectively. A CPU reads one of the factors, which corresponds to the mode designated by the stand-excluding or stand-including switch. A tone generator circuit generates a musical tone signal to be reproduced in one of the modes, based on the read factor.

A single platform for controller functionality for each of security, monitoring and automation, as well as providing a capacity to function as a bidirectional Internet gateway, is provided. Embodiments of the present invention provide such functionality by virtue of a configurable architecture that enables a user to adapt the system for the user's specific needs. Such configurability includes associating selected audio event tones with selected events associated with sensors and zones coupled to the system.

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.

A system and method is provided that employs a frequency domain interpolative CODEC system for low bit rate coding of speech which comprises a linear prediction (LP) front end adapted to process an input signal that provides LP parameters which are quantized and encoded over predetermined intervals and used to compute a LP residual signal. An open loop pitch estimator adapted to process the LP residual signal, a pitch quantizer, and a pitch interpolator and provide a pitch contour within the predetermined intervals is also provided. Also provided is a signal processor responsive to the LP residual signal and the pitch contour and adapted to perform the following: provide a voicing measure, where the voicing measure characterizes a degree of voicing of the input speech signal and is derived from several input parameters that are correlated to degrees of periodicity of the signal over the predetermined intervals; extract a prototype waveform (PW) from the LP residual and the open loop pitch contour for a number of equal sub-intervals within the predetermined intervals; normalize the PW by a gain value of the PW; encode a magnitude of the PW; and separate stationary and nonstationary components of the PW using a low complexity alignment process and a filtering process that introduce no delay. The ratio of the energy of the nonstationary component of the PW to that of the stationary component of the PW is averaged across 5 subbands to compute the nonstationarity measure as a frequency dependent vector entity. A measure of the degree of voicing of the residual is also computed using openloop pitchgain, pitch variance, relative signal power, PW correlation and PW nonstationarity in low frequency subbands. The nonstationarity measure and voicing measure are encoded using a 6-bit spectrally weighted vector quantization scheme using a codebook partitioned based on a voiced/unvoiced decision. At the decoder, a stationary component of PW is reconstructed as a weighted combination of the previous PW phase vector, a random phase perturbation and a fixed phase vector obtained from a voiced pitch pulse.

To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. A speech encoder employing various encoding schemes based upon parameters including an available transmission bit rate. In addition, the speech encoder is operable to identify and apply an optimal encoding scheme for a given speech signal. The speech encoder may be applied code-excited linear prediction when the available bit rate is above a predetermined upper threshold. Pitch preprocessing, including continuous warping, may be applied when it is below a predetermined lower threshold. The encoder considers varying characteristics of the speech signal including the long term prediction mode of a previous frame, and a spectral difference between the line spectral frequencies of a current and a previous frame, a predicted pitch lag, an open loop pitch lag, a closed loop pitch lag, a pitch gain, and a pitch correlation.

A self-contained, real-time self-use gait therapy device with a gait sensor, actuator, output speaker, and battery receptacle enclosed in a component case with a belt clip. Based upon step duration, step impact force, or step form data generated by the gait sensor, the actuator drives the speaker, producing beeps on a real-time basis with the pitch of each beep being a function of the step duration, step impact force, or step form for each step of the user. The speaker output is monitored and used by the user on a real-time basis to modify and improve his or her gait.

A karaoke apparatus includes a sound effect processing system provided in a microprocessor. The system decodes standard song data from an internal storage or an external storage connected to an extended system interface by a song decoding module; corrects pitches of sing voices by a pitch correcting system, so the pitches of the singing voices are corrected to the pitches of the standard song or close to the pitches of the standard song. The singing voices are processed with harmony adding, tonal modification and speed-changing by a harmony adding system to produce an effect of chorus being composed of three voice parts. A pitch evaluating system is used for comparing the pitch sequence of the singing voices with the pitch sequence of the standard song to draw a voice graph so as to visually show a difference between the pitches of the singing voices and the pitches of the standard song, while providing score and comment of the singing voices. Therefore, a singer can be aware of the effect of his/her performance to immediately so as to increase the amusement in a karaoke singing.

A humming transcription system and methodology is capable of transcribing an input humming signal into a standard notational representation. The disclosed humming transcription technique uses a statistical music recognition approach to recognize an input humming signal, model the humming signal into musical notes, and decide the pitch of each music note in the humming signal. The humming transcription system includes an input means accepting a humming signal, a humming database recording a sequence of humming data for training note models and pitch models, and a statistical humming transcription block that transcribes the input humming signal into musical notations in which the note symbols in the humming signal is segmented by phone-level Hidden Markov Models (HMMs) and the pitch value of each note symbol is modeled by Gaussian Mixture Models (GMMs), and thereby output a musical query sequence for music retrieval in later music search steps.

There is provided a singing synthesis parameter data estimation system that automatically estimates singing synthesis parameter data for automatically synthesizing a human-like singing voice from an audio signal of input singing voice. A pitch parameter estimating section 9 estimates a pitch parameter, by which the pitch feature of an audio signal of synthesized singing voice is got closer to the pitch feature of the audio signal of input singing voice based on at least both of the pitch feature and lyric data with specified syllable bondaries of the audio signal of input singing voice. A dynamics parameter estimating section 11 converts the dynamics feature of the audio signal of input singing voice to a relative value with respect to the dynamics feature of the audio signal of synthesized singing voice, and estimates a dynamics parameter, by which the dynamics feature of the audio signal of synthesized singing voice is got close to the dynamics feature of the audio signal of input singing voice that has been converted to the relative value.

A music retrieval system which take an input melody as the query. In one embodiment, changes or differences in the distribution of energy across the frequency spectrum over time are used to find breakpoints in the input melody in order to separate it into distinct notes. In another embodiment the breakpoints are identified based on changes in pitch over time. A confidence level is preferably associated with each breakpoint and/or note extracted from the input melody. The confidence level is based on one or more of: changes in pitch, absolute values of a spectral energy distribution indicator, relative values of the spectral energy distribution indicator, and the energy level of the input melody. The process of matching the input melody with songs in the music database is based on minimizing a cost computation that takes into account errors in the insertion and deletion of notes, and penalizes these errors in accordance with the confidence levels of the breakpoints and/or notes.

A technique of enabling a singer to auditorily recognize how to change the way of singing is provided. The CPU (11) of a karaoke device (1) associates model voice data stored in a model voice data storage area (14c) with inputted learner voice data in a time axis direction. Then the CPU (11) shifts the pitch of the learner voice data so that it may coincide with the corresponding pitch of the model voice data according to the result of the association, compresses or extends the section (mora) of the learner voice data in the time axis direction so that the section length of the learner voice data coincides with the corresponding section length of the model voice data, and outputs the resultant learner voice data to a voice processing section (18). The voice processing section (18) converts the learner voice data supplied from the CPU (11) into an analog signal and generates the sound from a loudspeaker (19).

A method of converting performance information is carried out by the steps of receiving identification information that identifies an target tone generator different from an available tone generator, reading out first performance information that indicates a music performance in the form of a sequence of tones, and that includes timbre information specifying a timbre of the tones, and changing the timbre information included in the read first performance information based on the received identification information so as to generate second performance information including the changed timbre information adapted to the target tone generator. By such a manner, the available tone generator can process the second performance information to generate the sequence of tones having a timbre as if generated by the target tone generator.

A bandwidth extension module, and an associated method and computer-readable medium, suitable for use in artificially extending the bandwidth of a lowband speech signal. The bandwidth extension module comprises a band-pass filter configured to produce a band-pass signal from the lowband speech signal; at least one carrier frequency modulator, each carrier frequency modulator configured to pitch-synchronously modulate the band-pass signal about a respective carrier frequency, the at least one carrier frequency modulator collectively producing a highband speech signal component; a synthesis filter configured to determine a highband speech signal based on the highband speech signal component; and a summation module configured to combine the lowband speech signal with the highband speech signal to obtain a bandwidth-extended speech signal.

The present invention relates to a gain quantization method and device for implementation in a technique for coding a sampled sound signal processed, during coding, by successive frames of L samples, wherein each frame is divided into a number of subframes and each subframe comprises a number N of samples, where N<L. In the gain quantization method and device, an initial pitch gain is calculated based on a number f of subframes, a portion of a gain quantization codebook is selected in relation to the initial pitch gain, and pitch and fixed-codebook gains are jointly quantized. This joint quantization of the pitch and fixed-codebook gains comprises, for the number f of subframes, searching the gain quantization codebook in relation to a search criterion. The codebook search is restricted to the selected portion of the gain quantization codebook and an index of the selected portion of the gain quantization codebook best meeting the search criterion is found.

Performance data and lyric data stored in an external storage device 35 is read out in accordance with the progression of a song. The performance data is sent to a tone generator 36, so that melody and accompaniment tones are reproduced by use of the performance data. The lyric data is sent to a display control circuit 14, so that lyrics represented by the lyric data are displayed on a display unit 12. The timing at which the lyrics are reproduced is determined by relative time data ΔT added to the lyric data, so that the lyrics are displayed at right positions corresponding to bar and beat positions. When a bar has lengthy lyrics, the width of the bar to be displayed on the display unit 12 is adjusted. As a result, the position of lyrics displayed corresponds to the progression of a song, resulting in users being capable of obtaining timing in both performing a musical instrument and singing when they sing a song while performing the musical instrument.

Described are methods, systems, and apparatuses, including computer program products, that provide a practice mode for multiple musical parts in a rhythm-action game. In one aspect this is accomplished by displaying, on a display in communication with a game platform, a first and second target musical data associated with a musical composition. The game platform receives a selection by the user of the first target musical data to be performed and produces an audio output associated with the first and second target musical data. The game platform also produces a synthesized tone associated with the first target musical data. In some versions, the target music data that is not selected is dimmed and made less visible.

A language learning system and method with a visualized pronunciation suggestion is disclosed. A sound wave corresponding to a sample voice output is used to suggest the user to make corrections with respect to tones and stresses. Through sectional evaluations and corrections, the invention can effectively improve the speaking ability of the user. The disclosed system includes a language database, a follow-reading module, a display control module, a sectional evaluation module, a correction module, and a suggestion-editing module. The disclosed method includes the steps of: extracting a full-sentence sample, outputting a full-sentence voice message and an associated message, prompting the user to imitate and recording the voice data, comparing the voice data with the full-sentence voice message and outputting the similarity, and evaluating the voice data.