Yamaha Corporation

Kazunobu Kondo

Yasuyuki Umeyama

Jordi Bonada

Yu Takahashi

Jordi Janer

Ricard Marxer

Kenji Ishizuka

Kenichi Yamauchi

Toshifumi Kunimoto

Motoichi Tamura

Koji Niimi

Eiko Kobayashi

Shinji Koezuka

Hideki Kemmochi

Kenichi Tamiya

Akitoshi Saito

Francisco Garcia

Makoto Yamada

Yoshifumi Mizuno

An audio processing method obtains observed envelopes of picked-up sound signals including a first observed envelope representing a contour of a first sound signal including a first target sound from a first sound source and a second spill sound from a second sound source and a second observed envelope representing a contour of a second sound signal including a second target sound from the second sound source and a first spill sound from the first sound source; and generates, based on the observed envelopes, output envelopes including a first output envelope representing a contour of the first target sound in the first observed envelope and a second output envelope representing a contour of the second target sound in the second observed envelope, using a mix matrix including a mix proportion of the second spill sound in the first sound signal and a mix proportion of the first spill sound in the second sound signal.

Audio processing method, audio processing system, and computer-readable medium

A method for setting a filter frequency response includes: dividing a target frequency band into multiple first partial bands at a frequency corresponding to an intersection between a first line representative of a target spectrum envelope and a second line representative of a first spectrum envelope derived from smoothing a measured frequency spectrum; and setting a frequency response of a filter corresponding to a first partial band specified by an operator from among the multiple first partial bands, in accordance with a setting operation for setting an adjustment amount performed by the operator with respect to the specified first partial band.

Method and apparatus for setting filter frequency response

A recording method includes acquiring each piece of acoustic data representing a sound from each of a plurality of portable terminal devices. The each of the plurality of portable terminal devices includes a recording unit configured to generate the piece of acoustic data. The recording method also includes executing synchronization processing for synchronizing the respective pieces of acoustic data and executing mixing processing for mixing a plurality of pieces of acoustic data for which the synchronization processing has been executed.

Recording method, recording system, recording program storage medium, acoustic processing method, and acoustic processing device

In a sound processing device, an index value calculation unit calculates a first index value that follows change of a sound signal at a first following degree and a second index value that follows the change of the sound signal at a second following degree which is lower than the first following degree. An adjustment value calculation unit calculates an adjustment value effective to adjust a reverberation component of the sound signal based on difference between the first index value and the second index value. A reverberation adjustment unit applies the adjustment value to the sound signal.

Sound processing device

Candidate frequencies per unit segment of an audio signal are identified. First processing section identifies an estimated train that is a time series of candidate frequencies, each selected for a different one of the segments, arranged over a plurality of the unit segments and that has a high likelihood of corresponding to a time series of fundamental frequencies of a target component. Second processing section identifies a state train of states, each indicative of one of sound-generating and non-sound-generating states of the target component in a different one of the segments, arranged over the unit segments. Frequency information which designates, as a fundamental frequency of the target component, a candidate frequency corresponding to the unit segment in the estimated train is generated for each unit segment corresponding to the sound-generating state. Frequency information indicative of no sound generation is generated for each unit segment corresponding to the non-sound-generating state.

Technique for estimating particular audio component

A signal processing device processes a plurality of observed signals at a plurality of frequencies. The plurality of the observed signals are produced by a plurality of sound receiving devices which receive a mixture of a plurality of sounds. In the signal processing device, a storage stores observed data of the plurality of the observed signals. The observed data represents a time series of magnitude of each frequency in each of the plurality of the observed signals. An index calculator calculates an index value from the observed data for each of the plurality of the frequencies. The index value indicates significance of learning of a separation matrix using the observed data of each frequency. The separation matrix is used for separation of the plurality of the sounds from each other at each frequency. A frequency selector selects one or more frequency according to the index value of each frequency. A learning processor determines the separation matrix by learning with a given initial separation matrix using the observed data of the selected frequency.

In a sound processing apparatus, a likelihood calculation unit calculates an in-region coefficient and an out-of-region coefficient indicating likelihood of generation of each frequency component of a sound signal inside and outside a target localization range, respectively, according to localization of each frequency component. A reverberation analysis unit calculates a reverberation index value according to the ratio of a reverberation component for each frequency component. A coefficient setting unit generates a process coefficient for suppressing or emphasizing a reverberation component generated inside or outside the target localization range, for each frequency component of the sound signal, on the basis of the in-region coefficient, the out-of-region coefficient and the reverberation index value. A signal processing unit applies the process coefficient of each frequency component to each frequency component of the sound signal.

Sound processing apparatus

A coefficient train processing section, which sequentially generates per unit segment a processing coefficient train for suppressing a target component of an audio signal, includes a basic coefficient train generation section and coefficient train processing section. The basic coefficient train generation section generates a basic coefficient train where basic coefficient values corresponding to frequencies within a particular frequency band range are each set at a suppression value that suppresses the audio signal while coefficient values corresponding to frequencies outside the particular frequency band range are each set at a pass value that maintains the audio signal. The coefficient train processing section generates the processing coefficient train, per unit segment, by changing, to the pass value, each of the coefficient values corresponding to frequencies other than the target component among the coefficient values corresponding to the frequencies within the particular frequency band range.

Technique for suppressing particular audio component

Signal processing section of a terminal converts acquired audio signals of a plurality of channels into frequency spectra set, calculates sound image positions corresponding to individual frequency components, and displays, on a display screen, the calculated sound image positions results by use of a coordinate system having coordinate axes of the frequency components and sound image positions. User-designated partial region of the coordinate system is set as a designated region and an amplitude-level adjusting amount is set for the designated region, so that the signal processing section adjusts amplitude levels of frequency components included in the frequency spectra and in the designated region, converts the adjusted frequency components into audio signals and outputs the converted audio signals.

Graphical audio signal control

In an audio processing device, a target sound emphasizer generates a target sound emphasized component by emphasizing a target sound component contained in a plurality of audio signals generated by a plurality of sound receiving devices. A stereo processor generates a stereo component of a plurality of channels from the plurality of audio signals. A first adjuster adjusts a sound pressure level of the target sound emphasized component according to a first adjustment value, and a second adjuster adjusts a sound pressure level of the stereo component according to a second adjustment value. A variable setter variably sets a zoom value which is changeable between a wide angle side and a telephoto side relative to a target. An adjustment controller controls the first adjustment value according to the zoom value such that the sound pressure level of the target sound emphasized component exponentially decreases as the zoom value changes toward the wide-angle side and controls the second adjustment value according to the zoom value such that the sound pressure level of the stereo component increases as the zoom value changes toward the wide-angle side.

Audio processing device

A noise suppressing apparatus suppresses a noise component of a sound signal which contains the noise component and a signal component. In the apparatus, a frequency analyzing section divides the sound signal into a plurality of frames such that adjacent frames overlap with each other along a time axis, and computes a first spectrum of each frame. A noise suppressing section suppresses a noise component of the first spectrum so as to provide a second spectrum of each frame in which the noise component is suppressed. A frequency specifying section specifies a frequency of a noise component of each frame. A phase controlling section varies a phase of the noise component corresponding to the specified frequency in the second spectrum by a different variation amount each frame. A signal synthesizing section combines the frames after the second spectrum of each frame is processed by the phase controlling section, such that adjacent frames overlap with each other along the time axis so as to output the sound signal.

Noise suppressing apparatus and program

A time-axis compression/expansion method and apparatus for multitrack signals is provided, which is capable of performing time-axis compression/expansion on a multitrack signal in such an appropriate manner as to prevent a degradation in the sound quality of a sound generated through a multichannel reproduction or a sound generated through reproduction of a musical tone signal obtained by mix-down. Positions of attacks of the rhythm track sound source signal of a plurality of track sound source signals are detected. Portions of the rhythm track sound source signal between the detected positions of attacks are subjected to a first time-axis compression/expansion process, and the other track sound source signals are subjected to a second time-axis compression/expansion process, based on the detected positions of attacks.

Time-axis compression/expansion method and apparatus for multitrack signals

A time-axis compression/expansion system compresses or expands a multichannel signal at a specified compression/expansion rate. Waveform segments are sequentially cut out from each channel signal. A cutting starting point of a leading end portion of a waveform segment following each preceding waveform segment of the cut out waveform segments is determined commonly between the channel signals, based on two portions of a waveform of a synthesized signal formed by synthesizing the channel signals within a range of predetermined search parameters of the waveform of the synthesized signal. The two portions correspond to a time period over which cross-fading is to be carried out and are most similar to each other. The preceding waveform segment and the following waveform segment cut from each channel signal are spliced together by cross-fading a trailing end portion of the preceding waveform segment and the leading end portion of the following waveform segment.

Time-axis compression/expansion method and apparatus for multichannel signals

A vector quantizer and a vector quantization method are provided, which do not allow the embedding position of the watermark information to be identified easily, and thus enable unfair use of the vector quantized information to be accurately found out. Waveform extraction is performed on input signal data to output extracted waveform data. Index data are generated based on the output extracted waveform data. As index data to be generated based on predetermined extracted waveform data in which no watermark information is to be embedded, out of the output extracted waveform data, index data is randomly selected, which corresponds to the predetermined extracted waveform data from a plurality of index data which, when decoded, have amounts of errors falling within a predetermined range with respect to the predetermined extracted waveform data before decoding.

Vector quantizer and vector quantization method

A time-scale modification method or apparatus is basically designed to effect a time-scale modification process (i.e., expansion or compression with respect to time) on rhythm source signals containing waves such that rhythm sounds are not substantially changed in pitches. Herein, attack positions are detected from the rhythm source signals by using thresholds which are determined in advance. Hence, the time-scale modification process is performed on intermediate signal portions of the rhythm source signals between the attacks in accordance with a desired time-scale modification factor. Then, the intermediate signal portions subjected to the time-scale modification process are smoothly connected with other signal portions such as the attacks and their proximal portions, which are not subjected to the time-scale modification process. Therefore, it is possible to secure the attacks and their proximal portions, which are left without being substantially changed, while accomplishing the time-scale modification on the rhythm source signals. Thus, it is possible to avoid occurrence of double beat and rhythm disorder in rhythm sounds, which are conventionally caused to occur by the time-scale modification.

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Hamamatsu, Japan

Kazunobu Kondo