1.

Embodied music system

Abstract: An embodied music system. The system creates an interactive interface between a listener and the external environment. The system includes a physical device located in the environment that provides sensory input to the listener. An audio signal of the system is adapted to be heard by the listener. An encoder embeds inaudible control data into the audio signal. A decoder extracts the control data from the audio signal and transmits the control data to the physical device, thereby controlling operation of the device. Finally, an audio reproduction device is connected to the decoder and plays the audio signal for the listener. The embodied music system allows the listener to experience multi-sensory compositions. Application Data Inventors: Pietrusko; Robert Gerard (Brooklyn, NY), Nataraj; Chandrasekhar (Radnor, PA) Assignee: Villanova University (Villanova, PA) Appl. No.: 13/009,185 Filed: January 19, 2011

TECHNICAL FIELD The present invention relates generally to the field of music interface and, more particularly, to a method of expanding the listening experience of a music composition.

CLAIMS An embodied music system, creating an interactive interface between a listener and the external environment, the system comprising: a physical device located in the environment and providing sensory input to the listener; an audio signal adapted to be heard by the listener; an encoder embedding control data inaudible to the listener into the audio signal using high frequency temporal coding, adding to the audio signal a high-frequency carrier signal, the encoder including (a) a frequency-domain filter adapted to clear all frequencies of a fast Fourier Transform analysis of the audio signal above about 15 kHz and create a filtered file; (b) a summation device adapted to sum the filtered file, after the filtered file has been returned to a time-domain signal, with the control data as embedded into the carrier signal to create a sound file; and (c) means for recording or storing the sound file; a decoder extracting the control data from the sound file and transmitting the control data to the physical device, thereby controlling operation of the device; and an audio reproduction device connected to the decoder and playing the audio signal for the listener, whereby the listener experiences multi-sensory compositions.

References: 5209695 May 1993 Rothschild 6400826 June 2002 Chen et al. 6650761 November 2003 Rodriguez et al. 6947893 September 2005 Iwaki et al. 6972363 December 2005 Georges et al. 7237198 June 2007 Chaney 7685324 March 2010 Fukui et al. 2003/0112260 June 2003 Gouzu 2006/0111910 May 2006 Nelson 2007/0250212 October 2007 Halloran et al.

FIG. 13 provides an overview of the encoding process for a scent application of the embodied music system; and FIG. 14 provides a simple overview of an embodied music scent device.

2.

Sound source tracking system, method and robot

Abstract A result of a sound source direction measurement based on an output of an REMA (first microphone array) (11) and a result of a sound source position measurement based on an output of an IRMA (second microphone array) (12) are integrated through a particle filter or in space. Thus, the different microphones, i.e., the REMA (11) and the IRMA (12) can cancel mutual defects or ambiguities with each other. Therefore, from views of improvement in accuracy and robustness a performance of sound source localization can be improved. Application Data Inventors: Nakadai; Kazuhiro (Wako, JP), Tsujino; Hiroshi (Wako, JP), Hasegawa; Yuji (Wako, JP), Okuno; Hiroshi (Kyoto, JP) Assignee: Honda Motor Co., Ltd. (Tokyo, JP) Appl. No.: 12/293,358 Filed: May 9, 2007 PCT Filed: May 09, 2007 PCT No.: PCT/JP2007/059599 371(c)(1),(2),(4) Date: September 17, 2008 PCT Pub. No.: WO2007/129731 PCT Pub. Date: November 15, 2007

Technical Field The present invention relates to a sound source tracking system for measuring in real time the position of a sound source. The present invention offers a solution by providing a sound source tracking system, which aims to improve performance in tracking of sound source localization from the viewpoint of robustness and accuracy. Claims A sound source tracking system for measuring in real time the position of a sound source, comprising: a first processor configured to measure a position or direction of said sound source according to a beam forming method based on an output from a first microphone array, the first microphone array built into a moving body having a movement function, and a posture of said first microphone array; a second processor configured to measure the position of said sound source based on an output from a second microphone array, the second microphone array fixedly arranged and separated from the moving body; and a third processor configured to measure the position of said sound source by integrating respective measurement results from said first and second processors utilizing a particle filter.

References: 2002/0181723 December 2002 Kataoka 2006/0245601 November 2006 Michaud et al. 2009/0030552 January 2009 Nakadai et al. 2002-366191 Dec., 2002 JP 2003-251583 Sep., 2003 JP 2004-198656 Jul., 2004 JP 2006-121709 May., 2006 JP

FIG. 1 is a schematic structural view of a sound source tracking system

3.

Tone detector and method used in a robot for detecting a tone

Abstract A tone detection device for detecting whether an input signal having a tone. The device includes a volume gain calculation unit performing a volume gain treatment on frame data and outputting the volume-gain-treated frame data and energy in time domain thereof, a threshold calculation unit calculating a threshold value on the basis of the energy of the volume-gain-treated frame data, a filter transforming the volume-gain-treated frame data by an algorithm and outputting a characteristic value in a first period, and a comparator comparing the first characteristic value and the threshold value to generate a result and determining the frame data of the input signal has the tone on the basis of the result. Application Data Inventors: Chen; Yao-Sheng (Hsin Chu County, TW), Chung; Min-Wei (Taipei County, TW), Chou; Sung-Tsun (Hsin Chu County, TW) Assignee: SONIX Technology Co., Ltd. (Chupei, Hsinchu, TW) Appl. No.: 12/483,896 Filed: June 12, 2009

Technical Field The invention relates to a tone detection device and a method thereof suitable to be used in a robot, particularly to a tone detection device and a method to acquire a command in a tone.

Claims A tone detection device for detecting whether an input signal having at least a frame data has a tone, the device comprising: a volume gain calculation unit, performing a volume gain treatment on the frame data and outputting the volume-gain-treated frame data and the energy in time domain of the volume-gain-treated frame data, wherein the volume gain treatment calculates energy in time domain of the frame data and adjusts the magnitude of the frame data based on the energy of the frame data; a threshold calculation unit, calculating a threshold value based on the energy of the volume-gain-treated frame data; a filter, transforming the volume-gain-treated frame data by an algorithm and outputting a first characteristic value in a first period; and a comparator, comparing the first characteristic value with the threshold value to generate a comparison result and determining that the frame data of the input signal has the tone based on the comparison result.

References 4689760 August 1987 Lee et al. 5412692 May 1995 Uchida 6381330 April 2002 Johanson 6636609 October 2003 Ha et al. 6671252 December 2003 Cannon et al. 6763106 July 2004 Li et al. 6826404 November 2004 Delfs et al. 6950511 September 2005 Das et al. 7245637 July 2007 Lam et al. 2003/0235291 December 2003 Sauvage et al. 2005/0030941 February 2005 Lee et al. 2006/0152463 July 2006 Furihata et al. 2009/0265173 October 2009 Madhavan et al.

FIG. 1 shows a flow chart illustrating the tone detection method according to one embodiment of the invention.

4.

Voice recognition apparatus and method for performing voice recognition comprising calculating a recommended distance range between a user and an audio input module based on the S/N ratio
Abstract A voice recognition apparatus includes: a voice recognition module that performs a voice recognition for an audio signal during a voice period; a distance measurement module that measures a current distance between the user and an voice input module; a calculation module that calculates a recommended distance range, in which being estimated that an S/N ratio exceeds a first threshold, based on the voice characteristic; and a display module that displays the recommended distance range and the current distance. Application Data Inventors: Sugiyama; Hiroshi (Kawasaki, JP), Suzuki; Kaoru (Yokohama, JP), Yamamoto; Daisuke (Kawasaki, JP), Koga; Toshiyuki (Kawasaki, JP) Assignee: Kabushiki Kaisha Toshiba (Tokyo, JP) Appl. No.: 12/370,133 Filed: February 12, 2009 Technical Field The present invention relates to a voice recognition apparatus and a method for performing a voice recognition. The conventional methods employ a method in which the user's voice level is adjusted through multiple voices made by the user. Hence, those methods require the user to perform the voice again every time the adjustment is required due to the change in environment such as the change in the surrounding noise level. Claims A voice recognition apparatus comprising: an audio input module that receives an audio input and outputs an audio signal, the audio input module having a gain being configured to be adjustable; a voice recognition module that detects a voice period, where a voice activity by a user is detected in the audio signal and performs a voice recognition for the audio signal during the voice period; a first level measurement module that measures a voice level of the audio signal in the voice period and outputs the voice level; a second level measurement module that measures a noise level of the audio signal in a noise period and outputs the noise level, the noise period being a time period except the voice period; a first calculation module that calculates an S/N ratio that is a ratio of the voice level to the noise level; a distance measurement module that measures a current distance between the user and the audio input module; a first memory module that stores a first threshold corresponding to an S/N ratio at which the voice recognition module is capable to perform the voice recognition with a given recognition rate; a second memory module that stores a voice characteristic having a set of the voice level, the current distance and the gain, by which the voice

recognition module has succeeded in performing the voice recognition; a second calculation module that calculates a recommended distance range based on the voice characteristic, the recommended distance range indicating a distance range of the user and the audio input module at which the S/N ratio exceeds the first threshold; and a display module that displays the recommended distance range and the current distance. References 7031917 April 2006 Asano 2006/0143017 June 2006 Sonoura et al. 2006/0195598 August 2006 Fujita et al. 2008/0312918 December 2008 Kim 1370387 Sep., 2002 CN 1202603 May., 2002 EP 06-236196 Aug., 1994 JP 2877350 Jan., 1999 JP 2000-089787 Mar., 2000 JP 2004-226656 Aug., 2004 JP 2004230480 Aug., 2004 JP 2006-181651 Jul., 2006 JP 2006-227499 Aug., 2006 JP 2005/013263 Feb., 2005 WO

FIG. 6 is a view showing the correlation between S/N ratio and voice recognition rate.

5.

Voice input-output device and communication device

Abstract A voice input-output device includes a voice input section and a voice output section. The voice input section includes a microphone unit, the microphone unit including a housing that has an inner space, a partition member that is provided in the housing and divides the inner space into a first space and a second space, the partition member being at least partially formed of a diaphragm, and an electrical signal output circuit that outputs an electrical signal that is the first voice signal based on vibrations of the diaphragm, a first through-hole through which the first space communicates with an outer space of the housing and a second through-hole through which the second space communicates with the outer space being formed in the housing. The voice output section includes: an ambient noise detection section that detects ambient noise during a call based on the first voice signal; and a volume control section that controls volume of the speaker based on a degree of the detected ambient noise. Application Data Inventors: Takano; Rikuo (Suginami, JP), Sugiyama; Kiyoshi (Mitaka, JP), Fukuoka; Toshimi (Yokohama, JP), Ono; Masatoshi (Tsukuba, JP), Horibe; Ryusuke (Hirakata, JP), Tanaka; Fuminori (Suita, JP), Choji; Hideki (Muko, JP), Inoda; Takeshi (Kyoto, JP) Assignee: Funai Electric Advanced Applied Technology Research Institute Inc. (JP) Funai Electric Co., Ltd. (JP) Appl. No.: 12/144,284 Filed: June 23, 2008 Technical Field The present invention relates to a voice input-output device and a communication device. It is desirable to pick up only desired sound (user's voice) during a telephone call, speech recognition, voice recording, or the like. However, sound (e.g., background noise) other than desired sound may also be present in an environment in which a voice input device is used. Therefore, a voice input device has been developed which has a function of removing noise. Claims A hands-free voice input-output device comprising: a hands-free voice input section that generates a first voice signal; and a voice output section that outputs a voice from a speaker based on a second voice signal, the hands-free voice input section including a microphone unit, the microphone unit including a housing that has an inner space, a partition member that is provided in the housing and divides the inner space into a first space and a second space, the partition member being at least partially formed of a diaphragm which has a first face and a second face, the first face faces the first space, and the second face faces the second space, and an electrical signal output circuit that outputs an electrical signal that is

the first voice signal based on vibrations of the diaphragm, a first through-hole formed in the housing through which the first space communicates with a space outside of the housing and a second through-hole formed in the housing through which the second space communicates with the space outside the housing, wherein sound pressures applied to the first face and the second face correspond to sound pressures of sounds which have entered the first through-hole and the second through-hole respectively. References 3842205 October 1974 Okamoto 5511132 April 1996 Yoshimi 5574437 November 1996 Schwinn et al. 5862234 January 1999 Todter et al. 6819938 November 2004 Sahota 6920230 July 2005 Usuki et al. 7092744 August 2006 Rodemer et al. 7187956 March 2007 Sugino et al. 7280958 October 2007 Pavlov et al. 7283850 October 2007 Granovetter et al. 7447630 November 2008 Liu et al. 7512425 March 2009 Fukuda 2002/0048373 April 2002 Paritsky et al. 2002/0137478 September 2002 Masamura

FIG. 7 is a flowchart showing a process of producing a microphone unit.