Texas Instruments Corporation

Kabushiki Kaisha Sega

Ryo Tsutsui

Steven David Trautmann

Yoshihide Iwata

Atsuhiro Sakurai

Akihiro Yonemoto

Hiroyuki Uchida

Keitaro Hashimoto

Ivan Setiawan

Methods and systems for amplitude modulation in a parametric speaker system are provided that perform truncated double sideband (TDSB) frequency modulation of audio signal in which most of the processing is performed in the frequency domain, thus permitting use of fast processing techniques for amplitude modulation (AM) and filtering and reducing computation cost over time domain processing. A maximum envelope value of the time domain audio signal may be to the carrier signal in the frequency domain that avoids emitting the carrier signal when the input signal level is low or mute. The application of the envelope value may be smoothed to reduce discontinuity at input block boundaries.

Modulation of audio signals in a parametric speaker

Methods for sample rate conversion are provided that use a polynomial interpolator with minimax stopband attenuation. A method for sample rate conversion of an input signal is provided that uses a time-varying polyphase filter having a discrete polyphase index m. Another method for sample rate conversion of an input signal is provided that uses a time-varying polyphase filter having a continuous polyphase index &#x3c4;. In these methods, an output time index is mapped to an input sample index and the polyphase index, the polynomial coefficients of a polyphase filter are computed using the polyphase index, and the polyphase filter is applied to an input sample at the input sample index to generate the output sample at the output time index.

Asynchronous sample rate conversion using a polynomial interpolator with minimax stopband attenuation

Methods, digital systems, and computer readable media are provided for estimating change of amplitude and frequency in a digital audio signal by transforming a frame of the digital audio signal to the frequency domain, locating a frequency peak in the transformed frame, determining an interpolated peak of the located frequency peak, computing inner products of a portion of the transformed frame about the interpolated peak with a plurality of test signals, and estimating change of amplitude and change of frequency for the frequency peak from results of the inner products.

Method and system for estimating frequency and amplitude change of spectral peaks

A game system for executing a competition game in which a plurality of players compete. The game system comprises a competition score storing unit; a handicap information storing unit; a matching control unit; a rank deciding unit; a handicap selecting unit; a handicap deciding unit; and a handicap setting unit. The game system allows players of different levels to enjoy a game together.

Game system and game control method

Rebalancing of an audio signal refers to achieving a balance of perceived loudness, typically of right and left channels, given an unbalanced input. A flexible method to automatically rebalance an audio input signal is robust against noise in extreme cases through the individual channels combined in various ways as a function of the loudness ratio between input channels.

Rebalancing of audio

Bandwidth expansion for audio signals by frequency band translations plus adaptive gains to create higher frequencies; use of a common channel for both stereo channels limits computational complexity. Adaptive cut-off frequency determination by power spectrum curve analysis, and bass expansion by both fundamental frequency illusion and equalization.

Audio bandwidth expansion

A method for creating a stereophonic sound image out of a monaural signal combines two sub-methods. Comb filters decorrelate the left and right channel signals. Intra-aural difference cues, such as an Intra-Aural Time Difference (ITD) and an Intra-aural Intensity Difference (IID) separated channels. Strict complementary (SC) linear phase FIR filters divide the incoming monaural signal into three frequency band separation. The comb filters and ITD/IID applied to the low and high frequency bands create a simulated stereo sound image for the instruments other than human voice. Listening tests indicate that this invention provides a wider stereo sound image than previous methods, while retaining human voice centralization. Since the comb filter solution and ITD/IID solution can share the same filter bank, the computational cost of this method is almost the same as the previous method.

Stereo synthesizer using comb filters and intra-aural differences

A method of converting single channel audio (mono) signals to two channel audio (stereo) signals using simple filters and an Intra-aural Time Difference (ITD) is presented. This method does not distort the spectral content of the original signal very much, and has low computation requirements. A variation is proposed which also uses Intra-aural Intensity Difference (IID).

Low computation mono to stereo conversion using intra-aural differences

A new method is proposed that produces stereophonic sound image out of monaural signal within a selected frequency regions. The system employs a strictly complementary (SC) linear phase FIR filter pair that separates input signal into different frequency regions. A pair of comb filters is applied to one of the filter's output. This implementation allows a certain frequency range to be relatively localized at center while the other sounds are perceived in a wider space.

Band-selectable stereo synthesizer using strictly complementary filter pair

Graphic equalizer as a cascade of equalization filters with the gain of each individual filter compensated for the gain leakage from other filters. A linear approximation allows individual filter gains to be set to give desired frequency responses.

Growing community of inventors

Ibaraki, Japan

Ryo Tsutsui