Matsushita Electric Industrial Co., Ltd.

That Corporation

Mikio Oda

Akihisa Kawamura

Masaharu Matsumoto

Hiroko Numazu

Mitsuhiko Serikawa

Ryou Tagami

Takeshi Norimatsu

Tomoe Kawane

Takeo Kanamori

Takashi Kawamura

Matthew Easley

Satoru Ibaraki

Tetsuhiko Kaneaki

Yasunori Tani

Kozo Nuriya

Ryo Tagami

Audio processing architectures are described for implementing a multi-band audio compression algorithm with advanced surround processing. Exemplary architectures can improve the fidelity and perceived sound field spread of inexpensive, cabinet mounted, stereo speakers such as those that might be found in televisions, wireless speaker systems and sound bars. Embodiments of the present disclosure can improve inexpensive, cabinet mounted, stereo speakers by providing, e.g., (i) an Advanced Surround algorithm that adds depth and height to the left/right/center sound field images, (ii) a Soft Clip algorithm to minimize the perceived artifacts caused by compressor overshoot, (iii) configurable crossover filter order adjustment to allow better isolation between bands, (iv) a compressor maximum gain adjustment to reduce overshoot and minimize noise boost, and/or (v) a center gain adjustment to emphasize the perception of the center image (dialog) in high ambient noise situations.

Configurable multi-band compressor architecture with advanced surround processing

In a voice pitch normalization device equipped in a voice recognition device VRAp for recognizing an incoming command voice Sva uttered by any speaker, and used to normalize the incoming command voice to be in an optimal pitch for voice recognition, a target voice generator produces a target voice signal by changing the incoming command voice Svd on the basis of a predetermined degree. A probability calculator calculates a probability indicating a degree of coincidence among the target voice signal and a plurality of words in sample data. A voice pitch changer repeatedly changes the target voice signal in voice pitch until a maximum probability becomes a predetermined probability or greater.

Device for normalizing voice pitch for voice recognition

A low-cost speech recognition apparatus for AV equipment capable of speech recognition with high accuracy while 2-channel sound is being produced from loudspeakers is achieved. A monaural conversion part converts 2-channel signals to be inputted to the loudspeakers into a monaural signal. A single echo canceller is provided with an output from a microphone and an output from the monaural conversion part (monaural signal). The echo canceller estimates an echo of multichannel sound based on the monaural signal, and then eliminates the echo sound from the microphone output. Thus, with only a single echo canceller, speech recognition can be carried out while 2-channel sound is being produced from the loudspeakers. Moreover, unlike the case where two echo cancellers are provided, the present invention can prevent the occurrence of mutual interference between the echo cancellers that leads to deterioration in speech recognition performance.

Speech recognition apparatus for AV equipment

A low frequency audio conversion circuit for converting the frequency of low frequency audio components. An input audio signal includes a low frequency audio component lower than the frequency a speaker can reproduce. The low frequency audio component is filtered and extracted by a low pass filter and full-wave rectified to generate even-numbered harmonics of the low frequency audio component. Secondary harmonics are extracted from the even-numbered harmonics and added to the input audio signal after being amplified to an appropriate level. When a speaker whose low frequency sound reproduction characteristics are poor is used, and a low frequency component lower than the frequency the speaker can reproduce is supplied, the low frequency audio component is reproduced as secondary harmonics which fall within the frequency range of the speaker. Thus, the low frequency audio component is compensated, and a powerful sound is reproduced at a low cost without degrading the sound.

Low frequency audio conversion circuit

The apparatus of the invention calculates filter coefficients for controlling sound field and sound image, based on a plurality of first impulse response signals and a pair of second impulse response signals. The plurality of first impulse response signals indicate impulse responses from loudspeakers reproducing audio signals to both ears of a listener. The pair of second impulse response signals indicate impulse responses from a reference loudspeaker at a position at which a sound image is localized to both ears of the listener. The apparatus includes: a feature extracting section for receiving the pair of second impulse response signals, for extracting parameters representing features of the pair of second impulse response signals, and for outputting parameter signals; a signal adjusting section for adjusting at least one of the plurality of first impulse response signals based on the parameter signals, and for outputting a pair of third impulse response signals having the same features as the extracted features; and a coefficient calculating section for calculating the filter coefficients for controlling the sound field and sound image, based on the plurality of first impulse response signals and the pair of third impulse response signals applied from the signal adjusting means.

Sound field and sound image control apparatus and method

A sound reproduction system for automatically conducting a tone control for an audio signal according to the present invention includes: an input terminal for inputting an audio signal; a judgment circuit for determining a ratio between a stereophonic signal component and a monophonic signal component of the input audio signal and for outputting a judgment signal; a signal processing section for receiving the input audio signal and processing the input audio signal so as to generate an output signal having uniform transmission characteristics irrespective of the listening position; an adder for receiving the input audio signal and the output signal and for adding the input audio signal with the output signal at a certain addition ratio based on the judgment signal so as to generate an added signal; and a loudspeaker for receiving the added signal and for reproducing the added signal in a plurality of positions.

Sound reproduction system and a sound reproduction method

According the present invention, a sound field controller for reproducing a sound field with presence comprising; input terminals for inputting an audio signal having a first and a second channel signals, a signal extracting circuit for receiving and processing the audio signal, and producing an extracted signal of the audio signal, an operation circuit for receiving the extracted signal from the signal extracting circuit, performing the convolution on the extracted signal, and generating a convolution sum signal, a delay circuit for delaying the convolution sum signal by a predetermined time, and producing a delayed signal, an adding circuit for receiving the audio signal and the delayed signal, and adding the audio signal and the delayed signal with a predetermined summation ratio to produce a summed signal, and output terminals for reproducing the summed signal to localize a sound image in a desirable direction.

Sound field controller

A sound field controller for generating apparent sound sources by adjusting the amplitude and delay time of a sound signal so that the sound will be perceived by plural listeners as sound coming from a location separated from the specific location of the front speakers, and for additionally controlling the effect of the apparent sound sources by evaluating the attributes of the source sound signal. The controller includes FIR filters for generating a left sound pattern signal, FIR filters for generating a right sound pattern signal, a first delay circuit for delaying the left and right sound pattern signals by a first predetermined time and applying the delayed left and right sound pattern signals to the left and right speakers, respectively, to introduce an apparent sound source located left rear of a center listener; and a second delay circuit for delaying the left and right sound pattern signals by a second predetermined time and applying the delayed left and right sound pattern signals to the right and left speakers, respectively, to introduce an apparent sound source located right rear of a center listener.

A music tone pitch shift apparatus which converts an original audio signal into digital data by way of pulse code modulation (PCM), shifting the pitch, and converting the pitch shifted digital data into an analog signal. The PCM digital data is stored in a ring memory at a given sampling speed, and is read out of the memory by a pair of identical read circuits at a common read addressing speed corresponding to the desired pitch. One of the read circuits starts reading from the opposite address location to the other on the ring memory. Since the read addressing speed is set faster than the write addressing speed when increasing the pitch, and vice versa, overtaking or lapping between the addresses could occur. In switching alternately the read circuits from a now-outputting side to a switching-to side, the read address on the switching-to side circuit is stopped increasing at an address location where a zero-amplitude data has been read, until a zero-amplitude data in phase with that which the switching-to side circuit has read is read by the now-outputting side circuit and the switching is made, immediately before the overtaking or lapping occurs on the now-outputting side circuit. Thus, a smooth connection of the pitch shifted audio signals can be made without including such amplitude modulated components as in the cross fade method, and therefore, a high-quality music tone pitch shift operation can be realized.

Music tone pitch shift apparatus

A first analog input signal is converted into a plurality of second analog signals having different levels respectively. The second analog signals are inputted into respective analog-to-digital converters. Values of output signals from the respective converters are adjusted in accordance with the values of the output signals from the respective converters to convert the output signals from the respective converters into respective adjusted signals. One of the adjusted signals is selected in accordance with the values of the output signals from the respective converters. The selected signal is produced as a system output signal.

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Yawata, Japan

Mikio Oda