Sony Corporation

Sony Group Corporation

Takao Fukui

Ayataka Nishio

Toru Chinen

Mitsuyuki Hatanaka

Minoru Tsuji

Kazutoshi Nomoto

Yoshihiro Murakami

Naoya Takahashi

Michimasa Sekiguchi

The present technology relates to a signal processing device, a method, and a program that can obtain a signal with higher sound quality. The signal processing device includes: a calculation unit that calculates a parameter for generating a difference signal corresponding to an input compressed sound source signal on the basis of a prediction coefficient and the input compressed sound source signal, the prediction coefficient being obtained by learning using, as training data, a difference signal between an original sound signal and a learning compressed sound source signal obtained by compressing and coding the original sound signal; a difference signal generation unit that generates the difference signal on the basis of the parameter and the input compressed sound source signal; and a synthesis unit that synthesizes the generated difference signal and the input compressed sound source signal. The present technology can be applied to a signal processing device.

Signal processing device, method, and program

The present technology relates to a signal processing apparatus, a signal processing method, and a program that are to enable acquisition of a signal with higher sound quality. A signal processing apparatus includes: a difference-signal generation unit configured to generate, on the basis of an input signal and a prediction coefficient that is acquired by learning with, as training data, a difference signal based on a re-quantized signal for learning acquired by re-quantization of an original sound signal and the original sound signal, the difference signal corresponding to the input signal; and a combining unit configured to combine the difference signal generated and the input signal. The present technology is applicable to a signal processing apparatus.

Signal processing apparatus, signal processing method, and program

The present technology relates to a signal processing device, a signal processing method, and a program that enable implementation of more effective distance feeling control. The signal processing device includes a reverb processing unit that generates a signal of a reverb component on the basis of object audio data of an audio object and a reverb parameter for the audio object. The present technology can be applied to a signal processing device.

Signal processing device, signal processing method, and program

A signal processing apparatus is provided that includes a sound source separation section configured to apply sound source separation processing to a mixed sound signal including a mixture of signals of a plurality of sound sources, and band extension sections configured to apply frequency band extension processing to respective sound source separation signals obtained by separation by the sound source separation section.

The present technology relates to a signal processing apparatus, a signal processing method, and a program that make it possible to cope with an output of a PCM signal using one DSD signal. A distribution apparatus includes an extraction section that, in a case where a PCM signal having a predetermined sampling frequency is generated from a DSD signal, extracts a predetermined number of samples from the DSD signal around samples at a predetermined interval determined by the predetermined sampling frequency, and a filtering section that generates the PCM signal having the predetermined sampling frequency by filtering the extracted predetermined number of samples. The present technology is applicable to, for example, a distribution apparatus, etc., that distributes the PCM signal to a client apparatus.

Signal processing apparatus and signal processing method

The present technology relates to a signal processing apparatus, a signal processing method, and a program that permit switching between a plurality of DSD signals having different sampling frequencies using a simple configuration. An acquisition section acquires a digital audio signal having a given sampling frequency selected from among the plurality of digital audio signals acquired by delta-sigma modulating an audio signal at a plurality of sampling frequencies. An interpolation section subjects the acquired digital audio signal to a pre-interpolation process when the sampling frequency of the acquired digital audio signal is lower than an operating clock of a delta-sigma demodulator. The present technology is applicable, for example, to a signal processing apparatus.

The present disclosure relates to a compression/encoding apparatus and method, a decoding apparatus and method, and a program that allow for provision of a lossless compression technology with higher compression ratio. A GOB data configuration section configures GOB data with a group of digital data that includes a plurality of blocks by treating a frame of delta-sigma-modulated digital data as a block. A table generation section generates a conversion table for encoding the GOB data. An encoding section compresses and encodes the digital data of each block included in the GOB data by using the conversion table. The present technology is applicable, for example, to audio signal compression and encoding, and so on.

Compression/encoding apparatus and method, decoding apparatus and method, and program

The present disclosure relates to a compressive encoding apparatus, a compressive encoding method, a decoding apparatus, a decoding method, and a program which can provide a lossless compression technology having a higher compression rate. An encoding unit of the compressive encoding apparatus converts M bits of a &#x394;&#x3a3;-modulated digital signal into N bits (M&#x3e;N) with reference to a first conversion table, and when the M bits are not able to be converted into the N bits with the first conversion table, converts the M bits into the N bits with reference to a second conversion table. When the number of bit patterns of the N bits is P, the first conversion table is a table storing (P&#x2212;1) number of codes having higher generation frequencies for past bit patterns, and the second conversion table is a table storing (P&#x2212;1) number of codes having higher generation frequencies for past bit patterns, which follow those of the first conversion table. The present disclosure is applicable to a compressive encoding apparatus that compressively encoding an audio signal, and the like, for example.

Compressive encoding apparatus, compressive encoding method, decoding apparatus, decoding method, and program

Provided is an audio signal processing device including a signal processing section that changes, at a time of generating and outputting 2-channel audio signals to be subjected to sound reproduction by two electroacoustic transducing means located at positions in the vicinities of both ears of a listener, from audio signals of a plurality of and more than two channels, virtual sound image localization positions on a circle around the listener, across the virtual sound image localization positions, the virtual sound image localization position that is supposed for each of the plurality of channels of audio signals provided on the circle.

Audio signal processing device, audio signal processing method, and computer program

A head-related transfer function (HRTF) convolution method arranged, when an audio signal is reproduced acoustically by an electro-acoustic conversion unit disposed in a nearby position of both ears of a listener, to convolute an HRTF into the audio signal, which allows the listener to listen to the audio signal such that a sound image is localized in a perceived virtual sound image localization position, the method including the steps of: measuring, when a sound source is disposed in the virtual sound image localization position, and a sound-collecting unit is disposed in the position of the electro-acoustic conversion unit, a direct-wave direction HRTF regarding the direction of a direct wave, and reflected-wave direction HRTFs regarding the directions of selected one or more reflected waves, from the sound source to the sound-collecting unit, separately beforehand; and convoluting the obtained direct-wave direction HRTF, and the reflected-wave direction HRTFs into the audio signal.

Head-related transfer function convolution method and head-related transfer function convolution device

Provided is a signal processing apparatus, including a filter unit that filters an audio signal created by decimating a portion of frequency components by an all-pass filter and outputs a filtering result thereof as improvement components to improve sound quality of the audio signal, and an adder that generates an improved sound in which the sound quality of the audio signal is improved by adding the improvement components to the audio signal.

A head related transfer function generation apparatus includes a first input unit that inputs a first head related transfer function generated in a first measurement environment, a second input unit that inputs a second head related transfer function generated in a second measurement environment, and a transform normalization processing unit that normalizes a first gain of the first head related transfer function represented in frequency-axis data with a second gain of the second head related transfer function represented in frequency-axis data and also calculates a square root thereof.

Head related transfer function generation apparatus, head related transfer function generation method, and sound signal processing apparatus

An audio signal processing device includes: head related transfer function convolution processing units convoluting head related transfer functions with audio signals of respective channels of plural channels, which allow the listener to listen to sound so that sound images are localized at assumed virtual sound image localization positions concerning respective channels of the plural channels of two or more channels when sound is reproduced by electro-acoustic transducer means; and 2-channel signal generation means for generating 2-channel audio signals to be supplied to the electro-acoustic transducer means from audio signals of plural channels from the head related transfer function convolution processing units, wherein, in the head related transfer function convolution processing units, at least a head related transfer function concerning direct waves from the assumed virtual image localization positions concerning a left channel and a right channel in the plural channels to both ears of the listener is not convoluted.

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Tokyo, Japan

Takao Fukui