Abstract: Audio signal processing that satisfies both sound quality and cost is enabled. A processing unit that performs processing based on an output audio signal of a first microphone unit focusing on sound quality and an output audio signal of a second microphone unit focusing on cost is included. For example, the processing performed by the processing unit includes processing of obtaining a beamforming output and processing of obtaining a sound source separation output. Furthermore, for example, the processing performed by the processing unit may include processing of generating a first audio signal on the basis of an output audio signal of the first microphone and processing of generating a second audio signal on the basis of output audio signals of a plurality of the second microphone units.

Abstract: To provide an audio processing device including a configuration in which a different microphone is used for generation of directivity depending on a frequency band.

Abstract: A sound processing device program that enables a sound signal adapted to an intended use to be output is provided. The sound processing device includes a signal processing part that processes a sound signal picked up by a microphone, and generates a recording sound signal to be recorded in a recording device, and generates an amplification sound signal different from the recording sound signal to be output from a speaker. The sound processing device can be applied to, for example, a sound amplification system that performs off-microphone sound amplification.

Abstract: A sound processing device program that enables a sound signal adapted to an intended use to be output is provided. The sound processing device includes a signal processing part that processes a sound signal picked up by a microphone, and generates a recording sound signal to be recorded in a recording device, and generates an amplification sound signal different from the recording sound signal to be output from a speaker. The sound processing device can be applied to, for example, a sound amplification system that performs off-microphone sound amplification.

Abstract: The present technology relates to a sound processing device, a sound processing method, and a program that enable a sound signal adapted to an intended use to be output. A sound signal adapted to an intended use can be output by providing a sound processing device including a signal processing part that processes a sound signal picked up by a microphone, and generates a recording sound signal to be recorded in a recording device and an amplification sound signal different from the recording sound signal to be output from a speaker. The present technology can be applied to, for example, a sound amplification system that performs off-microphone sound amplification.

Abstract: An echo canceller for executing adaptive processing for canceling an echo component mixed with an audio input signal includes a volume ratio learner configured to compute a volume ratio between an audio output signal externally outputted and the audio input signal mixed with an echo component caused by reflection of the audio output signal to the audio input signal, thereby learning the volume ratio in a regular status in own apparatus, a double-talk detector configured to detect the double-talk status depending on whether a this-time volume ratio computed this time adapts to a double-talk status predicted by the learning of volume ratio and an echo cancel processor configured to control a learning operation of the echo component for the adaptive processing on the basis of a result of the double-talk status detection by the double-talk detector.

Abstract: This technology relates to an image processing apparatus, an image processing method, and a program, which enable easier addition of an effect to a moving image. In a portable terminal device, an ambient environmental sound and a voice uttered by a user are picked up by different sound pickup units when the moving image is shot. A keyword detecting unit detects a keyword determined in advance from the voice uttered by the user and an effect generating unit generates an image effect and a sound effect associated with the detected keyword. Then, an effect adding unit superposes the generated image effect on the shot moving image and synthesizes the generated sound effect with the environmental sound, thereby applying image effects and sound effects to the moving image. According to the portable terminal device, it is possible to easily add a desired effect to the moving image only by uttering the keyword while shooting the moving image. This technology may be applied to a mobile phone.

Abstract: Provided is an audio processing apparatus including a frequency domain conversion unit configured to convert an audio signal input from a microphone to a frequency domain for each of frames, and a gain adjustment unit configured to perform gain adjustment for each of bands on the audio signal converted to the frequency domain. The gain adjustment unit acquires an autocorrelation value of power of the audio signal between the frames for each of the bands, and sets an adjustment amount of the gain in accordance with the acquired autocorrelation value.

Abstract: An audio processing device includes: a directivity adjustment unit adjusting directivity and sharpness thereof in audio picked up by plural microphones picking up audio; and a howling suppression adjustment unit adjusting intensity of suppressing howling of audio picked up by the plural microphones, wherein the directivity adjustment unit adjusts the directivity and sharpness thereof in preference to the howling suppression of audio performed by the howling suppression adjustment unit.

Abstract: An information processing apparatus including: a high-quality-voice determining section configured to determine a voice, which can be determined to have been collected under a good condition, as a good-condition voice included in mixed voices pertaining to a group of voices collected under different conditions; and a voice recognizing section configured to carry out voice recognition processing by making use of a predetermined parameter on the good-condition voice determined by the high-quality-voice determining section, modify the value of the predetermined parameter on the basis of a result of the voice recognition processing carried out on the good-condition voice, and carry out the voice recognition processing by making use of the predetermined parameter having the modified value on a voice included in the mixed voices as a voice other than the good-condition voice.

Abstract: A signal processing apparatus includes: an audio separator that separates audios into a first audio and a second audio using two inputted audio signals; an audio combiner that combines the first audio with the second audio based on proportions of the audios separated by the audio separator; and an image combiner that combines a first image corresponding to the first audio with a second image corresponding to the second audio based on the proportions of the audios separated by the audio separator.

Abstract: n information processing device includes: an estimating section which estimates an amplitude frequency function from a first signal output to a speaker and a second signal input from a microphone; a generating section which generates an estimated echo signal from the first signal and the amplitude frequency function; and a suppressing section which suppresses the estimated echo signal from the second signal, wherein the estimating section changes a coefficient of the amplitude frequency function on the basis of the correlation between the estimated amplitude frequency function and a short-time average amplitude frequency function.

Abstract: Disclosed herein is an echo canceller for use in a sound reinforcement communication system configured to carry out a sound reinforcement communication by utilizing a speaker and a microphone, the echo canceller including: an adaptive filter section configured to adaptively identify an impulse response of a feedback path formed by an acoustic coupling or the like between the speaker and the microphone to estimate an echo component in the feedback path from an input signal to the feedback path, and subtracting the echo component thus estimated from an output signal from the feedback path; and an echo suppressing section configured to execute echo suppressing processing for an output signal from the adaptive filter section.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: An echo canceller used for hands-free communication systems in which hands-free communication is performed by using a speaker and a microphone is disclosed. The echo canceller includes a step size control unit calculating a step size value in an adaptive filter and an adaptive filter unit estimating an echo component of a feedback path from an input signal to the feedback path by adaptively identifying an impulse response of the feedback path formed by an acoustical coupling and the like of the speaker and the microphone, and subtracting the echo component from an output signal from the feedback path, in which the step size control unit calculates a step size value by using an echo reduction amount defined based on the ratio between the output signal from the feedback path and a residual signal and outputs the value to the adaptive filter unit.

Abstract: An audio processor of a loud speech communication system including a speaker and a microphone is provided. The audio processor includes: an adaptive filter wherein an amount of update in a learning event is set to an arbitrary value, and a filter coefficient is serially determined corresponding to the set amount of update; a semi-fixed filter adapted to an echo cancellation process of an audio input signal input from the microphone; adaptive filter assessment unit that calculates a length of an update vector based on the filter coefficient determined by the adaptive filter and a length of an update vector based on a filter coefficient set in the semi-fixed filter and that performs assessment of the filter coefficients in accordance with the update vectors; and coefficient specifying unit that sets an optimal filter coefficient among the filter coefficients into the semi-fixed filter in accordance with the result of the assessment of the filter coefficients performed by the adaptive filter assessment unit.

Abstract: Disclosed herein is a sound source direction detecting apparatus including: a plurality of microphones configured to collect sounds from a sound source in order to form an audio frame; a frequency decomposition section configured to decompose the audio frame into frequency components; an error range determination section configured to determine the effects of noises collected together with the sounds as an error range relative to phases; a power level dispersion section configured to disperse power levels of the sounds for each of the frequency components decomposed by the frequency decomposition section, on the basis of the error range determined by the error range determination section; a power level addition section configured to add the power levels dispersed by the power level dispersion section; and a sound source direction detection section configured to detect the direction of the sound source based on the phase at which is located the highest of the power levels added by the power level addition sect

Abstract: An echo canceller formed of an adaptive filter is designed such that even under a condition where a system transmission delay is undefined, an appropriate delay time can be set in a delay circuit that absorbs a system delay, and that an effective echo cancellation effect can always be achieved. A time difference of a transmission path until a reproduction audio signal input to the delay circuit is input as a processing target signal of an adaptive filter system through a space between a speaker and a microphone is determined, and the delay time corresponding to this time difference is set in the delay circuit. At this time, the speaker and the microphone are placed so that the distance therebetween is small, and the delay time of the delay circuit is set to 0. Thus, the determined time difference indicates a system transmission delay in the above transmission path. That is, an accurate delay time corresponding to the system transmission delay can be set in the delay circuit.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.