摘要:
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.
摘要:
Disclosed herein is an audio processing apparatus for processing a plurality of pieces of audio data of sounds picked up by a plurality of microphones. The apparatus includes: a speaker identification section configured to identify a speaker based on the audio data; a simultaneous speech section identification section configured to, when at least first and second speakers have been identified, identify speech sections during which the first and second speakers have made speeches, and identify a section during which the first and second speakers have made the speeches at the same time as a simultaneous speech section; and an arranging section configured to separate audio data of the first speaker and audio data of the second speaker from the simultaneous speech section, and allow the audio data of the first speaker and the audio data of the second speaker to be outputted at mutually different timings.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
When an user inputs route search conditions such as place where the user is now, destination, and moving purpose, the route search conditions are transmitted to a navegation server. In the navegation server, map data corresponding to the route search conditions is detected and provided to a portable terminal.