Abstract: The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. The encoding device stores speaker arrangement information in a comment region in a PCE of an encoded bit stream and stores a synchronous word and identification information in the comment region such that other public comments and the speaker arrangement information stored in the comment region can be distinguished from each other. When an encoded bit stream is decoded, it is determined whether the speaker arrangement information is stored on the basis of the synchronous word and the identification information stored in the comment region. Audio data included in the encoded bit stream is output according to the arrangement of the speakers corresponding to the determination result. The present technique can be applied to an encoding device.

Abstract: The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: The present technology relates to an information processing device and method for allowing a sound image to be localized with higher precision, and a program. When a target sound image is outside a mesh, the target sound image is moved in a vertical direction while a position in a horizontal direction of the target sound image remains fixed, so that the target sound image is present on a boundary of the mesh. Specifically, a mesh detection unit detects a mesh including a position in the horizontal direction of the target sound image. A candidate position calculation unit calculates a position that is a movement target of the target sound image, based on loudspeaker positions that are at opposite ends of an arc of the detected mesh that is a destination, and the position in the horizontal direction of the target sound image. As a result, the target sound image can be moved onto a boundary of the mesh. The present technology is applicable to a sound processing device.

Abstract: In order to obtain coded data which does not strike viewers and listeners as being incongruous, when plural audio data are to be coded, a coding program groups the respective audio data into one audio data, codes the grouped audio data in sequence with a predetermined number of samples being treated as units, and sets delimitations corresponding to the delimitations of the plural audio data in the coded data at coding units of the coded data.

Abstract: [Object] To provide an information processing device capable of establishing a sound statistic database with high usability. [Solution] The information processing device includes: a feature amount extraction unit configured to extract a feature amount from sounds in a predetermined space; an identification unit configured to identify a type of a sound included in the sounds from the feature amount; an information combining unit configured to generate sound statistic information in which the feature amount, the type of the sound, and spatiotemporal information of the predetermined space at a time of acquisition of the sound are associated with each other; and a database establishing unit configured to establish a sound statistic database by using the sound statistic information.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, capable of improving audio signal transmission efficiency. An identification information generation unit determines whether or not an audio signal is to be encoded on the basis of the audio signal. The identification information generation unit generates identification information indicating the determination result. An encoding unit encodes only the audio signals which are determined to be encoded. A packing unit generates a bit stream containing the identification information and encoded audio signals. As a result of storing only encoded audio signals and the identification information indicating whether or not the respective audio signals are to be encoded in the bit stream, the transmission efficiency of the audio signals can be improved. The present technology can be applied to an encoder and a decoder.

Abstract: There is provided a decoding device including at least one circuit configured to acquire one or more encoded audio signals including a plurality of channels and/or a plurality of objects and priority information for each of the plurality of channels and/or the plurality of objects, and to decode the one or more encoded audio signals according to the priority information.

Abstract: The present invention pertains to an encoding device and method, a decoding device and method, and to a program, with which sound of an appropriate volume level can be obtained with a smaller quantity of codes. A first gain calculation circuit calculates a first gain for volume level correction of an input time series signal, and a second gain calculation circuit calculates a second gain for volume level correction of a downmixed signal obtained by downmixing of the input time series signal. A gain encoding circuit computes the gain differential between the first gain and the second gain, the gain differential between time frames, and the gain differential within time frames, and encodes the first gain and the second gain. The present invention can be applied in encoding devices and decoding devices.

Abstract: The present technology relates to an audio signal output device and a method, an encoding device and a method, a decoding device and a method, and a program for realizing audio reproduction with a more realistic feeling. In a case where an audio signal generated to be output as a sound from an ideal speaker that is a virtual speaker placed in an ideal position is input, the distance between the ideal speaker and a real reproduction speaker is determined. Gain adjustment is then performed on the audio signal with the gain corresponding to the determined distance, the audio signal subjected to the gain adjustment is reproduced by the reproduction speaker. Accordingly, even in a case where there is a difference in position between the ideal speaker and the reproduction speaker, audio reproduction with a more realistic feeling can be realized. The present technology can be applied to reproduction devices.

Abstract: A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted.

Abstract: The present technique relates to an encoding device and a method, a decoding device and a method, and a program capable of obtaining higher quality audio. An encoding unit encodes position information and a gain of an object in a current frame in multiple encoding modes. A compressing unit generates, for each combination of encoding modes of each pieces of position information and gains, encoded meta data including encoding mode information indicating the encoding modes and encoded data which are the encoded position information and gains, and compresses the encoding mode information included in the encoding meta data. A determining unit selects encoded meta data of which amount of data is the least from among the encoded meta data generated for each combination, thus determining the encoding mode of each pieces of position information and gains. The present technique can be applied to an encoder and a decoder.

Abstract: The present technology relates to a coding device and method, and a decoding device and method, and a program capable of reducing the amount of calculations for decoding. A separating unit separates a supplied bit stream into coded data of channel sources including a dialog source, coded data of additional data sources, and coded data of dialog information. A dialog information decoding unit decodes the coded data of the dialog information. When the dialog information acquired by the decoding is presented to a viewer, the viewer selects one source from the dialog source and some additional dialog sources. An additional dialog source decoding unit decodes only the coded data of an additional dialog source selected by the viewer. An additional dialog selection unit outputs a viewer-selected audio signal from among the audio signals of the dialog source and additional dialog sources in response to the selection instruction of the viewer.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program which can obtain high quality audio with a less code amount. A signal encoding unit encodes audio signals and outputs a resultant signal code string. A coefficient encoding unit encodes mixing coefficients for use in a mixing process of the audio signals and outputs a resultant coefficient code string. A multiplexing unit multiplexes the signal code string and the coefficient code string and outputs a resultant output code string. The coefficient encoding unit rearranges the mixing coefficients at the time of encoding the mixing coefficients on the basis of distances between input-side sound source positions and speaker positions on a reproduction side and calculates differential values between the mixing coefficients on the basis of arrangement order of the mixing coefficients, thereby encoding the mixing coefficients.

Abstract: A method, system, and computer program product for processing an encoded audio signal is described. In one exemplary embodiment, the system receives an encoded low-frequency range signal and encoded energy information used to frequency shift the encoded low-frequency range signal. The low-frequency range signal is decoded and an energy depression of the decoded signal is smoothed. The smoothed low-frequency range signal is frequency shifted to generate a high-frequency range signal. The low-frequency range signal and high-frequency range signal are then combined and outputted.

Abstract: The present technology relates to a sound processing apparatus and method, and a program for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: The present technology relates to an audio signal processing device and method, an encoding device and method, and a program, which are capable of obtaining a higher quality sound. A selection unit selects, from supplied multichannel audio signals, audio signals of a channel of a dialogue sound and audio signals of a channel to be downmixed. A downmixing unit downmixes the audio signals of the channel to be downmixed. An addition unit adds the audio signals of the channel of a dialogue sound to audio signals of a predetermined channel among audio signals of one or more channels obtained in the downmixing. The present technology can be applied to a decoder.

Abstract: The present technology relates to a sound processing apparatus and method, and a program for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: A signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. An envelope information generating unit generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. The high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream.

Abstract: An imaging system includes: a first lighting device for irradiating an imaging target object with visible light; a second lighting device for irradiating the imaging target object with near-infrared light; and an image sensor for photodetecting visible light caused by the visible light and coming from the imaging target object and fluorescence caused by the near-infrared light and coming from the imaging target object during a predetermined shutter open period every frame of a predetermined period. The image sensor outputs photodetection signals corresponding to photodetection amounts of the visible light and the fluorescence. The imaging system also includes a controller that generates a composite image of a visible image and a fluorescence image based on the photodetection signals, and has a lighting controller that turns on the second lighting device with optical power corresponding to the shutter open period in synchronism with the shutter open period.

Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.