Audio processing equipment, threshold value recognition devices, and audio processing methods.

Abstract

The invention relates to an acoustic processing device (1001) comprising a circuit (1402) and a memory (1404). Using the memory (1404), the circuit (1402) receives acoustic spatial information about the acoustic space; receives the volume ratio between the volume of the direct sound and the volume of the reflected sound, where the direct sound is produced from a sound source in the acoustic space and the reflected sound is produced in the acoustic space in response to the direct sound; receives information about additional processing; and controls whether or not to select the reflected sound based on the information about additional processing and the volume ratio.

Description

Sound processing device, threshold specifying device, and sound processing method

[0001] The present disclosure relates to a sound processing device and the like.

[0002] In recent years, products and services using ER (Extended Reality) (which may also be expressed as XR), including VR (Virtual Reality), AR (Augmented Reality), and MR (Mixed Reality), have become increasingly popular. Accordingly, the importance of acoustic processing technology that provides immersive audio to listeners in a virtual or real space by adding acoustic effects that occur according to the environment of the space to sounds emitted from a virtual sound source is increasing.

[0003] The listener may also be expressed as a listener or a user. Furthermore, Patent Documents 1, 2, 3, 4, Non-Patent Documents 1 and 2 disclose techniques related to the sound processing device and sound processing method of the present disclosure.

[0004] Japanese Patent No. 6288100 Japanese Patent Application Laid-Open No. 2019-22049 International Publication No. 2021 / 180938 International Publication No. 2023 / 083780

[0005] B. C. J. Moore, "Introduction to Auditory Psychology," Seishin Shobo, April 20, 1994, Chapter 6: Spatial Perception, p. 225Fabian Brinkmann et al. , “The FABIAN head-related transfer function data base”, 2020 / 1 / 22, https: / / sofacoustics. org / data / database / tu-berlin / FABIAN_Documentation. pdf

[0006] For example, Patent Literature 1 discloses a technology for performing signal processing on an object audio signal and presenting the processed signal to a listener. As ER technology becomes more widespread and services using ER technology become more diverse, there is a demand for acoustic processing that corresponds to differences in, for example, the acoustic quality required by each service, the signal processing capabilities of the terminal used, and the sound quality that can be provided by the sound presentation device. In addition, further improvements in acoustic processing technology are required to provide such services.

[0007] Here, an improvement in sound processing technology refers to a change to an existing sound processing. For example, the improvement in sound processing technology may provide a process for adding a new sound effect, a reduction in the amount of sound processing, an improvement in the quality of the sound obtained by the sound processing, a reduction in the amount of data used to perform the sound processing, or an simplification in the acquisition or generation of information used to perform the sound processing. Alternatively, the improvement in sound processing technology may provide a combination of any two or more of these.

[0008] In particular, improvements are required in devices or services that allow listeners to move freely in a virtual space. However, the above-mentioned effects obtained by improvements in sound processing technology are merely examples. One or more aspects grasped based on the present disclosure may be aspects conceived based on a different perspective than the above, aspects that achieve a different object than the above, or aspects that obtain an effect different from the above.

[0009] An audio processing device according to one aspect of the present disclosure includes a circuit and a memory, and the circuit uses the memory to acquire sound space information relating to a sound space, and based on the sound space information, acquires a volume ratio between the volume of a direct sound generated from a sound source in the sound space and the volume of a reflected sound generated in the sound space corresponding to the direct sound, acquires information relating to additional processing, and controls whether or not to select the reflected sound based on the information relating to the additional processing and the volume ratio.

[0010] These comprehensive or specific aspects may be realized as a system, an apparatus, a method, an integrated circuit, a computer program, or a non-transitory recording medium such as a computer-readable CD-ROM, or may be realized as any combination thereof.

[0011] One aspect of the present disclosure may provide, for example, processing to impart new acoustic effects, reducing the amount of processing required for acoustic processing, improving the sound quality of audio obtained by acoustic processing, reducing the amount of data required for acoustic processing, or facilitating the acquisition or generation of information required for acoustic processing. Alternatively, one aspect of the present disclosure may provide any combination of these. As a result, one aspect of the present disclosure may provide acoustic processing suited to the listener's usage environment, thereby contributing to an improved acoustic experience for the listener.

[0012] In particular, the above-described effects can be achieved in devices or services that allow listeners to move freely within a virtual space. However, the above-described effects are merely examples of the effects of various aspects grasped based on the present disclosure. Each of one or more aspects grasped based on the present disclosure may be an aspect conceived based on a different perspective than the above, an aspect that achieves a different purpose than the above, or an aspect that obtains a different effect than the above.

[0013] FIG. 1 is a diagram showing an example of direct sound and reflected sound generated in a sound space. FIG. 2 is a diagram showing an example of a stereophonic sound reproduction system according to an embodiment. FIG. 3A is a block diagram showing an example of the configuration of an encoding device according to an embodiment. FIG. 3B is a block diagram showing an example of the configuration of a decoding device according to an embodiment. FIG. 3C is a block diagram showing another example of the configuration of an encoding device according to an embodiment. FIG. 3D is a block diagram showing another example of the configuration of a decoding device according to an embodiment. FIG. 4A is a block diagram showing an example of the configuration of a decoder according to an embodiment. FIG. 4B is a block diagram showing another example of the configuration of a decoder according to an embodiment. FIG. 5 is a diagram showing an example of the physical configuration of an audio signal processing device according to an embodiment. FIG. 6 is a diagram showing an example of the physical configuration of an encoding device according to an embodiment. FIG. 7 is a block diagram showing an example of the configuration of a rendering unit according to an embodiment. FIG. 8 is a flowchart showing an example of the operation of the audio signal processing device according to an embodiment. FIG. 9 is a diagram showing a positional relationship between a listener and an obstacle object that is relatively far away. FIG. 10 is a diagram showing a positional relationship between a listener and an obstacle object that is relatively close. FIG. 11 is a diagram showing the relationship between the time difference between direct sound and reflected sound and a threshold. FIG. 12A is a diagram showing a part of an example of a method for setting threshold data. FIG. 12B is a diagram showing a part of an example of a method for setting threshold data. FIG. 12C is a diagram showing part of an example of a method for setting threshold data. FIG. 13 is a diagram showing an example of a method for setting thresholds. FIG. 14 is a flowchart showing an example of selection processing. FIG. 15 is a diagram showing a listening experiment of leading sounds and following sounds related to the leading sound effect. FIG. 16 is a diagram showing experimental parameters in the horizontal plane direction. FIG. 17 is a diagram showing experimental parameters in the vertical plane direction. FIG. 18 is a diagram showing the relationship between the direction of direct sound, the direction of reflected sound, the time difference, and the threshold. FIG. 19 is a diagram showing the relationship between the angle difference, the time difference, and the threshold. FIG. 20 is a block diagram showing another example of the configuration of a rendering unit. FIG. 21 is a flowchart showing another example of selection processing. FIG. 22 is a flowchart showing yet another example of selection processing. FIG. 23 is a flowchart showing a first modified example of the operation of the audio signal processing device in the embodiment. FIG. 24 is a flowchart showing a second modified example of the operation of the audio signal processing device in the embodiment.FIG. 25 is a diagram showing an example of the arrangement of avatars, sound source objects, and obstacle objects. FIG. 26 is a flowchart showing yet another example of selection processing. FIG. 27 is a block diagram showing an example of a configuration for a rendering unit to perform pipeline processing. FIG. 28 is a diagram showing sound transmission and diffraction. FIG. 29 is a graph showing thresholds. FIG. 30 is a diagram showing a first example of use of thresholds. FIG. 31 is a diagram showing a second example of use of thresholds. FIG. 32 is a diagram showing a third example of use of thresholds. FIG. 33 is a diagram showing a fourth example of use of thresholds. FIG. 34 is a flowchart showing a first example of basic operation of an audio signal processing device in accordance with an embodiment. FIG. 35 is a flowchart showing a second example of basic operation of an audio signal processing device in accordance with an embodiment. FIG. 36 is a flowchart showing a third example of basic operation of an audio signal processing device in accordance with an embodiment.

[0014] (Findings that form the basis of the present disclosure) Figure 1 is a diagram showing an example of direct sound and reflected sound generated in a sound space. In acoustic processing that expresses the characteristics of a virtual space with sound, it is effective to reproduce not only direct sound but also reflected sound in order to express the size of the space, the material of the walls, etc., and to accurately grasp the position of the sound source (localization of the sound image).

[0015] For example, when listening to sound in a rectangular room as shown in Figure 1, six primary reflections are generated for a single sound source, corresponding to the six walls. Reproducing these reflections provides clues for a proper understanding of the space and sound image. Furthermore, for each reflection, secondary reflections are generated from surfaces other than the surface that generated the reflection. These reflections also provide useful perceptual clues.

[0016] However, even if only secondary reflections are taken into account, one sound source will produce one direct sound and 36 (6 + 6 x 5) reflected sounds, resulting in 37 sound rays, and a considerable amount of calculation is required to process these sound rays.

[0017] Furthermore, in recent applications envisioned for the Metaverse, such as virtual meetings, virtual shopping, or virtual concerts, multiple sound sources will inevitably be present, requiring even greater amounts of computation.

[0018] In addition, listeners who listen to sounds in a virtual space use headphones or VR goggles. To provide such listeners with stereophonic sound, binaural processing is performed on each sound ray, which provides a sound pressure ratio and phase difference between the two ears to reproduce the direction of sound arrival and the sense of perspective. Therefore, if all reflected sounds are to be reproduced, the amount of calculation required becomes enormous.

[0019] On the other hand, for convenience, small storage batteries are sometimes used as the batteries for VR goggles worn by listeners who experience virtual space. In order to extend the battery life, it is desirable to reduce the computational load required for the above-mentioned processing. To achieve this, it is desirable to reduce the number of sound rays, which may number on the order of several hundred, to a degree that does not impair sound localization and spatial understanding.

[0020] Furthermore, in some sound reproduction systems, degrees of freedom such as 6 DoF (6 Degrees of Freedom) are allowed for the position and orientation of the listener. In this case, the positional relationship between the listener, the sound source, and the object that reflects the sound is not determined until playback (rendering). Therefore, the reflected sound is also not determined until playback. Therefore, it is difficult to determine the reflected sound to be processed in advance.

[0021] Therefore, appropriately selecting one or more reflected sounds to be processed or not to be processed from among a plurality of reflected sounds occurring in a sound space during playback is useful for appropriately reducing the amount of calculation and the calculation load.

[0022] Therefore, an object of the present disclosure is to provide a sound processing device and the like that can appropriately control whether or not to select a sound generated in a sound space.

[0023] Note that controlling whether to select a sound corresponds to determining whether to select a sound. Furthermore, selecting a sound may mean selecting the sound as a sound to be processed, or may mean selecting the sound as a sound not to be processed.

[0024] (Summary of the Disclosure) A sound processing device according to a first aspect grasped based on the present disclosure includes a circuit and a memory, and the circuit uses the memory to acquire sound space information relating to the sound space, acquires characteristics relating to a first sound generated from a sound source in the sound space based on the sound space information, and controls whether to select a second sound generated in the sound space corresponding to the first sound based on the characteristics relating to the first sound.

[0025] The device of the above aspect can appropriately control whether to select a second sound generated in a sound space corresponding to a first sound, based on the characteristics of the first sound generated in the sound space. That is, it is possible to appropriately control whether to select a sound generated in the sound space. Therefore, it is possible to appropriately reduce the amount of calculation and the calculation load.

[0026] A sound processing device according to a second aspect of the present disclosure may be the sound processing device of the first aspect, in which the first sound is a direct sound and the second sound is a reflected sound.

[0027] The device according to the above aspect can appropriately control whether or not to select reflected sounds based on the characteristics of the direct sounds.

[0028] A sound processing device according to a third aspect that can be understood based on the present disclosure may be the sound processing device of the second aspect, in which the characteristic related to the first sound is a volume ratio between the volume of the direct sound and the volume of the reflected sound, and the circuit calculates the volume ratio based on sound space information, and controls whether or not to select the reflected sound based on the volume ratio.

[0029] The device according to the above aspect can appropriately select reflected sounds that have a large influence on the listener's perception, based on the volume ratio between the volume of the direct sound and the volume of the reflected sounds.

[0030] A sound processing device according to a fourth aspect as understood based on the present disclosure may be the sound processing device of the third aspect, in which, when a reflected sound is selected, the circuit applies binaural processing to the reflected sound and the direct sound to generate sounds that arrive at each of the listener's ears.

[0031] The device according to the above aspect can appropriately select reflected sounds that have a large influence on the listener's perception, and apply binaural processing to the selected reflected sounds.

[0032] A sound processing device according to a fifth aspect as understood based on the present disclosure may be the sound processing device of the third or fourth aspect, in which the circuit calculates the time difference between the end time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference and the volume ratio.

[0033] The device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the time difference between the end time of the direct sound and the arrival time of the reflected sound and the volume ratio between the volume of the direct sound and the volume of the reflected sound. Therefore, the device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the post-masking effect.

[0034] A sound processing device according to a sixth aspect that can be understood based on the present disclosure may be the sound processing device of the fifth aspect, wherein the circuit selects reflected sound when the volume ratio is equal to or greater than a threshold, and the first threshold used as the threshold when the time difference is a first value is greater than the second threshold used as the threshold when the time difference is a second value that is greater than the first value.

[0035] The device of the above aspect can increase the likelihood of selecting a reflected sound with a large time difference between the end time of the direct sound and the arrival time of the reflected sound, thereby enabling the device of the above aspect to appropriately select a reflected sound that has a large influence on the listener's perception.

[0036] A sound processing device according to a seventh aspect as understood based on the present disclosure may be the sound processing device of the third or fourth aspect, in which the circuit calculates the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference and the volume ratio.

[0037] The device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the time difference between the arrival time of the direct sound and the arrival time of the reflected sound and the volume ratio between the volume of the direct sound and the volume of the reflected sound. Therefore, the device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the precedence effect.

[0038] An audio processing device according to an eighth aspect of the present disclosure may be the audio processing device of the seventh aspect, wherein the circuit selects reflected sound when the volume ratio is equal to or greater than a threshold, and the first threshold used as the threshold when the time difference is a first value is greater than the second threshold used as the threshold when the time difference is a second value greater than the first value.

[0039] The device of the above aspect can increase the likelihood of selecting a reflected sound with a large time difference between the arrival time of the direct sound and the arrival time of the reflected sound, thereby enabling the device of the above aspect to appropriately select a reflected sound that has a large influence on the listener's perception.

[0040] A sound processing device according to a ninth aspect as understood based on the present disclosure may be the sound processing device of the eighth aspect, in which the circuit adjusts the threshold value based on the direction of arrival of the direct sound and the direction of arrival of the reflected sound.

[0041] The device according to the above aspect can appropriately select reflected sounds that have a large influence on the listener's perception, based on the direction from which the direct sound and the reflected sounds come.

[0042] An audio processing device according to a tenth aspect as understood based on the present disclosure may be an audio processing device according to any one of the second to ninth aspects, wherein the circuit corrects the volume of the direct sound based on the volume of the reflected sound when the reflected sound is not selected.

[0043] The device of the above aspect can appropriately reduce the sense of incongruity that occurs when reflected sounds are not selected and the volume of the reflected sounds is lacking, with a small amount of calculation.

[0044] An audio processing device according to an eleventh aspect of the present disclosure may be an audio processing device according to any one of the second to ninth aspects, wherein the circuit synthesizes reflected sound into direct sound when reflected sound is not selected.

[0045] The device of the above aspect can more accurately reflect the characteristics of the reflected sound in the direct sound, thereby reducing the sense of discomfort that occurs when the reflected sound is not selected and is therefore absent.

[0046] A sound processing device according to a twelfth aspect as understood based on the present disclosure may be a sound processing device according to any one of the third to ninth aspects, in which the volume ratio is the volume ratio between the volume of a direct sound at a first time and the volume of a reflected sound at a second time different from the first time.

[0047] When the time at which direct sound is perceived and the time at which reflected sound is perceived are different, the device of the above aspect can appropriately select the reflected sound that has the greatest influence on the listener's perception based on the volume ratio between the direct sound and the reflected sound at the different times.

[0048] A sound processing device according to a thirteenth aspect as understood based on the present disclosure may be a sound processing device according to the first or second aspect, in which the circuit sets a threshold based on characteristics related to the first sound and controls whether or not to select the second sound based on the threshold.

[0049] The device of the above aspect can appropriately control whether or not to select the second sound based on a threshold value that is set based on the characteristics of the first sound.

[0050] A sound processing device according to a fourteenth aspect understood based on the present disclosure may be a sound processing device according to any one of the first, second, and thirteenth aspects, in which the characteristic related to the first sound is any one of the volume of the sound source, the visibility of the sound source, and the positioning of the sound source, or a combination of any two or more thereof.

[0051] The device of the above aspect can appropriately control whether or not to select the second sound based on the volume of the sound source, the visibility of the sound source, or the localization of the sound source.

[0052] A sound processing device according to a fifteenth aspect as understood based on the present disclosure may be a sound processing device according to any one of the first, second, and thirteenth aspects, in which the characteristic related to the first sound is the frequency characteristic of the first sound.

[0053] The device of the above aspect can appropriately control whether or not to select the second sound that is generated in response to the first sound, based on the frequency characteristics of the first sound.

[0054] A sound processing device according to a sixteenth aspect as understood based on the present disclosure may be a sound processing device according to any one of the first, second, and thirteenth aspects, in which the characteristic related to the first sound is a characteristic indicating the intermittency of the amplitude of the first sound.

[0055] The device of the above aspect can appropriately control whether or not to select a second sound that occurs in response to a first sound, based on a characteristic that indicates the intermittency of the amplitude of the first sound.

[0056] A sound processing device according to a seventeenth aspect as understood based on the present disclosure may be a sound processing device according to any one of the first, second, thirteenth and sixteenth aspects, in which the characteristic related to the first sound is a characteristic indicating the duration of a sound portion of the first sound or the duration of a silent portion of the first sound.

[0057] The device of the above aspect can appropriately control whether or not to select a second sound that occurs in response to a first sound, based on a characteristic indicating the duration of the sound portion of the first sound or the duration of the silent portion of the first sound.

[0058] An audio processing device according to an 18th aspect understood based on the present disclosure may be an audio processing device according to any one of the 1st, 2nd, 13th, 16th and 17th aspects, in which the characteristic related to the first sound is a characteristic indicating the duration of the sound portion of the first sound and the duration of the silent portion of the first sound in time series.

[0059] The device of the above aspect can appropriately control whether or not to select a second sound that occurs in response to a first sound, based on characteristics that indicate the duration of the sound portion of the first sound and the duration of the silent portion of the first sound in a time series.

[0060] A sound processing device according to a 19th aspect understood based on the present disclosure may be a sound processing device according to any one of the 1st, 2nd, 13th and 15th aspects, in which the characteristic related to the first sound is a characteristic indicating a fluctuation in the frequency characteristics of the first sound.

[0061] The device of the above aspect can appropriately control whether or not to select a second sound that occurs in response to a first sound, based on the characteristic indicating the fluctuation in the frequency characteristic of the first sound.

[0062] The sound processing device according to the twentieth aspect as understood based on the present disclosure may be a sound processing device that is any one of the sound processing devices according to the first, second, thirteenth, fifteenth and nineteenth aspects, in which the characteristic related to the first sound is a characteristic indicating the constancy of the frequency characteristics of the first sound.

[0063] The device of the above aspect can appropriately control whether or not to select the second sound that occurs in response to the first sound, based on the characteristic indicating the stationary nature of the frequency characteristics of the first sound.

[0064] The sound processing device according to the 21st aspect as understood based on the present disclosure may be a sound processing device according to any one of the 1st, 2nd, and 13th to 20th aspects, in which the characteristics related to the first sound are obtained from a bitstream.

[0065] The device of the above aspect can appropriately control whether or not to select a second sound that is generated in response to a first sound, based on information obtained from the bitstream.

[0066] The sound processing device according to the 22nd aspect as understood based on the present disclosure may be a sound processing device according to any one of the 1st, 2nd, and 13th to 21st aspects, in which the circuit calculates characteristics related to the second sound and controls whether to select the second sound based on the characteristics related to the first sound and the characteristics related to the second sound.

[0067] The device of the above aspect can appropriately control whether or not to select the second sound that is generated in response to the first sound, based on the characteristics related to the first sound and the characteristics related to the second sound.

[0068] A sound processing device according to a 23rd aspect that can be understood based on the present disclosure may be the sound processing device according to the 22nd aspect, in which the circuit acquires a threshold value indicating a volume that corresponds to the boundary between whether a sound can be heard or not, and controls whether to select the second sound based on the characteristics related to the first sound, the characteristics related to the second sound, and the threshold value.

[0069] The device of the above aspect can appropriately control whether to select the second sound based on the characteristics of the first sound, the characteristics of the second sound, and a threshold corresponding to whether the second sound can be heard.

[0070] A sound processing device according to a 24th aspect grasped based on the present disclosure may be the sound processing device of the 23rd aspect, in which the characteristic related to the second sound is the volume of the second sound.

[0071] The device of the above aspect can appropriately control whether or not to select the second sound based on the volume of the second sound.

[0072] A sound processing device according to a 25th aspect grasped based on the present disclosure may be the sound processing device of the first or second aspect, in which the sound space information includes information on the position of the listener in the sound space, the second sounds are each of a plurality of second sounds that occur in the sound space corresponding to the first sound, and the circuit selects one or more processing target sounds to which binaural processing is applied from the plurality of second sounds by controlling whether or not to select each of the plurality of second sounds based on characteristics related to the first sound.

[0073] The device of the above aspect can appropriately select one or more processing target sounds to which binaural processing is applied from among a plurality of second sounds that occur in the sound space corresponding to the first sound, based on characteristics related to the first sound that occurs in the sound space.

[0074] A sound processing device according to a 26th aspect as understood based on the present disclosure may be a sound processing device according to the 25th aspect, in which the timing for acquiring the characteristics related to the first sound is at least one of when the sound space is created, when processing of the sound space starts, and when an information update thread occurs during processing of the sound space.

[0075] The device of the above aspect can appropriately select one or more processing target sounds to which binaural processing is applied, based on information acquired at adaptive timing.

[0076] A sound processing device according to a 27th aspect of the present disclosure may be a sound processing device according to the 26th aspect, in which the characteristics related to the first sound are acquired periodically after processing of the sound space begins.

[0077] The device of the above aspect can appropriately select one or more processing target sounds to which binaural processing is applied, based on information acquired periodically.

[0078] A sound processing device according to a 28th aspect grasped based on the present disclosure may be a sound processing device according to the first or second aspect, in which the characteristic related to the first sound is the volume of the first sound, and the circuit calculates an evaluation value of the second sound based on the volume of the first sound, and controls whether to select the second sound based on the evaluation value.

[0079] The device of the above aspect can appropriately control whether or not to select the second sound, based on the evaluation value calculated for the second sound based on the volume of the first sound.

[0080] A sound processing device according to a 29th aspect grasped based on the present disclosure may be the sound processing device of the 28th aspect, in which the volume of the first sound has a transition.

[0081] The device according to the above aspect can appropriately control whether or not to select the second sound based on the evaluation value calculated based on the volume having the transition.

[0082] A sound processing device according to a thirtieth aspect as understood based on the present disclosure may be a sound processing device according to the twenty-eighth or twenty-ninth aspect, in which the circuit calculates an evaluation value such that the louder the first sound is, the more likely the second sound is to be selected.

[0083] The device of the above aspect can appropriately control whether or not to select the second sound based on an evaluation value that is set to a value that makes it more likely that the second sound will be selected as the volume of the first sound increases.

[0084] A sound processing device according to a 31st aspect grasped based on the present disclosure may be the sound processing device of the first or second aspect, wherein the sound space information is scene information including information on a sound source in the sound space and information on the position of a listener in the sound space, the second sound is each of a plurality of second sounds that occur in the sound space corresponding to the first sound, and the circuit acquires a signal of the first sound, calculates the plurality of second sounds based on the scene information and the signal of the first sound, acquires characteristics related to the first sound from the information on the sound source, and controls, based on the characteristics related to the first sound, whether or not to select each of the plurality of second sounds as a sound to which binaural processing is not applied, thereby selecting one or more second sounds to which binaural processing is not applied from among the plurality of second sounds.

[0085] The device of the above aspect can appropriately select one or more second sounds to which binaural processing is not applied from among multiple second sounds that occur in a sound space corresponding to a first sound, based on characteristics related to the first sound.

[0086] A sound processing device according to a 32nd aspect of the present disclosure may be the sound processing device according to the 31st aspect, in which the scene information is updated based on the input information, and the characteristics related to the first sound are acquired in response to the update of the scene information.

[0087] The device of the above aspect can appropriately select one or more second sounds to which binaural processing is not applied based on information acquired in response to an update of the scene information.

[0088] An audio processing device according to a 33rd aspect as understood based on the present disclosure may be an audio processing device according to the 31st or 32nd aspect, in which the scene information and the characteristics related to the first sound are obtained from metadata included in the bitstream.

[0089] The device of the above aspect can appropriately select one or more second sounds to which binaural processing is not applied based on information obtained from metadata included in the bitstream.

[0090] An acoustic processing method according to a 34th aspect grasped based on the present disclosure includes the steps of acquiring sound space information relating to a sound space, acquiring characteristics of a first sound generated from a sound source in the sound space based on the sound space information, and controlling whether to select a second sound generated in the sound space corresponding to the first sound based on the characteristics of the first sound.

[0091] The method of the above aspect can achieve the same effects as the sound processing device of the first aspect.

[0092] A program according to a thirty-fifth aspect grasped based on the present disclosure is a program for causing a computer to execute the acoustic processing method of the thirty-fourth aspect.

[0093] The program of the above aspect can achieve the same effect as the acoustic processing method of the 34th aspect when used with a computer.

[0094] A sound processing device according to a 36th aspect that can be understood based on the present disclosure may be a sound processing device according to any one of the 3rd to 9th and 12th aspects, and may include an addition means for applying additional processing to the second sound, an addition amount acquisition means for acquiring an amount of increase or decrease in amplitude in the addition means, and a volume ratio adjustment means for adjusting the volume ratio in accordance with the amount of increase or decrease acquired by the addition amount acquisition means.

[0095] The device according to the above aspect can appropriately adjust the volume ratio in accordance with the amplitude that increases or decreases due to the additional processing.

[0096] A sound processing device according to a 37th aspect as understood based on the present disclosure may be a sound processing device according to any one of the 5th to 9th aspects, and may be a sound processing device comprising: an addition means for applying additional processing to the second sound; an additional delay amount acquisition means for acquiring an amount of increase or decrease in the delay in the addition means; and a time difference adjustment means for adjusting the time difference according to the amount of increase or decrease acquired by the additional delay amount acquisition means.

[0097] The device of the above aspect can appropriately adjust the time difference according to the delay that increases or decreases due to the additional processing.

[0098] A sound processing device according to a 38th aspect as understood based on the present disclosure may be a sound processing device according to the 36th or 37th aspect, in which the additional processing is processing that is not applied to the first sound.

[0099] The device of the above aspect can impart additional processing to the second sound that is not imparted to the first sound. Therefore, the device of the above aspect can impart additional processing suitable for the second sound to the second sound.

[0100] A sound processing device according to a 39th aspect that can be understood based on the present disclosure may be a sound processing device according to the 36th or 37th aspect, in which the adding means is a filter means that auditorily emphasizes the diffusion of sound.

[0101] The device of the above aspect can impart additional processing to the second sound by means of a filter means that auditorily emphasizes the diffusion of the sound.

[0102] A sound processing device according to a fortieth aspect grasped based on the present disclosure may be a sound processing device that includes a circuit and a memory, in which the circuit uses the memory to acquire sound space information regarding the sound space, acquires a volume ratio between the volume of direct sound generated from a sound source in the sound space and the volume of reflected sound generated in the sound space corresponding to the direct sound based on the sound space information, acquires information regarding additional processing, and controls whether or not to select reflected sound based on the information regarding the additional processing and the volume ratio.

[0103] The device of the above aspect can appropriately control whether to select reflected sounds based on information about additional processing and the volume ratio between direct sound and reflected sounds. That is, it is possible to appropriately control whether to select reflected sounds that occur in response to direct sound generated from a sound source in a sound space. Therefore, it is possible to appropriately reduce the amount of calculation and the calculation load.

[0104] An audio processing device according to a forty-first aspect that can be understood based on the present disclosure may be an audio processing device according to the fortieth aspect, in which the circuit adjusts the volume ratio in accordance with information related to additional processing, and controls whether or not to select reflected sound based on the adjusted volume ratio.

[0105] The device according to the above aspect can appropriately control whether or not to select reflected sounds based on the volume ratio that reflects the influence of additional processing.

[0106] An audio processing device according to a 42nd aspect that can be understood based on the present disclosure may be an audio processing device according to the 40th or 41st aspect, in which the circuit calculates the time at which a sound emitted from a sound source arrives as a reflected sound, and controls whether or not to select the reflected sound based on the time, information related to additional processing, and a volume ratio.

[0107] The device of the above aspect can more appropriately select reflected sounds that have a greater impact on the listener's perception based on the arrival time of the reflected sounds, information related to additional processing, and the volume ratio between the direct sound and the reflected sounds.

[0108] A sound processing device according to a 43rd aspect as understood based on the present disclosure may be a sound processing device according to any one of the 40th to 42nd aspects, wherein the circuit calculates the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on sound space information, and controls whether or not to select the reflected sound based on the time difference, information related to additional processing, and the volume ratio.

[0109] The device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the time difference between the arrival time of the direct sound and the arrival time of the reflected sound, information related to additional processing, and the volume ratio between the direct sound and the reflected sound. Therefore, the device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the precedence effect.

[0110] A sound processing device according to a 44th aspect as understood based on the present disclosure may be a sound processing device according to any one of the 40th to 42nd aspects, wherein the circuit calculates the time difference between the end time of the direct sound and the arrival time of the reflected sound based on sound space information, and controls whether or not to select the reflected sound based on the time difference, information related to additional processing, and the volume ratio.

[0111] The device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the time difference between the end time of the direct sound and the arrival time of the reflected sound, information related to additional processing, and the volume ratio between the direct sound and the reflected sound. Therefore, the device of the above aspect can more appropriately select reflected sounds that have a large influence on the listener's perception based on the post-masking effect.

[0112] A sound processing device according to a 45th aspect that can be understood based on the present disclosure may be a sound processing device according to any one of the 40th to 44th aspects, in which the additional processing is processing that involves increasing or decreasing the amplitude of the reflected sound, the information related to the additional processing is information indicating the amount of increase or decrease in amplitude in the processing that involves increasing or decreasing the amplitude of the reflected sound, and the circuit adjusts the volume ratio according to the amount of increase or decrease in amplitude, and controls whether or not to select the reflected sound based on the adjusted volume ratio.

[0113] The device of the above aspect can appropriately adjust the volume ratio according to the amplitude that increases or decreases due to the additional processing, and can appropriately control whether or not to select reflected sounds based on the volume ratio adjusted by the additional processing.

[0114] A sound processing device according to a 46th aspect that can be understood based on the present disclosure may be a sound processing device according to the 43rd or 44th aspect, in which the additional processing is processing that involves increasing or decreasing the delay of reflected sound, the information related to the additional processing is information indicating the amount of increase or decrease in delay in the processing that involves increasing or decreasing the delay of reflected sound, and the circuit adjusts the time difference according to the amount of increase or decrease in delay, and controls whether or not to select reflected sound based on the adjusted time difference and volume ratio.

[0115] The device of the above aspect can appropriately adjust the time difference according to the delay that increases or decreases due to the additional processing, and can appropriately control whether or not to select reflected sounds based on the time difference adjusted by the additional processing.

[0116] The sound processing device according to the 47th aspect as understood based on the present disclosure may be a sound processing device that is any of the sound processing devices according to the 40th to 46th aspects, in which the additional processing is processing that is not applied to direct sound.

[0117] The device of the above aspect can impart additional processing to reflected sound without imparting additional processing to direct sound. That is, the device of the above aspect can impart additional processing to reflected sound that is not imparted to direct sound. Therefore, the device of the above aspect can impart additional processing suitable for reflected sound to reflected sound.

[0118] An audio processing device according to a 48th aspect as understood based on the present disclosure may be an audio processing device according to any one of the 40th to 47th aspects, in which the additional processing is a filter processing that auditorily emphasizes the diffusion of sound.

[0119] The device of the above aspect can appropriately control whether or not to select reflected sound based on information regarding filter processing that auditorily emphasizes the diffusion of sound and the volume ratio between direct sound and reflected sound.

[0120] A sound processing device according to a 49th aspect as understood based on the present disclosure may be a sound processing device according to the 41st or 45th aspect, which does not adjust the volume ratio when the information regarding the additional processing indicates that the additional processing will not be performed.

[0121] The device of the above aspect can appropriately control whether to adjust the volume ratio depending on whether to execute additional processing.

[0122] A sound processing device according to a 50th aspect as understood based on the present disclosure may be a sound processing device according to the 46th aspect, which does not adjust the time difference when the information regarding the additional processing indicates that the additional processing will not be performed.

[0123] The device of the above aspect can appropriately control whether to adjust the time difference depending on whether to execute additional processing.

[0124] A sound processing device according to a fifty-first aspect grasped based on the present disclosure may be a sound processing device that includes a circuit and a memory, in which the circuit uses the memory to acquire sound space information relating to a sound space, and determines whether or not a sound source in the sound space is a point sound source based on the sound space information, and if it is determined that the sound source is a point sound source, (i) acquires, based on the sound space information, a volume ratio between the volume of a direct sound generated from the sound source and the volume of a reflected sound generated in the sound space corresponding to the direct sound, (ii) calculates, based on the sound space information, a time difference between the arrival time of the direct sound and the arrival time of the reflected sound, and (iii) controls whether or not to select a reflected sound based on the volume ratio and the time difference, and if it is determined that the sound source is not a point sound source, selects a reflected sound regardless of the volume ratio and the time difference.

[0125] The device of the above aspect can select reflected sounds regardless of the volume ratio and time difference, even if the distance between the listener and the sound source is not determined and it is difficult to appropriately calculate the volume ratio and time difference between the direct sound and the reflected sound. Therefore, the device of the above aspect can suppress the deterioration of sound quality that can occur when the sound source is not a point sound source.

[0126] A threshold identification device according to a fifty-second aspect grasped based on the present disclosure includes a circuit and a memory, and the circuit uses the memory to acquire the time difference between the arrival time of a preceding sound and the arrival time of a following sound that occurs in response to the preceding sound. Based on the time difference, the circuit identifies a threshold for the volume ratio of the volume of the following sound to the volume of the preceding sound, which corresponds to the boundary between whether or not the following sound is affected by the preceding sound effect exerted by the preceding sound, and outputs the threshold. In identifying the threshold, in an area having a first axis of time difference and a second axis of volume ratio, the volume ratio corresponding to the time difference is identified as the threshold on a boundary line defined as a line indicating the boundary, and the domain of the boundary line relative to the first axis is the time range in which the preceding sound effect occurs.

[0127] The device of the above aspect can appropriately identify a threshold value corresponding to the boundary of the precedence effect based on the relationship between the time difference between the preceding sound and the following sound and the volume ratio between the preceding sound and the following sound. Furthermore, the device of the above aspect can efficiently identify a threshold value for the time range in which the precedence effect occurs.

[0128] A threshold determination device according to a 53rd aspect of the present disclosure may be a threshold determination device according to the 52nd aspect, in which, when the first axis is the horizontal axis and the second axis is the vertical axis, the boundary line is a curve that slopes downward to the right and is convex upward in the domain.

[0129] The device of the above aspect can appropriately specify the threshold based on boundary characteristics that the larger the time difference, the smaller the volume ratio and the larger the rate of decrease in the volume ratio with respect to an increase in the time difference, thereby providing an appropriate threshold for determining whether a subsequent sound is affected by the precedence effect.

[0130] A threshold determination device according to a 54th aspect as understood based on the present disclosure may be a threshold determination device according to the 52nd or 53rd aspect, in which the boundary line is defined by a mathematical formula having a time difference as a variable, and the threshold is calculated based on the mathematical formula.

[0131] The device of the above aspect can efficiently specify the threshold value based on a mathematical formula, and therefore can efficiently provide the threshold value.

[0132] An acoustic processing method according to a fifty-fifth aspect grasped based on the present disclosure includes the steps of acquiring sound space information relating to a sound space, acquiring a volume ratio between the volume of a direct sound generated from a sound source in the sound space and the volume of a reflected sound generated in the sound space corresponding to the direct sound based on the sound space information, acquiring information relating to additional processing, and controlling whether or not to select a reflected sound based on the information relating to the additional processing and the volume ratio.

[0133] The method of the above aspect can achieve the same effect as the sound processing device described in the fortieth aspect.

[0134] A program according to a fifty-sixth aspect grasped based on the present disclosure is a program for causing a computer to execute the acoustic processing method of the fifty-fifth aspect.

[0135] The program of the above aspect can achieve the same effect as the sound processing method of the 55th aspect when used with a computer.

[0136] These comprehensive or specific aspects may be realized as a system, an apparatus, a method, an integrated circuit, a computer program, or a computer-readable recording medium such as a CD-ROM, or may be realized as any combination of a system, an apparatus, a method, an integrated circuit, a computer program, or a recording medium.

[0137] The sound processing device, encoding device, decoding device, and stereophonic reproduction system according to the present disclosure will be described in detail below with reference to the drawings. The stereophonic reproduction system may also be expressed as an audio signal reproduction system.

[0138] Note that the embodiments described below are all comprehensive or specific examples. The numerical values, shapes, materials, components, component placement and connection configurations, steps, and step sequences shown in the following embodiments are merely examples and are not intended to limit the aspects understood based on the present disclosure. Furthermore, among the components in the following embodiments, for example, components not included in the basic aspects described in the present disclosure or components not described in the independent claims showing the highest concepts will be described as optional components.

[0139] (Embodiment) (Example of a stereophonic sound reproduction system) Fig. 2 is a diagram showing an example of a stereophonic sound reproduction system. Specifically, Fig. 2 shows a stereophonic sound reproduction system 1000, which is an example of a system to which the acoustic processing or decoding processing of the present disclosure can be applied. Stereophonic sound is also expressed as immersive audio. The stereophonic sound reproduction system 1000 includes an audio signal processing device 1001 and an audio presentation device 1002.

[0140] The audio signal processing device 1001, also referred to as an audio processing device, performs audio processing on an audio signal emitted by a virtual sound source to generate an audio signal after the audio processing to be presented to a listener. The audio signal is not limited to a voice, and may be any audible sound. The audio processing is, for example, signal processing performed on the audio signal to reproduce one or more effects that the sound undergoes from the time it is generated by the sound source until it reaches the listener.

[0141] The audio signal processing device 1001 performs acoustic processing based on spatial information that describes factors that cause the above-mentioned effects. The spatial information includes, for example, information indicating the positions of a sound source, a listener, and surrounding objects, information indicating the shape of a space, and parameters related to sound propagation. The audio signal processing device 1001 is, for example, a PC (Personal Computer), a smartphone, a tablet, a game console, or the like.

[0142] The signal after acoustic processing is presented to the listener from the audio presentation device 1002. The audio presentation device 1002 is connected to the audio signal processing device 1001 via wireless or wired communication. The audio signal after acoustic processing generated by the audio signal processing device 1001 is transmitted to the audio presentation device 1002 via wireless or wired communication.

[0143] When the audio presentation device 1002 is configured with a plurality of devices, such as a device for the right ear and a device for the left ear, the plurality of devices present sounds in synchronization through communication between the plurality of devices or communication between each of the plurality of devices and the audio signal processing device 1001. The audio presentation device 1002 is, for example, headphones, earphones, or a head-mounted display worn on the head of a listener, or a surround speaker configured with a plurality of fixed speakers.

[0144] The stereophonic sound reproduction system 1000 may be used in combination with an image presentation device or a stereoscopic video presentation device that provides a visual ER experience, including AR / VR. For example, the space handled by the spatial information is a virtual space, and the positions of a sound source, a listener, and an object in the space are the virtual positions of a virtual sound source, a virtual listener, and a virtual object in the virtual space. The space may also be expressed as a sound space. The spatial information may also be expressed as sound space information.

[0145] 2 shows an example of a system configuration in which the audio signal processing device 1001 and the audio presentation device 1002 are separate devices, the stereophonic sound reproduction system 1000 to which the audio processing method or decoding method of the present disclosure can be applied is not limited to the configuration shown in Fig. 2. For example, the audio signal processing device 1001 may be included in the audio presentation device 1002, which may perform both audio processing and sound presentation.

[0146] The acoustic processing described in the present disclosure may be shared between the audio signal processing device 1001 and the audio presentation device 1002. A server connected to the audio signal processing device 1001 or the audio presentation device 1002 via a network may perform part or all of the acoustic processing described in the present disclosure.

[0147] Furthermore, the audio signal processing device 1001 may perform audio processing by decoding a bit stream generated by encoding at least a portion of the data of the audio signal and spatial information used for the audio processing. Therefore, the audio signal processing device 1001 may be referred to as a decoding device.

[0148] (Example of Encoding Device) Fig. 3A is a block diagram showing an example configuration of an encoding device. Specifically, Fig. 3A shows the configuration of an encoding device 1100, which is an example of an encoding device of the present disclosure.

[0149] Input data 1101 is data to be coded, including spatial information and / or an audio signal, that is input to an encoder 1102. Details of the spatial information will be explained later.

[0150] The encoder 1102 encodes the input data 1101 to generate encoded data 1103. The encoded data 1103 is, for example, a bit stream generated by the encoding process.

[0151] The memory 1104 stores the encoded data 1103. The memory 1104 may be, for example, a hard disk or a solid-state drive (SSD), or may be other memory.

[0152] In the above description, a bitstream generated by an encoding process is given as an example of the encoded data 1103 stored in memory 1104, but the encoded data 1103 may be data other than a bitstream. For example, the encoding device 1100 may store converted data generated by converting a bitstream into a predetermined data format in memory 1104. The converted data may be, for example, a file or a multiplexed stream corresponding to one or more bitstreams.

[0153] Here, the file is a file having a file format such as ISO Base Media File Format (ISOBMFF), etc. The encoded data 1103 may be in the form of a plurality of packets generated by dividing the bit stream or file.

[0154] For example, the bitstream generated by the encoder 1102 may be converted into data different from the bitstream. In this case, the encoding device 1100 may include a conversion unit (not shown) and perform the conversion process in the conversion unit, or may perform the conversion process in a CPU (Central Processing Unit), which is an example of a processor described later.

[0155] (Example of Decoding Device) Fig. 3B is a block diagram showing an example configuration of a decoding device. Specifically, Fig. 3B shows the configuration of a decoding device 1110, which is an example of a decoding device according to the present disclosure.

[0156] The memory 1114 stores, for example, the same data as the coded data 1103 generated by the coding device 1100. The stored data is read from the memory 1114 and input to the decoder 1112 as input data 1113. The input data 1113 is, for example, a bitstream to be decoded. The memory 1114 may be, for example, a hard disk or an SSD, or may be some other memory.

[0157] Note that the decoding device 1110 may convert the data read from the memory 1114 and input the converted data to the decoder 1112 as input data 1113, rather than inputting the data directly to the decoder 1112 as input data 1113. The data before conversion may be, for example, multiplexed data including one or more bitstreams. Here, the multiplexed data may be a file having a file format such as ISOBMFF.

[0158] The data before conversion may also be a plurality of packets generated by dividing the bitstream or file. Data different from the bitstream may be read from memory 1114 and converted into a bitstream. In this case, decoding device 1110 may include a conversion unit (not shown) and perform the conversion process, or a CPU (an example of a processor, described later) may perform the conversion process.

[0159] Decoder 1112 decodes input data 1113 to produce an audio signal 1111 representing the audio to be presented to the listener.

[0160] (Another Example of Encoding Device) Fig. 3C is a block diagram showing another example of the configuration of an encoding device. Specifically, Fig. 3C shows the configuration of encoding device 1120, which is another example of an encoding device of the present disclosure. In Fig. 3C, the same components as those in Fig. 3A are assigned the same reference numerals as those in Fig. 3A, and descriptions of these components will be omitted.

[0161] Coding device 1100 stores coded data 1103 in memory 1104. On the other hand, coding device 1120 differs from coding device 1100 in that coding device 1120 includes a transmitting unit 1121 that transmits coded data 1103 to the outside.

[0162] The transmitter 1121 transmits to another device or a server a transmission signal 1122 generated based on the encoded data 1103 or data converted into another data format from the encoded data 1103. The data used to generate the transmission signal 1122 is, for example, the bit stream, multiplexed data, file, or packet described in the encoding device 1100.

[0163] (Another Example of Decoding Device) Fig. 3D is a block diagram showing another example of the configuration of a decoding device. Specifically, Fig. 3D shows the configuration of a decoding device 1130, which is another example of a decoding device of the present disclosure. In Fig. 3D, the same components as those in Fig. 3B are assigned the same reference numerals as those in Fig. 3B, and descriptions of these components will be omitted.

[0164] The decoding device 1110 reads input data 1113 from a memory 1114. On the other hand, the decoding device 1130 differs from the decoding device 1110 in that it includes a receiving unit 1131 that receives the input data 1113 from an external source.

[0165] The receiving unit 1131 receives a received signal 1132, acquires received data, and outputs input data 1113 to be input to the decoder 1112. The received data may be the same as the input data 1113 to be input to the decoder 1112, or may be data in a data format different from that of the input data 1113.

[0166] If the data format of the received data is different from the data format of the input data 1113, the receiving unit 1131 may convert the received data into the input data 1113. Alternatively, a conversion unit or a CPU (not shown) of the decoding device 1130 may convert the received data into the input data 1113. The received data is, for example, a bit stream, multiplexed data, a file, or a packet, as described in the encoding device 1120.

[0167] (Example of Decoder) Fig. 4A is a block diagram showing an example of the configuration of a decoder. Specifically, Fig. 4A shows the configuration of a decoder 1200, which is an example of the decoder 1112 in Fig. 3B or 3D.

[0168] The input data 1113 is an encoded bitstream, and includes encoded audio data, which is an encoded audio signal, and metadata used in acoustic processing.

[0169] The spatial information management unit 1201 acquires and analyzes metadata included in the input data 1113. The metadata includes information describing elements that act on sounds arranged in a sound space. The spatial information management unit 1201 manages spatial information used for acoustic processing obtained by analyzing the metadata, and provides the spatial information to the rendering unit 1203.

[0170] In the present disclosure, the information used for acoustic processing is expressed as spatial information, but other expressions may be used. For example, the information used for acoustic processing may be expressed as sound space information or scene information. Furthermore, when the information used for acoustic processing changes over time, the spatial information input to the rendering unit 1203 may be information expressed as a spatial state, a sound space state, a scene state, or the like.

[0171] Furthermore, the spatial information may be managed for each sound space or each scene. For example, when a plurality of different rooms are represented as virtual spaces, the rooms may be managed as a plurality of different scenes. Furthermore, the spatial information may be managed as different scenes depending on the situation represented in the same space.

[0172] Therefore, a plurality of pieces of spatial information may be managed for a plurality of sound spaces or a plurality of scenes. In managing the plurality of pieces of spatial information, an identifier for identifying each piece of spatial information may be assigned to the spatial information.

[0173] The spatial information data may be included in a bitstream, which is an example of input data 1113. Alternatively, the bitstream may include an identifier of the spatial information, and the spatial information data may be acquired from an information source other than the bitstream. Specifically, when the bitstream includes only the identifier of the spatial information, the identifier of the spatial information may be used in rendering to acquire the spatial information data stored in a memory within the device or an external server as input data 1113.

[0174] It should be noted that the information managed by the spatial information management unit 1201 is not limited to information included in the bitstream. For example, the input data 1113 may include data that is not included in the bitstream and indicates the characteristics and structure of a space acquired from software or a server that provides VR or AR.

[0175] The input data 1113 may also include data indicating the characteristics and positions of listeners or objects, etc. The input data 1113 may also include information about the positions of listeners acquired by sensors provided in the terminal including the decoding device (1110, 1130), or may include information indicating the position of the terminal estimated based on the information acquired by the sensors.

[0176] That is, the spatial information management unit 1201 may communicate with an external system or server to acquire spatial information and listener positions. The spatial information management unit 1201 may also acquire clock synchronization information from the external system and execute processing to synchronize with the clock of the rendering unit 1203.

[0177] Note that the space in the above description may be a virtually formed space, i.e., a VR space, or may be a real space or a virtual space corresponding to a real space, i.e., an AR space or an MR space. The virtual space may also be expressed as a sound field or a sound space. Furthermore, the information indicating a position in the above description may be information such as coordinate values ​​indicating a position within a space, information indicating a relative position with respect to a predetermined reference position, or information indicating the movement or acceleration of a position within a space.

[0178] The audio data decoder 1202 decodes the encoded audio data included in the input data 1113 to obtain an audio signal.

[0179] The encoded audio data acquired by the stereophonic sound reproduction system 1000 is a bitstream encoded in a predetermined format such as MPEG-H 3D Audio (ISO / IEC 23008-3). Note that MPEG-H 3D Audio is merely one example of an encoding method that can be used to generate the encoded audio data contained in the bitstream. The encoded audio data may also be a bitstream encoded using another encoding method.

[0180] For example, the encoding method may be a lossy codec such as MP3 (MPEG-1 Audio Layer-3), AAC (Advanced Audio Coding), WMA (Windows Media Audio), AC3 (Audio Codec-3), or Vorbis. Alternatively, the encoding method may be a lossless codec such as ALAC (Apple Lossless Audio Codec) or FLAC (Free Lossless Audio Codec).

[0181] Alternatively, any other encoding method may be used. For example, PCM data may be a type of encoded audio data. In this case, the decoding process may be, for example, a process of converting an N-bit binary number into a number format (e.g., floating-point format) that can be processed by the rendering unit 1203, where the number of quantization bits of the PCM data is N.

[0182] The rendering unit 1203 acquires the audio signal and spatial information, performs acoustic processing on the audio signal using the spatial information, and outputs the audio signal after the acoustic processing (audio signal 1111).

[0183] Before starting rendering, the spatial information management unit 1201 reads metadata of the input signal, detects rendering items such as objects and sounds defined in the spatial information, and transmits them to the rendering unit 1203. After starting rendering, the spatial information management unit 1201 grasps changes over time in the spatial information and the listener's position, updates and manages the spatial information, and transmits the updated spatial information to the rendering unit 1203.

[0184] The rendering unit 1203 generates and outputs an audio signal to which acoustic processing has been applied, based on the audio signal included in the input data 1113 and the spatial information received from the spatial information management unit 1201 .

[0185] The spatial information update process and the audio signal output process with added acoustic processing may be executed in the same thread. Furthermore, the spatial information management unit 1201 and the rendering unit 1203 may each allocate their processes to independent threads. When the spatial information management unit 1201 and the rendering unit 1203 execute the spatial information update process and the audio signal output process with added acoustic processing in different threads, they may set the thread startup frequency individually, or may execute the processes in parallel.

[0186] When the spatial information management unit 1201 and the rendering unit 1203 execute processes in different independent threads, it is possible to allocate computing resources preferentially to the rendering unit 1203. This makes it possible to safely execute sound output processing in which even the slightest delay is unacceptable, for example, in which a delay of one sample (0.02 msec) would cause a popping noise.

[0187] In this case, the allocation of computational resources to the spatial information management unit 1201 is limited. However, because updating of spatial information is a process that occurs less frequently than output processing of audio signals (for example, a process such as updating the direction of the listener's face), it does not necessarily have to be performed instantaneously like output processing of audio signals. Therefore, even if the allocation of computational resources is limited, it does not have a significant impact on acoustic quality.

[0188] The spatial information may be updated periodically at preset times or intervals, or when preset conditions are met. The spatial information may also be updated manually by a listener or a sound space manager, or may be updated in response to a change in an external system.

[0189] For example, the spatial information may be updated when a listener operates a controller to instantly warp the position of his / her avatar or instantly advance or reverse the time. Alternatively, the spatial information may be updated when an administrator of the virtual space suddenly changes the environment of the space. In these cases, the thread for updating the spatial information managed by the spatial information management unit 1201 may be started as a one-off interrupt process in addition to being started periodically.

[0190] Fig. 4B is a block diagram showing another example of the configuration of a decoder. Specifically, Fig. 4B shows the configuration of a decoder 1210, which is another example of the decoder 1112 in Fig. 3B or 3D.

[0191] Figure 4B differs from Figure 4A in that the input data 1113 includes an unencoded audio signal rather than encoded audio data. The input data 1113 includes a bitstream including metadata and an audio signal.

[0192] The spatial information management unit 1211 is the same as the spatial information management unit 1201 in FIG. 4A, and therefore a description thereof will be omitted.

[0193] The rendering unit 1213 is the same as the rendering unit 1203 in FIG. 4A, and therefore a description thereof will be omitted.

[0194] The decoders 1112, 1200, and 1210 may be expressed as audio processing units that perform audio processing. The decoding devices 1110 and 1130 may be the audio signal processing devices 1001, and may be expressed as audio processing devices.

[0195] (Physical configuration of audio signal processing device) Fig. 5 is a diagram showing an example of the physical configuration of the audio signal processing device 1001. Note that the audio signal processing device 1001 in Fig. 5 may be the decoding device 1110 in Fig. 3B or the decoding device 1130 in Fig. 3D. The multiple components shown in Fig. 3B or Fig. 3D may be implemented by the multiple components shown in Fig. 5. Furthermore, part of the configuration described here may be provided in the audio presentation device 1002.

[0196] The audio signal processing device 1001 in FIG. 5 includes a processor 1402 , a memory 1404 , a communication IF (Interface) 1403 , a sensor 1405 , and a speaker 1401 .

[0197] The processor 1402 is, for example, a CPU, a DSP (Digital Signal Processor), or a GPU (Graphics Processing Unit). The CPU, DSP, or GPU may perform the acoustic processing or decoding processing of the present disclosure by executing a program stored in the memory 1404. The processor 1402 is, for example, a circuit that performs information processing. The processor 1402 may also be a dedicated circuit that performs signal processing on audio signals, including the acoustic processing of the present disclosure.

[0198] The memory 1404 is configured, for example, with a RAM (Random Access Memory) or a ROM (Read Only Memory). The memory 1404 may include a magnetic recording medium such as a hard disk or a semiconductor memory such as an SSD. The memory 1404 may also be an internal memory incorporated in the CPU or GPU. The memory 1404 may also store spatial information managed by the spatial information management units (1201, 1211). Threshold data, which will be described later, may also be stored.

[0199] The communication IF 1403 is a communication module compatible with a communication method such as Bluetooth (registered trademark) or WIGIG (registered trademark). The audio signal processing device 1001 communicates with another communication device via the communication IF 1403, for example, to acquire a bitstream to be decoded. The acquired bitstream is stored in the memory 1404, for example.

[0200] The communication IF 1403 is configured with, for example, a signal processing circuit and an antenna corresponding to a communication method. The communication method is not limited to Bluetooth (registered trademark) and WIGIG (registered trademark), but may also be LTE (Long Term Evolution), NR (New Radio), Wi-Fi (registered trademark), or the like.

[0201] Furthermore, the communication method is not limited to the wireless communication method described above, but may be a wired communication method such as Ethernet (registered trademark), USB (Universal Serial Bus), or HDMI (registered trademark) (High-Definition Multimedia Interface).

[0202] The sensor 1405 performs sensing to estimate the position and orientation of the listener. Specifically, the sensor 1405 estimates the position and / or orientation of the listener based on one or more detection results of the position, orientation, movement, velocity, angular velocity, acceleration, etc. of a part or the whole of the body, and generates position / or orientation information indicating the position and / or orientation of the listener.

[0203] Note that a device external to the audio signal processing device 1001 may be equipped with the sensor 1405. The part of the body may be the listener's head, etc. The position / orientation information may be information indicating the position and / or orientation of the listener in real space, or information indicating a displacement of the position and / or orientation of the listener based on the position and / or orientation of the listener at a predetermined time. Furthermore, the position / or orientation information may be information indicating a position and / or orientation relative to the stereophonic sound reproduction system 1000 or an external device equipped with the sensor 1405.

[0204] The sensor 1405 is, for example, an imaging device such as a camera or a ranging device such as a LiDAR (Laser Imaging Detection and Ranging). The sensor 1405 may capture an image of the listener's head movement and detect the head movement by processing the captured image. Alternatively, the sensor 1405 may be a device that performs position estimation using a wireless signal of any frequency band, such as a millimeter wave.

[0205] Furthermore, the audio signal processing device 1001 may acquire position information from an external device equipped with a sensor 1405 via the communication IF 1403. In this case, the audio signal processing device 1001 may not include the sensor 1405. Here, the external device is, for example, the audio presentation device 1002 described in Fig. 2 or a 3D video playback device worn on the head of a listener. In this case, the sensor 1405 is configured by combining various sensors such as a gyro sensor and an acceleration sensor.

[0206] For example, the sensor 1405 may detect the angular velocity of rotation around at least one of three mutually orthogonal axes in the sound space as the axis of rotation as the speed of movement of the listener's head, or may detect the acceleration of displacement with at least one of the three axes as the direction of displacement.

[0207] For example, the sensor 1405 may detect the amount of rotation about at least one of three mutually orthogonal axes in the sound space as the rotation axis, or the amount of displacement about at least one of the three axes as the displacement direction, as the amount of movement of the listener's head. Specifically, the sensor 1405 detects the 6 DoF positions (x, y, z) and angles (yaw, pitch, roll) as the position of the listener. The sensor 1405 is configured by combining various sensors used for detecting movement, such as a gyro sensor and an acceleration sensor.

[0208] The sensor 1405 may be realized by a camera for detecting the position of the listener, a GPS (Global Positioning System) receiver, or the like. Position information obtained by performing self-position estimation using a LiDAR or the like as the sensor 1405 may also be used. For example, when the stereophonic sound reproduction system 1000 is realized by a smartphone, the sensor 1405 is built into the smartphone.

[0209] The sensor 1405 may also include a temperature sensor such as a thermocouple that detects the temperature of the audio signal processing device 1001. The sensor 1405 may also include a sensor that detects the remaining charge of a battery provided in the audio signal processing device 1001 or a battery connected to the audio signal processing device 1001.

[0210] The speaker 1401 has, for example, a diaphragm, a drive mechanism such as a magnet or a voice coil, and an amplifier, and presents an audio signal after acoustic processing as sound to a listener. The speaker 1401 operates the drive mechanism in response to an audio signal (more specifically, a waveform signal indicating the waveform of the sound) amplified via the amplifier, and the drive mechanism vibrates the diaphragm. In this way, the diaphragm vibrating in response to the audio signal generates sound waves, which propagate through the air to the listener's ears, causing the listener to perceive the sound.

[0211] Here, an example has been given in which the audio signal processing device 1001 is equipped with a speaker 1401 and presents an audio signal after acoustic processing via the speaker 1401, but the means for presenting the audio signal is not limited to the above configuration.

[0212] For example, the audio signal after acoustic processing may be output to an external audio presentation device 1002 connected via a communication module. Communication via the communication module may be wired or wireless. As another example, the audio signal processing device 1001 may have a terminal for outputting an analog audio signal, and a cable for earphones or the like may be connected to the terminal to present the audio signal from the earphones or the like.

[0213] In the above case, the audio presentation device 1002 may be headphones, earphones, a head-mounted display, a neck speaker, a wearable speaker, or the like that are worn on the head or part of the body of the listener. Alternatively, the audio presentation device 1002 may be a surround speaker or the like that is composed of multiple fixed speakers. The audio presentation device 1002 may then reproduce an audio signal.

[0214] (Physical Configuration of Encoding Apparatus) Fig. 6 is a diagram showing an example of the physical configuration of an encoding apparatus. Encoding apparatus 1500 in Fig. 6 may be encoding apparatus 1100 in Fig. 3A or encoding apparatus 1120 in Fig. 3C, and multiple components shown in Fig. 3A or 3C may be implemented by multiple components shown in Fig. 6.

[0215] The encoding device 1500 in FIG. 6 includes a processor 1501 , a memory 1503 , and a communication IF 1502 .

[0216] The processor 1501 is, for example, a CPU, a DSP, or a GPU. The CPU, DSP, or GPU may perform the encoding process of the present disclosure by executing a program stored in the memory 1503. The processor 1501 is, for example, a circuit that performs information processing. The processor 1501 may be a dedicated circuit that performs signal processing on an audio signal, including the encoding process of the present disclosure.

[0217] The memory 1503 is configured with, for example, a RAM or a ROM. The memory 1503 may include a magnetic recording medium such as a hard disk or a semiconductor memory such as an SSD. The memory 1503 may also be an internal memory incorporated in the CPU or GPU.

[0218] The communication IF 1502 is a communication module compatible with a communication method such as Bluetooth (registered trademark) or WIGIG (registered trademark). The encoding device 1500 communicates with another communication device via the communication IF 1502, for example, and transmits an encoded bitstream.

[0219] The communication IF 1502 is configured with, for example, a signal processing circuit and an antenna corresponding to the communication method. The communication method is not limited to Bluetooth (registered trademark) and WIGIG (registered trademark), but may be LTE, NR, Wi-Fi (registered trademark), or the like. Furthermore, the communication method is not limited to a wireless communication method. The communication method may be a wired communication method such as Ethernet (registered trademark), USB, or HDMI (registered trademark).

[0220] (Configuration of Rendering Unit) Fig. 7 is a block diagram showing an example of the configuration of the rendering unit. Specifically, Fig. 7 shows an example of the detailed configuration of a rendering unit 1300 corresponding to the rendering units 1203 and 1213 in Figs. 4A and 4B.

[0221] The rendering unit 1300 is composed of an analysis unit 1301, a selection unit 1302, and a synthesis unit 1303, and applies acoustic processing to the sound data contained in the input signal and outputs the result.

[0222] The input signal may be composed of, for example, spatial information, sensor information, and sound data. The input signal may also include a bitstream composed of sound data and metadata (control information), in which case the metadata may include spatial information.

[0223] The spatial information is information about the sound space (three-dimensional sound field) created by the stereophonic sound reproduction system 1000, and is composed of information about objects included in the sound space and information about the listener. Objects include sound source objects that emit sound and act as sound sources, and non-sound-emitting objects that do not emit sound. Sound source objects can also be simply referred to as sound sources.

[0224] A non-sound-emitting object acts as an obstacle object that reflects the sound emitted by a sound source object, but a sound source object may also act as an obstacle object that reflects the sound emitted by another sound source object. Obstacle objects may also be referred to as reflecting objects.

[0225] Information commonly assigned to sound source objects and non-sound generating objects includes position information, shape information, and the rate of attenuation of the volume when the object reflects sound.

[0226] The position information is expressed as coordinate values ​​on three axes, for example, the X-axis, Y-axis, and Z-axis, in Euclidean space, but does not necessarily have to be three-dimensional information. For example, the position information may be two-dimensional information expressed as coordinate values ​​on two axes, the X-axis and the Y-axis. The position information of an object is determined by a representative position of a shape expressed by a mesh or voxels.

[0227] The shape information may include information about the surface material.

[0228] The attenuation rate may be expressed as a real number between 0 and 1, or may be expressed as a negative decibel value. In real space, the volume is not amplified by reflection, so a negative decibel value is set as the attenuation rate, but for example, to create an eerie feeling in an unreal space, an attenuation rate of 1 or more, i.e., a positive decibel value, may be set.

[0229] The attenuation rate may be set to a different value for each of the frequency bands constituting the plurality of frequency bands, or may be set independently for each frequency band. Furthermore, if the attenuation rate is set for each type of material on the object surface, a corresponding attenuation rate value may be used based on information about the surface material.

[0230] The spatial information may also include information indicating whether the object belongs to a living thing, information indicating whether the object is a moving object, etc. If the object is a moving object, the position indicated by the position information may move over time. In this case, information on the changed position or the amount of change is transmitted to the rendering unit 1300.

[0231] The information about the sound source object includes information commonly assigned to the sound source object and the non-sound-producing object, as well as sound data and information necessary for radiating the sound data into the sound space. The sound data is data indicating information about the frequency and intensity of the sound, and is data that expresses the sound perceived by a listener.

[0232] The sound data is typically a PCM signal, but may also be data compressed using an encoding method such as MP3. In this case, the signal must be decoded at least before it reaches the synthesis unit 1303, so the rendering unit 1300 may include a decoding unit (not shown). Alternatively, the signal may be decoded by the audio data decoder 1202.

[0233] One piece of sound data may be set for one sound source object, or multiple pieces of sound data may be set for one sound source object. Furthermore, identification information for identifying each piece of sound data may be assigned to the sound data, and the information about the sound source object may include the identification information of the sound data.

[0234] The information necessary to radiate sound data into a sound space may include, for example, information on the reference volume used as a standard for playing back sound data, information indicating the properties (also called characteristics) of the sound data, information on the position of the sound source object, and information on the orientation of the sound source object (i.e., information on the directionality of the sound emitted by the sound source object).

[0235] The reference volume information may be, for example, the effective value of the amplitude value of the sound data at the sound source position when the sound data is emitted into the sound space, and may be expressed as a floating-point decibel (db) value.

[0236] For example, a reference volume of 0 db may indicate that sound is emitted into the sound space from the position indicated by the information regarding the position of the sound source object at the same volume as the signal level indicated by the sound data, without increasing or decreasing the volume.Alternatively, a reference volume of -6 db may indicate that sound is emitted into the sound space from the position indicated by the information regarding the position of the sound source object, with the volume of the signal level indicated by the sound data reduced to approximately half.

[0237] The reference volume information may be attached to each piece of sound data, or may be attached to a plurality of pieces of sound data collectively.

[0238] The information indicating the properties of the sound data may be, for example, information relating to the volume of the sound source, and may be information indicating time-series fluctuations in the volume of the sound source.

[0239] For example, if the sound space is a virtual conference room and the sound source is a speaker, the volume transitions intermittently over a short period of time. That is, sound and silence alternate. If the sound space is a concert hall and the sound source is a performer, the volume is maintained for a certain period of time. If the sound space is a battlefield and the sound source is an explosive, the volume of the explosion will increase for a moment and then remain silent or low.

[0240] In this way, the information on the volume of the sound source may include not only information on the loudness of the sound but also information on the transition of the loudness of the sound. Such information may be used as information indicating the properties of the sound data.

[0241] The transition information may be represented by data indicating frequency characteristics in a time series. The transition information may be represented by data indicating the duration of a sound section. The transition information may be represented by data indicating a time series of the duration of a sound section and the duration of a silent section. The transition information may be represented by data listing, in a time series, multiple pairs of durations during which the amplitude of a sound signal can be considered steady (considered to be roughly constant) and the amplitude values ​​of the signal during those durations.

[0242] The transition information may be expressed as data on the duration for which the frequency characteristics of the sound signal can be considered stationary. The transition information may be expressed as data listing, in time series, multiple pairs of durations for which the frequency characteristics of the sound signal can be considered stationary and the frequency characteristics during those periods. The transition information may be expressed, for example, in the form of data indicating the outline of a spectrogram.

[0243] Furthermore, the volume used as the reference for the frequency characteristics may be the reference volume. Information on the reference volume and information indicating the properties of the sound data may be used in a process of calculating the volume of direct sound or reflected sound to be perceived by the listener, or may be used in a process of selecting whether or not to perceive the direct sound or reflected sound. Other examples and usage methods of the information indicating the properties of the sound data will be described later.

[0244] The information about the sound source object may include, for example, information about the orientation of the sound source object (that is, information about the directivity of the sound emitted by the sound source object).

[0245] Information about the direction of the sound source object (orientation information) is typically expressed using yaw, pitch, and roll. Alternatively, the roll rotation may be omitted, and the direction information of the sound source object may be expressed using azimuth (yaw) and elevation (pitch). The direction information of the sound source object may change over time, and if it changes, it is transmitted to the rendering unit 1300.

[0246] Information about the listener is information about the listener's position and orientation in sound space. The information about the position (position information) is expressed as a position on the XYZ axes in Euclidean space, but it does not necessarily have to be three-dimensional information and may be two-dimensional information. Information about the listener's orientation (orientation information) is typically expressed using yaw, pitch, and roll. Alternatively, the roll rotation may be omitted, and the listener's orientation information may be expressed using azimuth (yaw) and elevation (pitch).

[0247] The position information and orientation information of the listener may change over time, and if so, is transmitted to the rendering unit 1300 .

[0248] The sensor information includes the amount of rotation or displacement detected by a sensor 1405 worn by the listener, as well as the listener's position and orientation. The sensor information is transmitted to the rendering unit 1300, which updates the listener's position and orientation information based on the sensor information. The sensor information may include, for example, position information obtained by a mobile terminal performing self-position estimation using a GPS, a camera, LiDAR, or the like.

[0249] Furthermore, information acquired from outside via a communication module may be detected as sensor information instead of the sensor 1405. Information indicating the temperature of the audio signal processing device 1001 and information indicating the remaining battery capacity may be acquired from the sensor 1405. Furthermore, the computational resources (CPU capacity, memory resources, PC performance, etc.) of the audio signal processing device 1001 or the audio presentation device 1002 may be acquired in real time.

[0250] The analysis unit 1301 analyzes the audio signal contained in the input signal and the spatial information received from the spatial information management unit (1201, 1211), and detects the information necessary to generate direct sound and reflected sound, as well as the information necessary to select whether or not to generate reflected sound.

[0251] The information required to generate direct sound and reflected sound includes, for example, values ​​relating to the path taken by each of the direct sound and reflected sound to reach the listening position, the time it takes for each sound to arrive, and the volume at the time of arrival.

[0252] The information required to select the reflected sound to be output is information indicating the relationship between the direct sound and the reflected sound, such as a value related to the time difference between the direct sound and the reflected sound, and a value related to the volume ratio between the direct sound and the reflected sound at the listening position.

[0253] It goes without saying that when the volume is expressed in decibel units on a logarithmic axis (when the volume is expressed in the decibel domain), the volume ratio of two signals is expressed as the difference in decibel values. Specifically, the volume ratio of two signals may be the difference between the amplitude values ​​of each signal when expressed in the decibel domain. This value may be calculated based on an energy value, a power value, or the like. Furthermore, in the decibel domain, this difference may be referred to as a gain difference or simply a gain difference.

[0254] That is, the volume ratio in the present disclosure is essentially a ratio of signal amplitudes, and may be expressed as a sound volume ratio, a volume ratio, an amplitude ratio, a sound level ratio, a sound intensity ratio, a gain ratio, etc. Furthermore, when the unit of volume is decibels, the volume ratio in the present disclosure can of course be rephrased as a volume difference.

[0255] In the present disclosure, the term "volume ratio" typically refers to the gain difference when the volume of two sounds is expressed in decibel units, and in the example embodiments, the threshold data is also typically defined as a gain difference expressed in the decibel domain. However, the volume ratio is not limited to a gain difference in the decibel domain. When a volume ratio expressed in a domain other than the decibel domain is used, the threshold data defined in the decibel domain may be converted into the unit of the calculated volume ratio and used. Alternatively, threshold data defined in each unit may be stored in advance in memory.

[0256] In other words, it is clear that the algorithm in the present disclosure can be applied to solving the problem of the present disclosure even if a ratio of energy values ​​or power values, for example, is used instead of the volume ratio.

[0257] The time difference between a direct sound and a reflected sound is, for example, the time difference between the arrival time (arrival time) of the direct sound and the arrival time (arrival time) of the reflected sound. The time difference between a direct sound and a reflected sound may be the time difference between the times when the direct sound and the reflected sound arrive at the listening position, the difference in the time it takes for the direct sound and the reflected sound to arrive at the listening position, or the time difference between the time when the direct sound ends and the time when the reflected sound arrives at the listening position. Methods for calculating these values ​​will be described later.

[0258] The selection unit 1302 uses the information calculated by the analysis unit 1301 and the threshold data to select whether or not to generate a reflected sound. In other words, the selection unit 1302 determines whether or not to select a reflected sound as a target reflected sound to be generated. In other words, the selection unit 1302 selects which of the multiple reflected sounds to generate.

[0259] The threshold data is expressed as a boundary (threshold) between whether the reflected sound is perceived or not, for example, on a graph with the value of the time difference between the direct sound and the reflected sound on the horizontal axis and the volume ratio between the direct sound and the reflected sound on the vertical axis. The threshold data may be expressed as an approximation formula having the value of the time difference between the direct sound and the reflected sound as a variable, or may be expressed as an array having the value of the time difference between the direct sound and the reflected sound as an index and a corresponding threshold.

[0260] The selection unit 1302 selects to generate reflected sound when, for example, the volume ratio between the volume of the direct sound at the time of arrival and the volume of the reflected sound at the time difference between the arrival time of the direct sound and the arrival time of the reflected sound is greater than a threshold value set by referring to threshold data.

[0261] The time difference between the arrival time of the direct sound and the arrival time of the reflected sound is, in other words, the difference in the time it takes for the direct sound and the reflected sound to arrive at the listening position. Alternatively, the time difference between the end of the direct sound and the arrival of the reflected sound at the listening position may be used as the time difference between the direct sound and the reflected sound. In this case, threshold data different from the threshold data determined based on the time difference between the arrival time of the direct sound and the arrival time of the reflected sound may be used, or a common threshold data may be used.

[0262] The threshold data may be acquired from the memory 1404 of the audio signal processing device 1001, or may be acquired from an external storage device via a communication module. A method for storing the threshold data and a method for setting the threshold will be described later.

[0263] The synthesis unit 1303 synthesizes the audio signal of the direct sound with the audio signal of the reflected sound that the selection unit 1302 has selected to generate.

[0264] Specifically, the synthesis unit 1303 processes the input audio signal to generate a direct sound based on information about the direct sound arrival time and volume at the time of direct sound arrival calculated by the analysis unit 1301. The synthesis unit 1303 also processes the input audio signal to generate a reflected sound based on information about the reflected sound arrival time and volume at the time of reflected sound arrival for the reflected sound selected by the selection unit 1302. The synthesis unit 1303 then synthesizes and outputs the generated direct sound and reflected sound.

[0265] (Operation of Rendering Unit) Fig. 8 is a flowchart showing an example of operation of the audio signal processing device 1001. Fig. 8 mainly shows processing executed by the rendering unit 1300 of the audio signal processing device 1001.

[0266] In the input signal analysis process (S101 in FIG. 8), the analysis unit 1301 analyzes the input signal input to the audio signal processing device 1001 to detect direct sound and reflected sound that may be generated in the sound space. The reflected sound detected here is a candidate for reflected sound that is selected by the selection unit 1302 as the reflected sound that will ultimately be generated by the synthesis unit 1303. The analysis unit 1301 also analyzes the input signal to calculate information necessary for generating direct sound and reflected sound, and information necessary for selecting the reflected sound to be generated.

[0267] First, the characteristics of each of the direct sound and the reflected sound are calculated. Specifically, the arrival time and volume of each of the direct sound and the reflected sound when they reach the listener are calculated. If multiple objects exist in the sound space as reflecting objects, the characteristics of the reflected sound are calculated for each of the multiple objects.

[0268] The direct sound arrival time (td) is calculated based on the direct sound arrival path (pd). The direct sound arrival path (pd) is a path connecting the position information S (xs, ys, zs) of the sound source object and the position information A (xa, ya, za) of the listener. The direct sound arrival time (td) is a value obtained by dividing the length of the path connecting the position information S (xs, ys, zs) and the position information A (xa, ya, za) by the speed of sound (approximately 340 m / s).

[0269] For example, the path length (X) can be calculated as (xs-xa)^2 + (ys-ya)^2 + (zs-za)^2)^0.5. The volume attenuates in inverse proportion to the distance. Therefore, if the volume of the sound source object at the position information S(xs, ys, zs) is N and the unit distance is U, the volume of the direct sound (ld) when it arrives can be calculated as ld=N*U / X.

[0270] The volume N at the sound source position may be the reference volume described above.

[0271] The reflected sound arrival time (tr) is calculated based on the reflected sound arrival path (pr), which is a path connecting the position of the sound image of the reflected sound and the position information A (xa, ya, za).

[0272] The position of the sound image of the reflected sound may be derived using, for example, the "mirror image method" or "ray tracing method," or any other method for deriving the sound image position. The mirror image method is a method for simulating a sound image by assuming that a mirror image of a wave reflected from a wall in a room exists at a position symmetrical to the sound source with respect to the wall, and that a sound wave is emitted from the position of the mirror image. The ray tracing method is a method for simulating an image (sound image) observed at a certain point by tracing waves that propagate in a straight line, such as light rays or sound rays.

[0273] Fig. 9 is a diagram showing a positional relationship between a listener and an obstacle object that is relatively far away. Fig. 10 is a diagram showing a positional relationship between a listener and an obstacle object that is relatively close. That is, Fig. 9 and Fig. 10 each show an example in which a sound image of a reflected sound is formed at a position symmetrical with respect to the sound source position across a wall. By determining the position of the sound image of the reflected sound on the x, y, and z axes based on this relationship, the arrival time of the reflected sound can be determined in the same way as the method for calculating the arrival time of a direct sound.

[0274] The arrival time of a reflected sound (tr) is a value obtained by dividing the length (Y) of the path connecting the position of the sound image of the reflected sound and the position information A (xa, ya, za) by the speed of sound (approximately 340 m / sec). The volume attenuates inversely proportional to the distance. Therefore, if the volume at the sound source position is N, the unit distance is U, and the rate of attenuation of the volume upon reflection is G, the volume at the time of arrival of the reflected sound (lr) can be calculated as lr = N * G * U / Y.

[0275] As explained above, the attenuation factor G may be expressed as a real number between 0 and 1, or may be expressed as a negative decibel value. In this case, the volume of the entire signal is attenuated by G. The attenuation factor may also be set for each frequency band constituting multiple frequency bands. In this case, the analysis unit 1301 multiplies each frequency component of the signal by a specified attenuation factor. In order to reduce the amount of calculation, the analysis unit 1301 may use a representative value or average value of multiple attenuation factors for multiple frequency bands as the overall attenuation factor, and attenuate the volume of the entire signal by that amount.

[0276] Next, the analysis unit 1301 calculates the volume ratio (L), which is the ratio between the volume at the time of arrival of the direct sound (ld) and the volume at the time of arrival of the reflected sound (lr), and the time difference (T) between the direct sound and the reflected sound, which are necessary for selecting the reflected sound to be generated.

[0277] The volume ratio (L), which is the ratio of the volume (ld) when the direct sound arrives to the above lr, can be calculated, for example, as follows: L = (N * G * U / Y) / (N * U / X) = G * X / Y. Since the value to be calculated is the volume ratio, the values ​​of N and U may be any predetermined values.

[0278] The volume ratio (L) may be adjusted in the following cases. For example, in order to remove unnecessary noise or emphasize specific components, a sound signal may be subjected to filtering, such as passing or attenuating components in a specific frequency band or changing the phase. The filtering may be, for example, a low-pass filter, a high-pass filter, a band-pass filter, or an all-pass filter, or may be other filtering.

[0279] Reflected sounds selected without taking the above-described filtering into consideration may not be appropriate for the reflected sounds to be generated after the filtering process is applied. That is, a newly applied acoustic effect may result in an inappropriate selection of reflected sounds, resulting in a deterioration in quality. Therefore, the volume ratio (L) may be adjusted to match the appropriate selection of reflected sounds to the newly applied acoustic effect.

[0280] Specifically, when a filter process is applied to a sound signal, parameters of the set filter coefficients may be acquired, and the volume ratio (L) may be adjusted according to the parameters. By adjusting the volume ratio (L) according to the parameters of the filter coefficients, it is possible to appropriately select reflected sounds to be output, taking into account the volume ratio changed by the filter process.

[0281] Furthermore, the filtering process may pass or attenuate sounds in a specific frequency band, change the phase, or perform other processing, as described above. For example, the filtering process may pass sounds in all frequency bands, or may only change the phase.

[0282] Specifically, the rendering unit 1300 may include one or more of an adding means (adding unit), an added increase / decrease amount acquisition means (additional increase / decrease amount acquisition unit), a volume ratio adjustment means (volume ratio adjustment unit), an added delay amount acquisition means (additional delay amount acquisition unit), and a time difference adjustment means (time difference adjustment unit).

[0283] For example, the adding means is a filter means (filter unit) that performs filter processing that auditorily emphasizes the phase variation of reflected sound in order to emphasize the spread of sound or the naturalness of reflected sound, as an example of the above-mentioned filter processing.

[0284] The added increment / decrement acquisition means acquires the increment / decrement of amplitude in the adding means. The volume ratio adjustment means adjusts the volume ratio according to the increment / decrement acquired by the added increment / decrement acquisition means. The added delay acquisition means acquires the increment / decrement of delay in the adding means. The time difference adjustment means adjusts the time difference according to the increment / decrement acquired by the added delay acquisition means.

[0285] These means may be implemented by the processor 1402 and the memory 1404, etc.

[0286] For example, a filter (diffusion filter) that auditorily emphasizes the phase variation of a reflected sound may be applied to the reflected sound. When a diffusion filter that significantly varies the phase is applied to the reflected sound, the amplitude of the reflected sound may be amplified. Therefore, a parameter that controls the degree of phase variation of the diffusion filter may be acquired, and the volume ratio (L) may be adjusted by taking into account an increase or decrease amount according to the parameter.

[0287] Furthermore, for example, if a parameter for increasing or decreasing the amplitude of the filter coefficient of the diffusion filter can be set by the user, the set parameter may be acquired, and the amount of increase or decrease in amplitude specified by the parameter may be multiplied by the volume ratio (L).

[0288] The above-mentioned diffusion filter may be called a dispersion filter or a diffusion filter, or may be a decorrelation filter that varies the phase of a sound signal. The above-mentioned diffusion filter may also be a filter used for purposes other than emphasizing the diffusion of sound. For example, the above-mentioned diffusion filter may be a filter that amplifies amplitude, delay, or the like.

[0289] Furthermore, it may be determined whether or not to apply filter processing to the sound signal, and if it is determined that filter processing is to be applied, the volume ratio (L) may be adjusted. In determining whether or not to apply filter processing to the sound signal, for example, the spatial information management unit (1201, 1211) or the adding means may analyze metadata included in the bitstream, and if information included in the metadata indicates that filter processing is effective, determine to apply filter processing.

[0290] Furthermore, it may be determined whether or not to adjust the volume ratio (L) depending on the parameters of the filter coefficients applied to the sound signal.

[0291] Furthermore, the determination of whether to apply filtering and the determination of whether to adjust the volume ratio (L) may be switched for each type of sound. Possible types of sound include direct sound, early reflection sound, nth-order reflection sound, reverberation sound, and diffracted sound. In other words, a determination may be made to apply filtering to early reflection sound and not to apply filtering to diffracted sound.

[0292] The time difference (T) between the direct sound and the reflected sound may be, for example, the time difference between the time it takes for the direct sound and the reflected sound to reach the listening position. For example, the time difference (T) between the time it takes for the direct sound and the reflected sound to reach the listening position can be calculated as T = tr - td.

[0293] The time difference (T) may also be the difference in time between when the direct sound and the reflected sound arrive at the listening position. The time difference (T) may also be the time difference between when the direct sound ends and when the reflected sound arrives at the listening position. In other words, the time difference (T) may be the time difference between when the direct sound ends and when the reflected sound starts at the listening position.

[0294] The time difference (T) may be adjusted in the following cases. For example, in order to remove unnecessary noise or emphasize specific components, a sound signal may be subjected to filtering, such as passing or attenuating components in a specific frequency band or changing the phase. The filtering may be, for example, a low-pass filter, a high-pass filter, a band-pass filter, or an all-pass filter, or may be other filtering.

[0295] Reflected sounds selected without taking the above-described filtering into consideration may not be appropriate for the reflected sounds to be generated after the filtering process is applied. That is, a newly applied acoustic effect may result in an inappropriate selection of reflected sounds, resulting in a deterioration in quality. Therefore, the volume ratio (L) may be adjusted to match the appropriate selection of reflected sounds to the newly applied acoustic effect.

[0296] Specifically, when a filter process is applied to a sound signal, parameters of the set filter coefficients may be acquired, and the time difference (T) may be adjusted according to the parameters. By adjusting the time difference (T) according to the parameters of the filter coefficients, it is possible to appropriately select a reflected sound to be output, taking into account the time difference changed by the filter process.

[0297] Furthermore, the filtering process may pass or attenuate sounds in a specific frequency band, change the phase, or perform other processing, as described above. For example, the filtering process may pass sounds in all frequency bands, or may only change the phase.

[0298] Specifically, the rendering unit 1300 may include one or more of an adding means (adding unit), an added increase / decrease amount acquisition means (additional increase / decrease amount acquisition unit), a volume ratio adjustment means (volume ratio adjustment unit), an added delay amount acquisition means (additional delay amount acquisition unit), and a time difference adjustment means (time difference adjustment unit).

[0299] For example, the adding means is a filter means (filter unit) that performs filter processing that auditorily emphasizes the phase variation of reflected sound in order to emphasize the spread of sound or the naturalness of reflected sound, as an example of the above-mentioned filter processing.

[0300] The added increment / decrement acquisition means acquires the increment / decrement of amplitude in the adding means. The volume ratio adjustment means adjusts the volume ratio according to the increment / decrement acquired by the added increment / decrement acquisition means. The added delay acquisition means acquires the increment / decrement of delay in the adding means. The time difference adjustment means adjusts the time difference according to the increment / decrement acquired by the added delay acquisition means.

[0301] These means may be implemented by the processor 1402 and the memory 1404, etc.

[0302] For example, a filter (diffusion filter) that auditorily emphasizes the phase variation of a reflected sound may be applied to the reflected sound. When a diffusion filter that significantly varies the phase is applied to the reflected sound, the arrival time of the reflected sound (the time it takes to arrive) may increase or decrease. Therefore, parameters that control the diffusion time of the diffusion filter (filter length, order of filter coefficients, number of taps) may be obtained, and the time difference (T) may be adjusted by taking into account the amount of increase or decrease in the arrival time according to the parameters.

[0303] Furthermore, for example, if a parameter for increasing or decreasing the filter length of the diffusion filter can be set by the user, the set parameter may be acquired, and the amount of increase or decrease in the filter length specified by the parameter may be added to the time difference (T).

[0304] The above-mentioned diffusion filter may be called a dispersion filter or a diffusion filter, or may be a decorrelation filter that varies the phase of a sound signal, etc. Furthermore, the above-mentioned diffusion filter may be a filter that is used for purposes other than emphasizing the diffusion of sound.

[0305] Furthermore, it may be determined whether or not to apply filter processing to the audio signal, and if it is determined that filter processing should be applied, the time difference (T) may be adjusted. In determining whether or not to apply filter processing to the audio signal, for example, the spatial information management unit (1201, 1211) or the adding means may analyze metadata included in the bitstream, and if information included in the metadata indicates that filter processing is effective, determine to apply filter processing.

[0306] Furthermore, it may be determined whether or not to adjust the time difference (T) depending on the parameters of the filter coefficients applied to the sound signal.

[0307] Furthermore, the determination of whether to apply filtering and the determination of whether to adjust the time difference (T) may be switched for each type of sound. Possible types of sound include direct sound, early reflection sound, nth-order reflection sound, reverberation sound, and diffracted sound. In other words, a determination may be made to apply filtering to early reflection sound and not to apply filtering to diffracted sound.

[0308] The metadata of the input signal may include various parameters as information related to the filtering process, and this information may be used in the selection process.

[0309] For example, as described in Patent Document 4, the parameter may include a flag for switching on / off filtering, a flag for switching on / off filtering for early reflected sounds, a flag for switching on / off filtering for diffracted sounds, a parameter indicating the duration of filtering, a parameter indicating the gain to be increased or decreased in filtering, or a parameter of angle information indicating spatial spread. The parameter indicating the duration of filtering may indicate the order (number of taps) of the filter coefficient.

[0310] The adding means may analyze metadata included in the input signal and determine whether to execute filter processing. The added increase / decrease amount acquiring means or the added delay amount acquiring means may analyze metadata included in the input signal and acquire parameters of the set filter processing to acquire the increase / decrease amount of amplitude or the increase / decrease amount of delay in the adding means.

[0311] The above-described filtering process is also referred to as an additional process. The additional process may be a process applied only to the reflected sound, a process applied only to the direct sound, or a process applied to both the reflected sound and the direct sound.

[0312] Next, in the reflected sound selection process (S102 in FIG. 8), the selection unit 1302 selects whether or not to generate the reflected sound calculated by the analysis unit 1301. In other words, the selection unit 1302 determines whether or not to select the reflected sound as a target reflected sound to be generated. When there are multiple reflected sounds, the selection unit 1302 selects whether or not to generate each of the reflected sounds. In other words, the selection unit 1302 selects one or more target reflected sounds to be generated from the multiple reflected sounds.

[0313] The selection unit 1302 may select reflected sounds to which other processing is to be applied, not limited to the generation processing. For example, the selection unit 1302 may select reflected sounds to which binaural processing is to be applied. Furthermore, the selection unit 1302 basically selects only one or more reflected sounds to be processed. However, the selection unit 1302 may also select only one or more reflected sounds that are not to be processed. Then, processing may be applied to one or more reflected sounds that are not selected.

[0314] For example, the selection of reflected sounds is performed based on the volume ratio (L) and time difference (T) calculated by the analysis unit 1301. By performing the selection process based on the time difference (T) between the direct sound and the reflected sound, it is possible to more appropriately select reflected sounds that have a greater impact on the listener's perception than when the selection process is performed based only on the volume difference between the direct sound and the reflected sound.

[0315] Specifically, the selection of whether to generate reflected sound is made by comparing, for example, the volume ratio between the direct sound and the reflected sound, which corresponds to the time difference between the direct sound and the reflected sound, with a preset threshold. The threshold is set with reference to threshold data. The threshold data is an index indicating the boundary between whether a reflected sound relative to the direct sound is perceptible by a listener, and is defined as the ratio between the volume (Id) of the direct sound at the time of arrival and the volume (lr) of the reflected sound at the time of arrival.

[0316] The threshold corresponds to a value expressed by a numerical value or the like determined in correspondence with the time difference (T). The threshold data corresponds to the relationship between the time difference (T) and the threshold, and corresponds to table data or a relational expression used to identify or calculate the threshold for the time difference (T). The format and type of the threshold data are not limited to table data or a relational expression.

[0317] Fig. 11 is a diagram showing the relationship between the time difference between direct sound and reflected sound and a threshold. For example, threshold data of a volume ratio that is predetermined for each value of the time difference between direct sound and reflected sound as shown in Fig. 11 may be referenced. Alternatively, threshold data obtained by interpolation or extrapolation from the threshold data shown in Fig. 11 may be referenced.

[0318] Then, a threshold value for the volume ratio at the time difference (T) calculated by the analysis unit 1301 is identified from the threshold data. Then, the selection unit 1302 determines whether or not to select the reflected sound as a reflected sound to be generated, depending on whether or not the volume ratio (L) between the direct sound and the reflected sound calculated by the analysis unit 1301 exceeds the threshold value.

[0319] By performing selection processing using threshold data of volume ratios that are predetermined for each value of the time difference between direct sound and reflected sound, it is possible to realize selection processing that takes post-masking or precedence effect into consideration. The type, format, storage method, and setting method of threshold data will be described in detail later.

[0320] Next, in the process of generating direct sound and reflected sound (S103 in FIG. 8), the synthesis unit 1303 generates and synthesizes an audio signal of the direct sound and an audio signal of the reflected sound selected by the selection unit 1302 as the reflected sound to be generated.

[0321] The audio signal of the direct sound is generated by applying the arrival time (td) and arrival volume (ld) calculated by the analysis unit 1301 to the sound data of the sound source object included in the input information. Specifically, the sound data is delayed by the arrival time (td) and multiplied by the arrival volume (ld). The process of delaying the sound data is a process of moving the position of the sound data forward or backward on the time axis. For example, a process of delaying sound data without degrading sound quality, as disclosed in Patent Document 2, may be applied.

[0322] The audio signal of the reflected sound is generated by applying the arrival time (tr) and arrival volume (ld) calculated by the analysis unit 1301 to the sound data of the sound source object, just like the direct sound.

[0323] However, unlike the volume of direct sound arriving at the time of arrival, the volume of arrival (lr) when generating reflected sound is a value to which an attenuation rate G of the volume of reflection is applied. G may be an attenuation rate applied to all frequency bands at once. Alternatively, a reflectance rate may be specified for each predetermined frequency band to reflect the bias in frequency components caused by reflection. In this case, the process of applying the volume of arrival (lr) may be performed as a frequency equalizer process, which multiplies each band by an attenuation rate.

[0324] In the above example, the path lengths of the direct sound and the reflected sound candidates as they arrive at the listener are calculated. Furthermore, the arrival times and volumes at the time of arrival are calculated based on the respective path lengths. Then, the reflected sound candidates are selected based on the time difference and volume ratio between them.

[0325] As another example, the selection process may be performed based on the path lengths of the direct sound and the reflected sound as they reach the listener, and the calculation of the arrival times and arrival volumes of the direct sound and the reflected sound, as well as the calculation of the time difference and volume ratio, may be omitted. In this case, a threshold value corresponding to the path length difference may be predetermined for the path length ratio. The selection process may then be performed based on whether the calculated path length ratio is equal to or greater than the threshold value corresponding to the calculated path length difference. This makes it possible to perform the selection process based on the path length difference corresponding to the time difference while reducing the amount of calculation.

[0326] In addition to the path length difference, the value of a parameter indicating the sound propagation velocity or the value of a parameter that affects the sound propagation velocity parameter may also be used.

[0327] (Details of Selection Process) Details of the selection process of whether or not to generate reflected sound will be described.

[0328] The selection of the reflected sound is performed by comparing a threshold value that defines a volume ratio, which is the ratio between the volume of the direct sound when it arrives and the volume of the reflected sound when it arrives, during the time difference (T) between the direct sound and the reflected sound, with the volume ratio (L) calculated by the analysis unit 1301. For example, of the volume ratio threshold values ​​that are predetermined for each value of the time difference between the direct sound and the reflected sound, the volume ratio threshold value for the time difference (T) between the direct sound and the reflected sound calculated by the analysis unit 1301 is referenced. Then, whether or not to select the reflected sound as a reflected sound to be generated is determined depending on whether or not the volume ratio (L) calculated by the analysis unit 1301 exceeds the threshold value.

[0329] The time difference (T) may be, for example, the difference in the time when the direct sound and the reflected sound arrive at the listening position, the time difference between the time it takes for the direct sound and the reflected sound to arrive at the listening position, or the time difference between the time when the direct sound ends and the time when the reflected sound arrives at the listening position. Here, the end time of the direct sound may be calculated by adding the duration of the direct sound to the arrival time of the direct sound.

[0330] The threshold data may be determined based on the minimum time difference at which a listener can perceptually detect a discrepancy between two sounds due to auditory nerve activity or cognitive activity in the brain, more specifically, due to the precedence effect (described below), the temporal masking phenomenon (described below), or a combination thereof. Specific values ​​may be derived from already known research results on the temporal masking effect, the precedence effect, or the echo detection limit, or may be determined through listening experiments assuming application to the virtual space.

[0331] 12A, 12B, and 12C are diagrams showing examples of a method for setting threshold data. As shown in Fig. 12A, 12B, and 12C, the threshold data is represented by a graph in which the horizontal axis represents the time difference between direct sound and reflected sound and the vertical axis represents the volume ratio between direct sound and reflected sound, and the threshold is the boundary (threshold) between whether the reflected sound is perceived or not.

[0332] The threshold data may be expressed by an approximation formula having the time difference between the direct sound and the reflected sound as a variable. Alternatively, the threshold data may be stored in an area of ​​memory 1404 as an array of indexes of the time difference between the direct sound and the reflected sound and thresholds corresponding to the indexes, as shown in FIG.

[0333] Note that when the height of the line parallel to the horizontal axis (minimum audibility limit) in Example 4 of Fig. 12C is used as the threshold, the volume of the reflected sound itself is compared with the threshold, not the volume ratio (L) between the reflected sound and the direct sound. This is because the threshold indicates the volume at the boundary between whether a sound can be perceived by a listener and is a threshold for determining that sounds lower in volume than the threshold will not be reproduced. In other words, the threshold corresponding to the minimum audibility limit is not a threshold for the ratio between the volume of the reflected sound and the volume of the direct sound.

[0334] When the minimum audible limit is used as the threshold, the time difference (T) does not need to be calculated because the threshold is constant regardless of the time difference (T).

[0335] When multiple reflected sounds are generated in the analysis process (S101 in FIG. 8), the selection process may be performed on all reflected sounds, or on only those reflected sounds with high evaluation values ​​based on evaluation values ​​derived for each reflected sound using a preset evaluation method. Here, the evaluation value of a reflected sound corresponds to the perceptual importance of the reflected sound. A high evaluation value corresponds to a large evaluation value, and these expressions may be interchangeable.

[0336] The selection unit 1302 may calculate an evaluation value of the reflected sound using a pre-set evaluation method based on, for example, the volume of the sound source, the visibility of the sound source, the positioning of the sound source, the visibility of the reflecting object (obstacle object), or the geometric relationship between the direct sound and the reflected sound.

[0337] Specifically, the louder the volume of the sound source, the higher the evaluation value may be. Furthermore, in order to match the visual localization with the acoustic localization, the evaluation value may be high when the sound source object or a reflective object (obstacle object) is visible to the listener, or when the localization of the sound source object is high.

[0338] Furthermore, the difference in the arrival angle between the direct sound and the reflected sound and the difference in the arrival time between the direct sound and the reflected sound have a significant impact on the perception of the space, so if the difference in the arrival angle between the direct sound and the reflected sound is large or if the difference in the arrival time between the direct sound and the reflected sound is large, the evaluation value may be high.

[0339] The information on the volume of the sound source may indicate a reference volume defined for each content, a temporal transition of the volume, or both.

[0340] For example, if the virtual space is a virtual conference room and the direct sound is conversation, the volume transitions intermittently over a short period of time. That is, sound and silence alternate. If the virtual space is a concert hall and the direct sound is a musical performance, the volume is maintained for a certain period of time. If the virtual space is a battlefield and the direct sound is an explosion, the volume increases for a moment and then remains silent or low.

[0341] In this way, the volume information of the sound source may include not only information on the reference volume corresponding to the volume setting when the sound is emitted into the virtual space, but also information on the transition of the volume of the sound.

[0342] The transition information may be represented by data indicating frequency characteristics in a time series. The transition information may be represented by data indicating the duration of a sound section. The transition information may be represented by data indicating a time series of the duration of a sound section and the duration of a silent section. The transition information may be represented by data listing, in a time series, multiple pairs of durations during which the amplitude of a sound signal can be considered steady (considered to be roughly constant) and the amplitude values ​​of the signal during those durations.

[0343] The transition information may be expressed as data on the duration for which the frequency characteristics of the sound signal can be considered to be stationary, or may be expressed as data listing, in chronological order, multiple pairs of durations for which the frequency characteristics of the sound signal can be considered to be stationary and the frequency characteristics during those periods.

[0344] Furthermore, efforts to use temporal transitions in the frequency characteristics of signals in acoustic processing of virtual spaces have been widely undertaken in the past (see, for example, Patent Document 1). In light of such prior art, it goes without saying that the above pair may be a pair of a time length during which the frequency characteristics are constant and the frequency characteristics themselves.

[0345] The geometric relationship may be the relationship between the positions of the sound source, the listener, and the reflecting object in the virtual space. These relationships allow the geometric calculation of the path lengths of the direct sound and the reflected sound. Therefore, by utilizing the relationship in which the volume is inversely proportional to the distance, it is possible to calculate the reference volume of the reflected sound relative to the reference volume of the direct sound.

[0346] The reference volume of the reflected sound may be calculated using the reflection coefficient of the reflecting object. A commonly used typical value may also be used as the reflection coefficient. On the other hand, if a special condition exists, such as the reflecting object being covered with a sound-absorbing material, a specially assigned reflection coefficient may be used as the reflection coefficient of the reflecting object.

[0347] The reflected sound may be evaluated based on its volume, which may be calculated from the geometric relationship between the direct sound and the reflected sound and the index assigned to the reflective object, as described above, and may be evaluated by comparing the volume with a predetermined threshold.

[0348] Furthermore, information indicating the temporal transition of the volume of the sound source may be reflected in the evaluation. For example, if the information indicating the temporal transition of the volume of the sound source indicates the duration of a sound section, and the time is within the sound section, the evaluation value of the reflected sound may be maintained as is. On the other hand, if the time is outside the sound section, processing may be performed to reduce or set the evaluation value of the reflected sound to zero even if the reference volume of the reflected sound exceeds the threshold.

[0349] Alternatively, the information indicating the temporal transition of the volume of the sound source may be data listing, in time series, multiple pairs of durations during which the amplitude of a sound signal is considered to be roughly constant and the amplitude values ​​of the signal during those durations. In this case, the reference volume of the reflected sound may be changed in conjunction with changes in the amplitude values ​​in the data to evaluate the reflected sound.

[0350] Furthermore, both information on the reference volume and information on the volume that changes over time may be used as information indicating the volume of the direct sound. For example, after an evaluation value is calculated based on information on the reference volume, the evaluation value may be corrected using information on the volume that changes over time.

[0351] In the evaluation of reflected sounds, all of the above-described methods may be executed, or only some of them may be executed. For example, reflected sounds may be evaluated using a plurality of evaluation methods, or may be evaluated using a single evaluation method.

[0352] When reflected sound is evaluated using multiple evaluation methods, whether or not to select the reflected sound may be determined based on an evaluation value determined comprehensively using the multiple evaluation methods, or may be determined based on the evaluation values ​​for each of the multiple evaluation methods.

[0353] When determining whether to select a reflected sound based on each of a plurality of evaluation methods, the audio signal processing device 1001 may select a sound if all of the plurality of evaluation results based on the plurality of evaluation methods indicate that the sound should be selected. Alternatively, the audio signal processing device 1001 may select a sound if any one of the plurality of evaluation results based on the plurality of evaluation methods indicates that the sound should be selected.

[0354] Furthermore, for example, priorities may be assigned to the first to third evaluation methods. Then, when it is determined that sound should not be selected using the first evaluation method, the audio signal processing device 1001 may ultimately determine that sound should not be selected without depending on the determination results of the second and third evaluation methods. Furthermore, when it is determined that sound should not be selected using one of the second and third evaluation methods but that sound should be selected using the other, the audio signal processing device 1001 may ultimately determine that sound should be selected.

[0355] Furthermore, the selection process and the evaluation process may be performed independently, or only one of them may be performed. Alternatively, the evaluation process may be performed only on reflected sounds that have been determined to be selected in the selection process, and the evaluation process may then re-determine whether or not to select the reflected sounds. Alternatively, the evaluation process may be performed only on reflected sounds that have been determined not to be selected in the selection process, and the evaluation process may then re-determine whether or not to select the reflected sounds.

[0356] The above-described selection process can be interpreted as a process of selecting reflected sounds according to the properties of direct sounds. For example, in the process of selecting reflected sounds according to the properties of direct sounds, a threshold value used for selecting reflected sounds is set or adjusted according to the properties of the direct sounds. Alternatively, an evaluation value used for selecting reflected sounds is calculated based on one or more of the volume of a sound source, the visibility of a sound source, the localization of a sound source, the visibility of a reflecting object (obstacle object), and the geometric relationship between the direct sound and the reflected sound.

[0357] Furthermore, the process of selecting reflected sounds according to the properties of direct sounds is not limited to the process of setting or adjusting a threshold value according to the properties of direct sounds and the process of calculating an evaluation value used to select reflected sounds to be processed, and other processes may be performed. Even when the process of setting or adjusting a threshold value according to the properties of direct sounds or the process of calculating an evaluation value used to select reflected sounds to be processed is performed, the process may be partially changed or new processes may be added.

[0358] Note that setting the threshold value may include adjusting the threshold value, changing the threshold value, and the like.

[0359] (Method of Setting Thresholds) The threshold data used in the selection process may be set with reference to, for example, an echo detection limit based on the already known precedence effect or a masking threshold based on the post-masking effect.

[0360] The precedence effect is a phenomenon in which, when sounds are heard from two locations, the one heard first is perceived as the source of the sound. If two short sounds merge and sound like a single sound, the location where the entire sound is heard (localization) is largely determined by the location of the first sound. The echo detection limit is a phenomenon caused by the precedence effect, and is the minimum time difference at which a listener can perceive a discrepancy between two sounds.

[0361] 12C, the horizontal axis corresponds to the arrival time of the reflected sound (echo), specifically, the delay time from the arrival time of the direct sound to the arrival time of the reflected sound, and the vertical axis corresponds to the volume ratio of the detectable reflected sound to the direct sound, specifically, the threshold value for whether the reflected sound arriving with a delay is detectable.

[0362] Fig. 13 is a diagram showing an example of a method for setting a threshold value. The horizontal axis in Fig. 13 corresponds to the arrival time of the reflected sound, specifically, the time difference (T) between the direct sound and the reflected sound. The vertical axis in Fig. 13 corresponds to the volume of the reflected sound. Specifically, the vertical axis in Fig. 13 may correspond to the volume of the reflected sound determined relatively to the volume of the direct sound (volume ratio), or may correspond to the volume of the reflected sound determined absolutely regardless of the volume of the direct sound.

[0363] For example, when the listener and the obstacle object are relatively far apart as shown in Fig. 9, the arrival time of the reflected sound is delayed, and the threshold value is set low, as shown in C of Fig. 13. As a result, reflected sound is generated in the case of Fig. 9. On the other hand, when the listener and the obstacle object are relatively close as shown in Fig. 10, the arrival time of the reflected sound is earlier than in the case of Fig. 9, and the threshold value is set high, as shown in B of Fig. 13. As a result, reflected sound is not generated in the case of Fig. 10.

[0364] The threshold data may also be stored in the memory 1404, retrieved from the memory 1404 during the selection process, and used in the selection process.

[0365] 14 is a flowchart showing an example of the selection process. First, the selection unit 1302 specifies the reflected sound detected by the analysis unit 1301 (S201). Then, the selection unit 1302 detects the volume ratio (L) between the direct sound and the reflected sound and the time difference (T) between the direct sound and the reflected sound (S202 and S203).

[0366] The time difference (T) may be, for example, the time difference between the time it takes for a direct sound and a reflected sound to arrive at the listening position, the time difference between the arrival time of the direct sound and the arrival time of the reflected sound, or the time difference between the time when the direct sound ends and the time when the reflected sound arrives at the listening position. Here, an example based on the time difference between the arrival time of the direct sound and the arrival time of the reflected sound will be described.

[0367] Specifically, the selection unit 1302 calculates the difference between the path length of the direct sound and the path length of the reflected sound from the position information of the sound source object and the listener, and the position information and shape information of the obstacle object.The selection unit 1302 then divides this difference in length by the speed of sound to detect the time difference (T) between the time when the direct sound arrives at the listener's position and the time when the reflected sound arrives at the listener's position.

[0368] The volume of the sound reaching the listener attenuates in proportion to the distance to the listener (inversely proportional to the distance) relative to the volume of the sound source. Therefore, the volume of the direct sound is obtained by dividing the volume of the sound source by the path length of the direct sound. The volume of the reflected sound is obtained by dividing the volume of the sound source by the path length of the reflected sound and then multiplying the result by the attenuation rate assigned to the virtual obstacle object. The selection unit 1302 detects the volume ratio by calculating the ratio between these volumes.

[0369] The selection unit 1302 also uses the threshold data to identify a threshold corresponding to the time difference (T) (S204), and determines whether the detected volume ratio (L) is equal to or greater than the threshold (S205).

[0370] If the volume ratio (L) is equal to or greater than the threshold (Yes in S205), the selection unit 1302 selects the reflected sound as the reflected sound to be generated (S206). If the volume ratio (L) is smaller than the threshold (No in S205), the selection unit 1302 does not select the reflected sound as the reflected sound to be generated (S207). That is, in this case, the selection unit 1302 determines that the reflected sound is not to be generated.

[0371] Thereafter, the selection unit 1302 determines whether or not there is an unspecified reflected sound (S208). If there is an unspecified reflected sound (Yes in S208), the selection unit 1302 repeats the above-described processing (S201 to S207). If there is no unspecified reflected sound (No in S208), the selection unit 1302 ends the processing.

[0372] This selection process may be performed on all reflected sounds generated in the analysis process, or may be performed only on the reflected sounds with high evaluation values ​​described above.

[0373] (Details of Threshold Storage Method) The threshold data according to this embodiment is stored in the memory 1404 of the audio signal processing device 1001. The format and type of the stored threshold data may be any format and any type. When multiple formats and multiple types of thresholds are stored, the selection process may determine which format and type of threshold to use in the selection process of the reflected sound. The method of determining which threshold data to use in the selection process will be described later.

[0374] Furthermore, threshold data of multiple formats and types may be stored in combination. The combined threshold data may be read from the spatial information management units (1201, 1211) and a threshold to be used in the selection process may be set. The threshold data stored in the memory 1404 may be stored in the spatial information management units (1201, 1211).

[0375] The threshold data may be stored as thresholds at each time difference, for example, as shown in [Example 1] and [Example 2] of FIG. 12C.

[0376] Furthermore, the threshold data may be stored as table data in which thresholds and time differences (T) are associated with each other, as shown in FIG. 11 . That is, the threshold data may be stored as table data having the time difference (T) as an index. Of course, the thresholds shown in FIG. 11 are merely an example, and the thresholds are not limited to the example of FIG. 11 . Furthermore, instead of storing the thresholds themselves, the thresholds may be approximated by a function having the time difference (T) as a variable, and the coefficients of the function may be stored. Furthermore, a combination of multiple approximation formulas may be stored.

[0377] For example, the threshold data may be expressed by the following formula, where the time difference (T) is timeDiff and the threshold is gainThresh.

[0378]

[0379] The threshold is defined only within the time range in which the precedence effect occurs. For time differences outside that time range (values ​​of 1 ms or less or greater than 40 ms in the above formula 1), the determination using gainThresh is not performed, and as described below, the determination may be performed using only the threshold indicating the minimum volume reproduced in the virtual space. For example, outside that time range, the reflected sound may be selected as the processing target sound regardless of the precedence effect. Alternatively, outside that time range, the reflected sound may be selected as the processing target sound if it is equal to or greater than the minimum volume, regardless of the precedence effect.

[0380] Experiments conducted by the present inventors have revealed that it is desirable to approximate the threshold with an upwardly convex function in the time range in which the precedence effect occurs. The above formula 1 is an example of an approximation formula generated based on the experiments.

[0381] 15 is a diagram illustrating a listening experiment of a leading sound and a following sound related to the precedence effect. Specifically, FIG. 15 illustrates a first virtual speaker, a second virtual speaker, a listener, an avatar of the listener, and the like.

[0382] In the listening experiment, the first virtual speaker is a virtual speaker that outputs a direct sound, which is a preceding sound, in the virtual space. The direct sound is a sound heard in a normal scene such as speech or music. The second virtual speaker is a virtual speaker that outputs a reflected sound, which is a following sound, in the virtual space. The reflected sound is a sound obtained by delaying and attenuating the direct sound.

[0383] The first HRTF is a head-related transfer function when sound output from the first virtual speaker reaches the listener's avatar. The second HRTF is a head-related transfer function when sound output from the second virtual speaker reaches the listener's avatar. The HRTFs (Head-Related Transfer Functions) disclosed in Non-Patent Document 2 are used for each of the first HRTF and the second HRTF.

[0384] By sequentially applying various values ​​to the delay and attenuation amounts used to generate reflected sound, the volume ratio and time difference between the sound output from the first virtual speaker and the sound output from the second virtual speaker are sequentially set.

[0385] Fig. 16 is a diagram showing experimental parameters in the horizontal plane direction. Fig. 17 is a diagram showing experimental parameters in the vertical plane direction. As shown in Figs. 16 and 17 , the experimental parameters include a plurality of variations. Furthermore, the volume ratio in the experimental parameters in the horizontal plane direction is set according to the time difference. Specifically, it is expected that the threshold value for the volume ratio will decrease as the time difference increases, and the threshold value for the volume ratio will increase as the time difference decreases. Therefore, the range of the experiment is narrowed to avoid unnecessary experiments.

[0386] The test subject, a listener, listens to the sound reaching the listener's avatar using headphones. The listener listens to the sound when the second virtual speaker is ON and when the second virtual speaker is OFF, and answers whether or not it is possible to perceive a difference between ON and OFF. Here, "being able to perceive a difference" includes being able to perceive a change in the timbre (including the level of impression) of the direct sound due to the reflected sound, and hearing the direct sound and the reflected sound as separate sounds.

[0387] As a result of experiments, the inventors have found that the boundary between whether it is possible to distinguish between ON and OFF can be expressed, for example, as in the above-mentioned formula 1 (gainThresh).

[0388] As shown in the above formula 1, the boundary is an upwardly convex curve (broken line) in the time range in which the precedence effect is assumed to occur. This experimental result does not contradict that this phenomenon is based on the precedence effect. In other words, it is assumed that the precedence effect does not occur when the time difference between the preceding sound and the following sound exceeds about 40 msec, and this assumption agrees well with the experimental result that the threshold value attenuates more steeply as the time difference becomes longer.

[0389] Furthermore, the domain for expressing the boundary only needs to be the time range in which the precedence effect is expected to occur. This allows the boundary to be expressed as a mathematical expression, which can be expressed with a combination of a small number of mathematical expressions. Furthermore, when the boundary is expressed as a line (a collection of closely spaced dots: table data), the boundary line can be expressed with a small memory size.

[0390] The boundary was not significantly related to the placement position of each virtual speaker. Therefore, Equation 1 does not include an azimuth angle factor. However, to more precisely control the selection of reflected sounds, an azimuth angle factor may be incorporated. For example, an equation similar to Equation 1 may be set for each azimuth angle. Alternatively, an azimuth angle term may be added to Equation 1.

[0391] The memory 1404 may store information regarding a relational expression showing the relationship between the time difference (T) and the threshold value. That is, an expression having the time difference (T) as a variable may be stored. The threshold value of each time difference (T) may be approximated by a straight line or a curve, and parameters indicating the geometric shape of the line or curve may be stored. For example, if the geometric shape is a straight line, the starting point and slope for expressing the straight line may be stored.

[0392] Furthermore, the type and format of threshold data may be determined and stored for each characteristic of the direct sound. Furthermore, parameters for adjusting the threshold according to the characteristic of the direct sound and using it in the selection process may be stored. The process of adjusting the threshold according to the characteristic of the direct sound and using it in the selection process will be described later as a modified example of the threshold setting method.

[0393] As an example of storing a combination of multiple types of threshold data, the larger of the masking threshold and the echo detection limit threshold may be stored for each time difference (T) as shown in [Example 3] of Fig. 12C. Alternatively, the larger of the minimum volume reproduced in the virtual space and the echo detection limit threshold may be stored for each time difference (T) as shown in [Example 4] of Fig. 12C.

[0394] The combination of multiple types of threshold data is not limited to this. For example, maximum value information for each time difference (T) in multiple types of threshold data may be stored.

[0395] In the above description, the information about the threshold value has a one-dimensional index representing the time. The information about the threshold value may also have a two-dimensional or three-dimensional index including a variable relating to the direction of arrival.

[0396] 18 is a diagram showing the relationship between the direction of a direct sound, the direction of a reflected sound, the time difference, and a threshold value. For example, as shown in FIG. 18, threshold values ​​calculated in advance according to the relationship between the direction of a direct sound (θ), the direction of a reflected sound (γ), the time difference (T), and the volume ratio (L) may be stored.

[0397] The direction of direct sound (θ) corresponds to the angle of the direction from which the direct sound arrives relative to the listener. The direction of reflected sound (γ) corresponds to the angle of the direction from which the reflected sound arrives relative to the listener. Here, the direction the listener is facing is defined as 0 degrees. The time difference (T) corresponds to the difference between the arrival time of the direct sound and the arrival time of the reflected sound at the listening position. The volume ratio (L) corresponds to the volume ratio between the volume of the direct sound when it arrives and the volume of the reflected sound when it arrives.

[0398] Of course, the thresholds shown in Fig. 18 are merely an example, and the thresholds are not limited to the example of Fig. 18. Also, Fig. 18 mainly illustrates thresholds when the angle (θ) of the arrival direction of the direct sound is 0 degrees. However, thresholds when the arrival direction (θ) of the direct sound is other than 0 degrees are also stored in memory 1404.

[0399] In the above example, the thresholds are stored in an array having the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound as independent variables or indices. However, the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound do not necessarily have to be used as independent variables.

[0400] For example, the angle difference between the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound may be used. This angle difference corresponds to the angle between the arrival direction of the direct sound and the arrival direction of the reflected sound, and may be expressed as the arrival angle between the direct sound and the reflected sound.

[0401] Fig. 19 is a diagram showing the relationship between the angle difference, the time difference, and the threshold. For example, a threshold calculated in advance using the angle difference (Φ) between the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound as a variable may be stored as in the example shown in Fig. 19. Of course, the threshold shown in Fig. 19 is just an example, and the threshold is not limited to the example of Fig. 19.

[0402] 19, it is possible to reduce the number of variables used to derive thresholds, which in turn makes it possible to reduce the number of thresholds stored in memory 1404. Therefore, it is possible to reduce the amount of data stored in memory 1404.

[0403] In addition, when the angle difference (Φ) between the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound is used, the threshold data may be stored in a two-dimensional array. In addition, in the selection process, the difference between the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound may be calculated using a three-dimensional array.

[0404] A method for selecting reflected sounds using a threshold value according to the direction of arrival will be described later.

[0405] 12A, 12B, and 12C, a plurality of formats and a plurality of types of thresholds may be stored in the spatial information management units (1201, 1211). Then, it may be determined which of the plurality of formats and the plurality of types of thresholds to use in the process of selecting reflected sounds. Specifically, as shown in example 3 of FIG. 12C, the highest threshold may be adopted for the time difference (T) corresponding to the arrival time of the reflected sound.

[0406] Furthermore, as shown in Example 4, a masking threshold, an echo detection threshold, and a threshold indicating the minimum volume to be reproduced in the virtual space may be stored, and the highest threshold may be adopted for the time difference (T) corresponding to the arrival time of the reflected sound.

[0407] (Second Modification of Threshold Setting Method) As another example of the threshold setting method, a method of setting a threshold depending on the properties of the direct sound will be described.

[0408] Fig. 20 is a block diagram showing another example configuration of the rendering unit 1300 shown in Fig. 7. The rendering unit 1300 in Fig. 20 differs from the rendering unit 1300 in Fig. 7 in that it includes a threshold adjustment unit 1304. Descriptions of the components other than the threshold adjustment unit 1304 are the same as those described in Fig. 7 and will therefore be omitted.

[0409] The threshold adjustment unit 1304 selects a threshold to be used by the selection unit 1302 from the threshold data based on information indicating the properties of the audio signal. Alternatively, the threshold adjustment unit 1304 may adjust the threshold included in the threshold data based on information indicating the properties of the audio signal.

[0410] The information indicating the properties of the audio signal may be included in the input signal. Then, the threshold adjustment unit 1304 may acquire the information indicating the properties of the audio signal from the input signal. Alternatively, the analysis unit 1301 may derive the properties of the audio signal by analyzing the audio signal included in the received input signal, and output the information indicating the properties of the audio signal to the threshold adjustment unit 1304.

[0411] The information indicating the characteristics of the audio signal may be obtained before the rendering process begins, or may be obtained each time the rendering process is performed.

[0412] Furthermore, the threshold adjustment unit 1304 does not have to be included in the audio signal processing device 1001, and another communication device may fulfill the role of the threshold adjustment unit 1304. In this case, the analysis unit 1301 or the selection unit 1302 may acquire information indicating the properties of the audio signal, threshold data according to the properties, or information for adjusting the threshold data according to the properties from the other communication device via the communication IF 1403.

[0413] Fig. 21 is a flowchart showing another example of the selection process. Fig. 22 is a flowchart showing yet another example of the selection process. In Fig. 21 and Fig. 22, a threshold is set according to the properties of the direct sound. Specifically, in Fig. 21, the threshold adjustment unit 1304 specifies a threshold from threshold data based on the time difference (T) and the properties of the audio signal. In Fig. 22, the threshold adjustment unit 1304 adjusts the threshold specified from the threshold data based on the time difference (T) based on the properties of the audio signal.

[0414] The operation of each example will be described below, with the explanation of the processes common to the example in FIG.

[0415] First, an example of the processing shown in Fig. 21 will be described. Here, threshold data for each property of the direct sound is stored in advance in memory 1404. As a result, multiple threshold data corresponding to multiple properties are stored in advance in memory 1404. Then, the threshold adjustment unit 1304 identifies threshold data to be used in the selection processing of reflected sounds from the multiple threshold data.

[0416] For example, the threshold adjustment unit 1304 acquires the characteristics of the direct sound based on the input signal (S211). The threshold adjustment unit 1304 may acquire the characteristics of the direct sound associated with the input signal. Then, the threshold adjustment unit 1304 identifies a threshold corresponding to the time difference (T) and the characteristics of the direct sound (S212).

[0417] As shown in FIG. 22, the threshold value adjusting unit 1304 may adjust the threshold value specified by the selecting unit 1302 based on the properties of the direct sound (S221).

[0418] In either case, the input signal may include information indicating the characteristics of the audio signal, information for adjusting the threshold in accordance with the characteristics of the audio signal, or both of these, and the threshold adjustment unit 1304 may adjust the threshold using one or both of these.

[0419] Furthermore, the information indicating the properties of the audio signal, the information for adjusting the threshold, or both may be transmitted in an input signal other than the input signal containing the audio signal. In this case, the input signal containing the audio signal may include information associating the other input signal with the input signal, or the information associating the other input signal with the input signal may be stored in memory 1404 together with information regarding the threshold.

[0420] In the examples of Figures 21 and 22, the threshold value used to select the reflected sound is set according to the properties of the direct sound, i.e., the properties of the audio signal. Threshold data set in advance for each property may be used, as in Figure 21, or the threshold value may be adjusted according to the properties of the audio signal, as in Figure 22. Furthermore, the parameters of the threshold data may be adjusted according to the properties of the audio signal.

[0421] The operation performed by the threshold adjustment unit 1304 may be performed by the analysis unit 1301 or the selection unit 1302. For example, the analysis unit 1301 may acquire the properties of the audio signal. Alternatively, the selection unit 1302 may set the threshold according to the properties of the audio signal.

[0422] Next, the relationship between the characteristics of the audio signal and the threshold will be described.

[0423] Two short sounds that arrive consecutively at a listener's ears will be heard as a single sound if the time interval between them is sufficiently short. This phenomenon is called the precedence effect. It is known that the precedence effect occurs only for discontinuous, i.e., transient, sounds (Non-Patent Document 1). Therefore, when an audio signal represents a stationary sound, the echo detection threshold may be set lower than when the audio signal represents a non-stationary sound.

[0424] That is, in accordance with the characteristics of such precedence effect, for example, if the direct sound is a steady sound, the threshold value is set to be small. Also, the higher the steadyness, the smaller the threshold value may be set.

[0425] An example of processing when the nature of the audio signal is stationary will be described. First, the threshold adjustment unit 1304 or the analysis unit 1301 determines stationarity based on the amount of fluctuation in the frequency components of the audio signal over time. For example, if the amount of fluctuation is small, the stationarity is determined to be high. Conversely, if the amount of fluctuation is large, the stationarity is determined to be low. As a result of the determination, a flag indicating the level of stationarity may be set, or a parameter indicating stationarity may be set according to the amount of fluctuation.

[0426] Next, the threshold adjustment unit 1304 may adjust the threshold data or threshold based on information indicating stationarity, such as a flag or parameter indicating the stationarity of the audio signal, and set the adjusted threshold data or threshold as the threshold data or threshold to be used in the selection unit 1302.

[0427] Alternatively, parameters for setting threshold data according to information indicating the continuity of the direct sound may be stored in advance in the memory 1404. In this case, the threshold adjustment unit 1304 may determine the continuity of the audio signal, and set threshold data used for selecting reflected sounds based on the information indicating the continuity and the parameters.

[0428] Alternatively, multiple parameters of the threshold data may be stored in advance in memory 1404 in correspondence with multiple patterns of the continuity of the direct sound. In this case, threshold adjustment unit 1304 may determine the continuity of the audio signal, select parameters of the threshold data based on the pattern of the continuity of the direct sound, and set threshold data to be used for selecting reflected sounds based on the parameters of the threshold data.

[0429] The constancy of an audio signal may be determined based on the amount of fluctuation in the frequency components of the audio signal each time the audio signal is input.

[0430] Alternatively, the continuity of the audio signal may be determined based on information indicating the continuity that is pre-linked to the audio signal. That is, the information indicating the continuity of the audio signal may be pre-linked to the audio signal and stored in the memory 1404. The analysis unit 1301 may acquire the information indicating the continuity that is pre-linked to the audio signal every time an audio signal is input. Then, the threshold adjustment unit 1304 may adjust the threshold based on the information indicating the continuity that is pre-linked to the audio signal.

[0431] As another example of how the threshold may be set depending on the nature of the audio signal, the echo detection limit may be set to a shorter range if the audio signal represents a short sound (such as a click) than if the audio signal represents a long sound. This process is based on the properties of the precedence effect.

[0432] It is known that due to the precedence effect, two short sounds that arrive consecutively at a listener's ears are perceived as a single sound if the time interval between them is sufficiently short. The upper limit of this time interval depends on the duration of the sounds. For example, the upper limit of this time interval is about 5 ms for a click sound, but can be as long as 40 ms for complex sounds such as human voices or music (Non-Patent Document 1).

[0433] According to the characteristics of such precedence effect, for example, if the duration of the direct sound is short, a short threshold value is set. Also, the shorter the duration of the direct sound, the shorter the threshold value is set.

[0434] Setting a short threshold value means that a threshold value corresponding to an echo detection limit based on the characteristics of the precedence effect is set within a range where the time difference (T) between the direct sound and the reflected sound is small. Outside this range, a threshold value corresponding to an echo detection limit based on the characteristics of the precedence effect is not set. In other words, outside this range, the threshold value is small. Therefore, setting a short threshold value for a short sound can correspond to setting a small threshold value for a short sound.

[0435] As another example of setting the threshold depending on the characteristics of the direct sound, if the direct sound is an intermittent sound (such as speech), the threshold may be set lower than if the direct sound is a continuous sound (such as music).

[0436] For example, when the direct sound corresponds to speech, sound and silence portions are repeated, and only the post-masking effect occurs in the silence portions. On the other hand, when the direct sound is a continuous sound such as music content, both the post-masking effect and the simultaneous masking effect due to the sound occurring at that time occur. Therefore, the overall masking effect is higher in the case of music than in the case of speech.

[0437] According to the characteristics of the masking effect as described above, the threshold may be set higher for music, etc. than for speech, etc. Conversely, the threshold may be set lower for speech, etc. than for music, etc. In other words, if the direct sound has many intermittent parts, the threshold may be set lower.

[0438] As described above, the information indicating the properties of the direct sound may be information indicating the constancy, intermittency, duration, etc. of the direct sound. Furthermore, the information indicating the properties of the direct sound may be any combination of these. Furthermore, the information indicating the properties of the direct sound may be information indicating the time variation of any of these, or information indicating the time variation of any combination of these. In other words, the information indicating the properties of the direct sound may be information indicating the time variation of the direct sound.

[0439] For example, as described in the description of the stationarity determination, the information indicating the properties of the direct sound may be time-series data of frequency characteristics, where the frequency characteristics may be expressed in a commonly used format such as a gain value for each frequency band, a Fourier series for a time-domain signal, or an LPC coefficient or cepstrum coefficient for determining a frequency envelope.

[0440] Furthermore, the information indicating the properties of the direct sound may be information indicating the intermittency of the direct sound, which lists in chronological order a plurality of pairs of durations during which the amplitude of a signal is steady and the amplitude values ​​of the signal during those durations (an outline of the amplitude envelope). Here, the amplitude values ​​may be expressed as a ratio to a reference volume.

[0441] Furthermore, the information indicating the properties of the direct sound may be information regarding the frequency characteristics of the direct sound. For example, the information indicating the properties of the direct sound may be information indicating the constancy of the frequency characteristics of the direct sound. Specifically, the information indicating the properties of the direct sound may be information (approximate spectrogram shapes) listing in time series multiple pairs of durations during which the frequency characteristics are small and the frequency characteristics of the signal during those durations. Here, the volume used as a reference for the frequency characteristics may be the reference volume.

[0442] For example, the information indicating the time variation of the direct sound is information indicating the envelope of the direct sound. The information indicating the time variation of the direct sound may be used when the "minimum audible limit" described in [Example 4] of Fig. 12C is the threshold. The signal to be compared with the minimum audible limit is the volume of the reflected sound.

[0443] The volume of reflected sound is obtained by geometric calculation using information on the positions of the sound source, listener, and reflecting object. Specifically, the reference volume of the reflected sound relative to the reference volume of the sound source is obtained. By increasing or decreasing the reference volume of the reflected sound using information on the transition of the sound source's loudness as information indicating the properties of the direct sound, it is possible to accurately determine the volume of the reflected sound from moment to moment. This is because fluctuations in the volume of the sound source are reflected in fluctuations in the volume of the reflected sound.

[0444] After adjusting the volume of the reflected sound, the volume of the reflected sound is compared with a threshold value, thereby making it possible to more accurately select the reflected sound that is required for auditory perception.

[0445] Of course, it goes without saying that the same result can be obtained by adjusting the threshold based on the inverse of the information on the transition in loudness of the sound source, without adjusting the reference volume of the reflected sound, and then comparing the adjusted threshold with the reference volume of the reflected sound. In other words, the reference volume of the reflected sound may be adjusted using the information on the transition in loudness of the sound source, or the threshold may be adjusted using the information on the transition in loudness of the sound source. Adjustment of the reference volume of the reflected sound and adjustment of the threshold correspond to each other.

[0446] Depending on the composition of the surface of an object that reflects sound, the sound reflectance (the rate at which sound decays due to reflection) varies for each frequency band. Therefore, as will be described later, a sound reflectance (decay rate) may be associated with each sound reflecting object for each frequency band. Using such reflectance information and spectrogram information, it is possible to more accurately determine whether or not to select the reflected sound. For example, the following processing is performed.

[0447] Specifically, for example, spectrogram information may indicate that high frequency components are more prevalent than low frequency components in a certain time interval, and sound reflectance information may indicate that high frequency components have significantly lower reflectance than low frequency components.

[0448] In this case, even if the amplitude of the sound source signal on the time axis is large, the volume of the reflected sound obtained by multiplying the frequency components indicated by the spectrogram information by the attenuation rate for each frequency band indicated by the reflectivity information will be small, and the reflected sound may not be selected.

[0449] As described above, the information indicating the characteristics of the direct sound may be information indicating a time variation of the direct sound. For example, the information indicating the characteristics of the direct sound may indicate a value obtained by analyzing the direct sound for a predetermined time length.

[0450] Specifically, the information indicating the properties of the direct sound may be information obtained by calculating the average energy or average amplitude of the direct sound for each predetermined time length. Alternatively, the information indicating the properties of the direct sound may be information obtained by calculating the energy or average amplitude of the direct sound for each short-term analysis length and calculating a weighted average of the energy or average amplitude for each long-term analysis length longer than the short-term analysis length.

[0451] More specifically, for example, the information indicating the time variation of the direct sound may be information obtained by calculating the energy or average amplitude of the direct sound for each predetermined short time length (for example, 5 ms; hereinafter, frames of this time length will be referred to as analysis frames). Furthermore, the information indicating the time variation of the direct sound may be information represented by a weighted average of the energy or average amplitude calculated for the past N-1 analysis frames.

[0452] If the energy of the n-th analysis frame is expressed as E(n), information I(n) indicating the properties of the direct sound can be calculated according to the following equation.

[0453]

[0454] Here, the parameter a(i) represents a weighting coefficient. Generally, a(i) is set so that a(i)≧0 and the sum of a(i) is 1. However, the method for setting a(i) is not limited to this.

[0455] Note that information I(n) indicating the properties of the direct sound is calculated every 5 ms that the direct sound is captured. In other words, it is possible to calculate the time variation of information I(n) indicating the properties of the direct sound with low delay. Therefore, this method is suitable for application requiring real-time performance.

[0456] Furthermore, information I(n) indicating the properties of the direct sound may be calculated according to the following formula:

[0457]

[0458] Here, the parameter b(i) represents a weighting coefficient. Generally, b(i) is set so that b(i)≧0 and the sum of b(i) is 1. However, the method for setting b(i) is not limited to this.

[0459] In this equation, information I(n) indicating the properties of the direct sound is calculated recursively, making it possible to calculate the average energy over a long time period with a small amount of calculation.

[0460] The above equations 2 and 3 can be considered as filters in which E(n) is the input signal and I(n) is the output signal. In this case, equation 2 is a moving average (MA) model filter, and equation 3 is an autoregressive (AR) model filter, both of which have low-pass filter characteristics. Alternatively, an ARMA model filter, which is a combination of both, may be used.

[0461] Note that the method of deriving the information indicating the time variation of the direct sound is not limited to the above-described formula or filter, and other known methods may be used. As described above, the information indicating the time variation of the direct sound indicates a value obtained by analyzing the direct sound for a predetermined time length. The direct sound may be analyzed from a perspective other than average energy.

[0462] As described above, the information indicating the properties of the direct sound may be information related to the frequency characteristics of the direct sound. The information related to the frequency characteristics of the direct sound may be information calculated using the frequency characteristics of the direct sound. For example, the information related to the frequency characteristics of the direct sound may be information obtained as the average energy of the low-frequency components by averaging the low-frequency components of the direct sound over a predetermined analysis length.

[0463] Specifically, a low-pass filter is applied to the direct sound included in the analysis frame length to obtain the low-frequency components of the direct sound. Information indicating the properties of the direct sound is derived from the energy or average amplitude of the low-frequency components, as in the above-described Equation 2.

[0464] If the energy of the low-frequency component of the n-th analysis frame is E ＬWhen the direct sound is expressed as (n), information I(n) indicating the properties of the direct sound is calculated according to the following formula:

[0465]

[0466] Here, the parameter c(i) represents a weighting coefficient. Generally, c(i) is set so that c(i) ≥ 0 and the sum of c(i) is 1. However, the method for setting c(i) is not limited to this.

[0467] Note that information I(n) indicating the properties of the direct sound is calculated every 5 ms that the direct sound is captured. In other words, it is possible to calculate the time variation of information I(n) indicating the properties of the direct sound with low delay. Therefore, this method is suitable for application requiring real-time performance.

[0468] Similarly to Equation 3, information I(n) indicating the properties of the direct sound may be calculated according to the following equation:

[0469]

[0470] Here, the parameter d(i) represents a weighting coefficient. Generally, d(i) is set so that d(i)≧0 and the sum of d(i) is 1. However, the method for setting d(i) is not limited to this.

[0471] In this equation, information I(n) indicating the properties of the direct sound is calculated recursively, making it possible to calculate the average energy over a long time period with a small amount of calculation.

[0472] The above equations 4 and 5 can be considered as filters in which E(n) is the input signal and I(n) is the output signal. In this case, equation 4 is a moving average (MA) model filter, and equation 5 is an autoregressive (AR) model filter, both of which have low-pass filter characteristics. Alternatively, an ARMA model filter, which is a combination of both, may be used.

[0473] In the above, a filter having low-pass characteristics is used to calculate the low-frequency components of the direct sound, but the method for calculating the low-frequency components of the direct sound is not limited to this. Furthermore, the method for deriving information indicating the time variation of the direct sound is not limited to the above-described formula or filter, and other known methods may be used. For example, the spectrum of the direct sound may be calculated by performing a frequency conversion on the direct sound. Then, the energy or average amplitude of the low-frequency components of the spectrum may be calculated.

[0474] In the above, the MA model or the AR model is used to derive the information indicating the time variation of the direct sound. The coefficients of these models may be predetermined fixed values ​​or may be variable values ​​that change over time.

[0475] The relationship between the analysis frame length and the interval at which the information update thread occurs may be as follows:

[0476] For example, if the time length of the analysis frame is TA (msec) and the occurrence interval of the information update thread is TU (msec), the value of N in the above (Equation 2) and (Equation 4) for the MA filter may be approximately the value given by TU / TA. Also, the values ​​of b(i) and d(i) (1≦i<N) in the above (Equation 3) and (Equation 5) for the AR filter may be values ​​such that the time constant of the filter is approximately TU (msec).

[0477] The reason for the above setting is that the filter is expected to converge within the interval period of information update.

[0478] On the other hand, if the value of the information indicating the time variation of the direct sound fluctuates too sharply with the above settings, I(n) may be calculated in advance. Then, the pre-calculated I(n) may be applied to the selection process of the reflected sounds. For example, I(t+tau) may be used in the processing of the t-th frame. Here, tau is a value determined according to the convergence characteristics of the filter. When convergence is slow, the value of tau is larger than when convergence is fast.

[0479] Furthermore, auditory masking (frequency masking) information calculated from the direct sound may be used as information indicating the characteristics of the direct sound. The auditory masking information indicates a threshold value for the amplitude value in the frequency domain that is masked by the direct sound. The amplitude value of the reflected sound in the same frequency domain may be compared with the threshold value, and processing may be performed to not select reflected sounds with amplitude values ​​smaller than the threshold value. The amplitude value of the reflected sound in the frequency domain may be acquired by the analysis unit 1301 as information indicating the characteristics of the reflected sound.

[0480] In this way, by setting the threshold value used to select reflected sounds according to the properties of the direct sound, it becomes possible to appropriately select reflected sounds that are auditorily necessary, and it becomes possible to effectively reflect the characteristics of hearing in the stereophonic sound reproduction system 1000. The process of detecting the properties of the direct sound, the process of determining the threshold value according to the properties, and the process of adjusting the threshold value according to the properties may be performed during the rendering process or before the rendering process starts.

[0481] For example, these processes may be performed when the virtual space is created (when the software is created), when processing of the virtual space starts (when the software is launched or rendering starts), or when an information update thread that occurs periodically in processing of the virtual space occurs, etc. Furthermore, when the virtual space is created may be when the virtual space is constructed before the start of acoustic processing, or when information about the virtual space (spatial information) is acquired, or when the software is acquired.

[0482] Here, in the information update thread, processing is performed to update the spatial information managed by the spatial information management units (1201, 1211).

[0483] The role of the information update thread is, for example, to update the position and orientation of the listener's avatar placed in the virtual space based on the position and orientation of the VR goggles worn by the listener, or to update the position of an object moving in the virtual space, etc. Such processing is handled within a processing thread that runs at a relatively low frequency of about several tens of Hz.

[0484] The process of updating information indicating the characteristics of the direct sound may be performed in such a processing thread that occurs less frequently. This is because the characteristics of the direct sound change less frequently than the frequency with which audio processing frames for audio output occur. This makes it possible to relatively reduce the computational load of this process. Furthermore, updating information at an unnecessarily high frequency poses a risk of generating pulsive noise. Updating information at a low frequency makes it possible to avoid such a risk.

[0485] (Third Modification of Threshold Setting Method) As another example of a method for setting a threshold, the threshold may be set according to the computational resources (CPU power, memory resources, PC performance, remaining battery power, etc.) used to process the reproduction of the virtual space. More specifically, the sensor 1405 of the audio signal processing device 1001 detects the amount of computational resources, and if the amount of computational resources is low, the threshold is set high. This makes the volume of more reflected sounds lower than the threshold, making it possible to reduce the amount of reflected sounds that are subjected to binaural processing and reduce the amount of computation.

[0486] Alternatively, when signal processing is performed in a device powered by a battery, such as a smartphone or VR goggles, it is expected that priority will be given to continuing processing for a long period of time and that computational resources will be saved. In such a case, the threshold may be set high without detecting the amount or remaining amount of computational resources.

[0487] (Fourth variant of threshold setting method) As another example of a threshold setting method, the audio signal processing device 1001 or the audio presentation device 1002 may be provided with a threshold setting unit (not shown), so that the threshold can be set by an administrator or listener of the virtual space.

[0488] For example, a listener wearing the audio presentation device 1002 may be able to select between an "energy saving mode" with less target reflected sounds and less computational effort, and a "high performance mode" with more target reflected sounds and more computational effort. Alternatively, the mode may be selectable by an administrator managing the stereophonic sound reproduction system 1000 or a creator of the stereophonic content. Alternatively, the threshold or threshold data may be directly selectable instead of the mode.

[0489] (First Modification of Operation of Rendering Unit) Fig. 23 is a flowchart showing a first modification of the operation of the audio signal processing device 1001. Fig. 23 shows the processing executed mainly by the rendering unit 1300 of the audio signal processing device 1001. In this modification, a volume compensation processing is added to the operation of the rendering unit 1300.

[0490] For example, the analysis unit 1301 acquires data (input signal) (S301). Next, the analysis unit 1301 analyzes the data (S302). Next, the selection unit 1302 determines whether or not to select reflected sounds based on the analysis results (S303). Next, the synthesis unit 1303 performs volume compensation processing based on the reflected sounds that are not selected (S304). Next, the synthesis unit 1303 performs acoustic processing on the direct sound and reflected sounds (S305). Then, the synthesis unit 1303 outputs the direct sound and reflected sounds as audio (S306).

[0491] Of the above processes (S301 to S306), the processes other than the volume compensation process (S304) are common to the other examples described above, and therefore description thereof will be omitted.

[0492] The volume compensation process is performed in response to reflected sounds that were not selected in the selection process. For example, a lack of perceived loudness occurs when reflected sounds are not selected in the selection process. The volume compensation process suppresses the sense of discomfort that accompanies such a lack of perceived loudness. The following two methods are disclosed as examples of methods for compensating for perceived loudness. Either of the two methods may be used.

[0493] First, we will explain a method for compensating for the sense of volume by increasing the volume of the direct sound. The synthesis unit 1303 generates a direct sound by increasing the volume of the direct sound by the amount of the volume of the unselected reflected sound. This compensates for the sense of volume that would be lost by not generating reflected sound.

[0494] When increasing the volume, the synthesis unit 1303 may increase the volume for each frequency component in accordance with the frequency characteristics of the reflected sound. To enable such processing, a volume attenuation rate at which the reflective object attenuates the volume may be assigned to each predetermined frequency band. This makes it possible to derive the frequency characteristics of the reflected sound.

[0495] Next, a method for compensating for the perceived loudness by synthesizing reflected sounds with direct sounds will be described. In this method, the synthesizer 1303 adds unselected reflected sounds to the direct sound to generate a direct sound, thereby compensating for the perceived loudness caused by not generating reflected sounds. The generated direct sound reflects the volume (amplitude), frequency, delay, etc. of the unselected reflected sounds.

[0496] In the case of the method of increasing the volume of direct sound, the amount of calculation required for the compensation process is extremely small, but only the volume is compensated. In the case of the method of combining direct sound with reflected sound, the amount of calculation required for the compensation process is greater than in the method of increasing the volume of direct sound, but the characteristics of the reflected sound are compensated more accurately.

[0497] In either case, the overall amount of calculation is reduced because only direct sound is generated, without generating reflected sound. In particular, the amount of calculation required for binaural processing, including the process of convolving HRTFs, is reduced, resulting in a significant reduction in the overall amount of calculation. This is because the amount of calculation required for binaural processing is far greater than the amount of calculation required for the compensation process described above.

[0498] If the reason why the reflected sound is not selected is that the volume of the reflected sound is below the masking threshold, the perceived volume is not lost, so the reflected sound may simply be removed without performing compensation processing.

[0499] (Second Modification of Operation of Rendering Unit) Fig. 24 is a flowchart showing a second modification of the operation of the audio signal processing device 1001. Fig. 24 shows the processing executed mainly by the rendering unit 1300 of the audio signal processing device 1001. In this modification, a left-right volume difference adjustment process is added to the operation of the rendering unit 1300.

[0500] For example, the analysis unit 1301 analyzes an input signal (S401). Next, the analysis unit 1301 detects the direction from which the sound is coming (S402). Next, the selection unit 1302 adjusts the difference in volume between the sounds perceived by the left and right ears (S403). The selection unit 1302 also adjusts the difference in arrival time (delay) between the sounds perceived by the left and right ears (S404). The selection unit 1302 determines whether to select a reflected sound based on the adjusted sound information (S405).

[0501] Of the above processes (S401 to S405), the processes other than left-right volume difference adjustment (S403) and delay adjustment (S404) are common to the other examples described above, and therefore description thereof will be omitted.

[0502] Fig. 25 is a diagram showing an example of the arrangement of an avatar, a sound source object, and an obstacle object. For example, when the front direction of the listener is 0 degrees, and the polarity (e.g., positive or negative) of the direction from which the direct sound comes and the direction from which the reflected sound comes are different, as shown in Fig. 25, the volume difference between the two ears is corrected.

[0503] Specifically, when the polarities of θ and γ are different, the ear that primarily (first) perceives the direct sound and the reflected sound is different. In this case, the selection unit 1302 adjusts the volume of the direct sound according to the position of the ear that primarily perceives the reflected sound, as the left-right volume difference adjustment (S403). For example, the selection unit 1302 attenuates the volume of the direct sound when it reaches the listener by multiplying the volume by (1.0-0.3 sin(θ)) (0≦θ≦180).

[0504] The selection unit 1302 calculates the volume ratio between the volume of the direct sound corrected as described above and the volume of the reflected sound, and compares the calculated volume ratio with a threshold value to determine whether to select the reflected sound. This corrects the volume difference that occurs between the two ears, more accurately derives the volume of the direct sound that affects the reflected sound, and more accurately determines whether to select the reflected sound.

[0505] Furthermore, in addition to adjusting the left-right volume difference (S403), the selection unit 1302 may also perform delay adjustment (S404) by delaying the arrival time of the direct sound in accordance with the position of the ear that perceives the reflected sound. Specifically, the selection unit 1302 may delay the arrival time of the direct sound by adding (a(sin θ+θ) / c) ms (where a is the radius of the head and c is the speed of sound) to the arrival time of the direct sound.

[0506] (Third Modification of the Operation of the Rendering Unit) A method of setting a threshold value according to the direction of arrival will be described.

[0507] Fig. 26 is a flowchart showing yet another example of the selection process. A description of the process common to the example in Fig. 14 will be omitted. In the example in Fig. 26, the selection unit 1302 selects reflected sounds using a threshold value according to the arrival direction.

[0508] Specifically, the selection unit 1302 calculates the direct sound arrival direction (θ) and the reflected sound arrival direction (γ) based on the avatar orientation, from the direct sound arrival path (pd), the reflected sound arrival path (pr), and the avatar orientation information D calculated by the analysis unit 1301. That is, the selection unit 1302 detects the direct sound arrival direction (θ) and the reflected sound arrival direction (γ) (S231). The orientation of the avatar corresponds to the orientation of the listener. The avatar orientation information D may be included in the input signal.

[0509] The selection unit 1302 uses three indexes including the direct sound arrival direction (θ), the reflected sound arrival direction (γ) and the time difference (T) to identify the threshold to be used in the selection process from a three-dimensional array such as that shown in FIG. 18 (S232).

[0510] As an example, a method for setting a threshold value used in the selection process when an avatar, a sound source object, and an obstacle object are arranged as shown in FIG. 25 will be described.

[0511] From the input signal, position information of the avatar, sound source object, and obstacle object, as well as avatar orientation information D, are obtained. Using this position information and orientation information D, the direction of the direct sound (θ) and the direction of the sound image of the reflected sound (γ) are calculated when the orientation of the avatar is set to 0 degrees. In the case of Figure 25, the direction of the direct sound (θ) is about 20 degrees, and the direction of the sound image of the reflected sound (γ) is about 265 degrees (-95 degrees).

[0512] 18, threshold values ​​are identified from an array region corresponding to the values ​​of the two directions (θ) and (γ) and the value of the time difference (T) calculated by the analysis unit 1301. If there is no index corresponding to the calculated values ​​of (θ), (γ), and (T), a threshold value corresponding to the closest index may be identified.

[0513] Alternatively, the threshold value may be determined by performing a process such as interpolation, extrapolation, or the like based on one or more threshold values ​​corresponding to one or more indexes close to the calculated values ​​of (θ), (γ), and (T). For example, a threshold value corresponding to (20°, 265°, T) may be determined based on four threshold values ​​corresponding to four indexes, namely, (0°, 225°, T), (0°, 270°, T), (45°, 225°, T), and (45°, 270°, T).

[0514] The selection process based on the difference between the angle (θ) of the direction from which the direct sound arrives and the angle (γ) of the direction from which the reflected sound arrives will be described.

[0515] For example, threshold data having the angular difference (Φ) between the arrival direction (θ) of the direct sound and the arrival direction (γ) of the reflected sound and the time difference (T) as a two-dimensional index array may be created and set in advance, as shown in Fig. 19. In this case, the angular difference (Φ) and the time difference (T) are referenced in the selection process. Alternatively, the angular difference (Φ) between the angle (θ) of the arrival direction of the direct sound and the angle (γ) of the arrival direction of the reflected sound may be calculated in the selection process, and the calculated angular difference (Φ) may be used to specify the threshold.

[0516] Alternatively, threshold data may be set that has, as an index array, a combination of the angle difference (Φ), the direction of arrival of the direct sound (θ), and the time difference (T), or a combination of the angle difference (Φ), the direction of arrival of the reflected sound (γ), and the time difference (T).

[0517] Alternatively, threshold data having the values ​​of (θ), (γ) and (T) as a three-dimensional index array as shown in FIG. 18 may be set.

[0518] (Fourth Modification of Operation of Rendering Unit) The processes performed by the analysis unit 1301, selection unit 1302, and composition unit 1303 described above may be performed as pipeline processes as described in, for example, Patent Document 3.

[0519] FIG. 27 is a block diagram showing an example of the configuration for the rendering unit 1300 to perform pipeline processing.

[0520] The rendering unit 1300 in Fig. 27 includes a reverberation processing unit 1311, an early reflection processing unit 1312, a distance attenuation processing unit 1313, a selection unit 1314, a generation unit 1315, and a binaural processing unit 1316. These multiple components may be configured from the multiple components of the rendering unit 1300 shown in Fig. 7, or may be configured from at least some of the multiple components of the audio signal processing device 1001 shown in Fig. 5.

[0521] Pipeline processing refers to dividing the process for applying sound effects into multiple processes and executing the multiple processes one by one in sequence. Each of the multiple processes performs, for example, signal processing on an audio signal or generation of parameters used in the signal processing.

[0522] The rendering unit 1300 may perform reverberation processing, early reflection processing, distance attenuation processing, binaural processing, and the like as pipeline processing. However, these processes are merely examples, and the pipeline processing may include other processes or may not include some of the processes. For example, the pipeline processing may include diffraction processing and occlusion processing. Furthermore, for example, reverberation processing may be omitted if it is not necessary.

[0523] Each process may be expressed as a stage. An audio signal such as a reflected sound generated as a result of each process may be expressed as a rendering item. The multiple stages in the pipeline process and their order are not limited to the example shown in FIG. 27 .

[0524] Here, the parameters used in the selection process (arrival paths, arrival times, and volume ratios for direct sound and reflected sound) are calculated in one of multiple stages for generating a rendering item. In other words, the parameters used to select reflected sounds are calculated as part of the pipeline processing for generating a rendering item. Note that not all stages need to be performed by the rendering unit 1300. For example, some stages may be omitted or may be performed by a unit other than the rendering unit 1300.

[0525] The following describes reverberation processing, early reflection processing, distance attenuation processing, selection processing, generation processing, and binaural processing that may be included as stages in the pipeline processing. At each stage, metadata included in the input signal may be analyzed to calculate parameters used to generate reflected sounds.

[0526] In the reverberation processing, the reverberation processor 1311 generates an audio signal indicating a reverberant sound or parameters used to generate an audio signal. A reverberant sound is a sound that arrives at a listener as reverberation after a direct sound. As an example, a reverberant sound is a sound that arrives at a listener after a relatively late stage (e.g., about 150 ms after the arrival of the direct sound) after an early reflected sound (described later) arrives at the listener, and after having been reflected more times (e.g., several tens of times) than an early reflected sound.

[0527] The reverberation processor 1311 refers to the audio signal and spatial information contained in the input signal, and calculates the reverberation sound using a predetermined function prepared in advance as a function for generating the reverberation sound.

[0528] The reverberation processor 1311 may generate reverberant sounds by applying a known reverberation generation method to the audio signal included in the input signal. An example of a known reverberation generation method is the Schroeder method, but known reverberation generation methods are not limited to the Schroeder method. Furthermore, when applying a known reverberation generation method, the reverberation processor 1311 uses the shape and acoustic characteristics of the sound reproduction space indicated by the spatial information. This allows the reverberation processor 1311 to calculate parameters for generating reverberant sounds.

[0529] In the early reflection process, the early reflection processor 1312 calculates parameters for generating early reflection sounds based on spatial information. The early reflection sounds are reflected sounds that arrive at the listener after one or more reflections at a relatively early stage after a direct sound from a sound source object arrives at the listener (for example, about several tens of milliseconds after the direct sound arrives).

[0530] The early reflection processing unit 1312 refers to, for example, the audio signal and metadata, and calculates the path of the reflected sound that travels from the sound source object to the listener after being reflected by the reflecting object. For example, the path calculation may use the shape of the three-dimensional sound field (space), the size of the three-dimensional sound field, the positions of reflecting objects such as structures, and the reflectance of the reflecting object.

[0531] The early reflection processing unit 1312 may also calculate the path of the direct sound. Information about the path may be used as a parameter by which the early reflection processing unit 1312 generates the early reflected sound, or may be used as a parameter by which the selection unit 1314 selects the reflected sound.

[0532] In the distance attenuation process, the distance attenuation processor 1313 calculates the volume of the direct sound and the reflected sound that reach the listener based on the path lengths of the direct sound and the reflected sound. The volume of the direct sound and the reflected sound that reach the listener attenuates in proportion to the distance of the path to the listener (inversely proportional to the distance) relative to the volume of the sound source. Therefore, the distance attenuation processor 1313 can calculate the volume of the direct sound by dividing the volume of the sound source by the path length of the direct sound, and can calculate the volume of the reflected sound by dividing the volume of the sound source by the path length of the reflected sound.

[0533] In the selection process, the selection unit 1314 selects a generation target reflected sound based on parameters calculated before the selection process. Any of the selection methods disclosed herein may be used to select the generation target reflected sound.

[0534] The selection process may be performed on all reflected sounds, or may be performed only on reflected sounds with high evaluation values ​​based on the evaluation process as described above. In other words, reflected sounds with low evaluation values ​​may be determined not to be selected without even undergoing the selection process. For example, a reflected sound with a very low volume may be considered to have a low evaluation value and may be determined not to be selected.

[0535] Alternatively, for example, a selection process may be performed on all reflected sounds, and the evaluation values ​​of the reflected sounds selected in the selection process may be determined, and reflected sounds with low evaluation values ​​may be re-determined as not being selected.

[0536] The selection process and the evaluation process may be performed independently or in combination. When the selection process and the evaluation process are performed in combination, either of the two processes may be performed first.

[0537] In the generation process, the generation unit 1315 generates direct sound and reflected sound. For example, the generation unit 1315 generates direct sound from an audio signal included in the input signal based on the arrival time and volume of the direct sound at the time of arrival. Furthermore, for the reflected sound selected in the selection process, the generation unit 1315 generates reflected sound from an audio signal included in the input signal based on the arrival time and volume of the reflected sound at the time of arrival.

[0538] In the binaural processing, the binaural processing unit 1316 performs signal processing so that the audio signal of the direct sound is perceived by the listener as a sound arriving from the direction of the sound source object. Furthermore, the binaural processing unit 1316 performs signal processing so that the reflected sound selected by the selection unit 1314 is perceived by the listener as a sound arriving from the reflecting object.

[0539] For example, the binaural processing unit 1316 performs processing to apply the HRIR DB based on the position and orientation of the listener in the sound space so that sound arrives at the listener from the position of a sound source object or the position of an obstacle object.

[0540] HRIR (Head-Related Impulse Responses) is a response characteristic when one impulse is generated. Specifically, HRIR is a response characteristic obtained by converting a head-related transfer function, which represents changes in sound caused by surrounding objects including the auricle, the human head, and shoulders, from a frequency domain representation to a time domain representation by Fourier transform. The HRIR DB is a database containing such information.

[0541] Furthermore, the position and orientation of the listener in the sound space are, for example, the position and orientation of the virtual listener in the virtual sound space. The position and orientation of the virtual listener in the virtual sound space may change in accordance with the movement of the listener's head. The position and orientation of the virtual listener in the virtual sound space may also be determined based on information acquired from the sensor 1405.

[0542] The programs, spatial information, HRIR DB, threshold data, and other parameters used in the above processing are obtained from the memory 1404 provided in the audio signal processing device 1001 or from outside the audio signal processing device 1001.

[0543] The pipeline processing may also include other processes. The rendering unit 1300 may also include processing units (not shown) for performing other processes included in the pipeline processing. For example, the rendering unit 1300 may include a diffraction processing unit and an occlusion processing unit.

[0544] The diffraction processing unit executes processing to generate an audio signal representing a sound including diffracted sound caused by an obstacle object between the listener and the sound source object in a three-dimensional sound field (space). When an obstacle object exists between the sound source object and the listener, the diffracted sound is a sound that travels from the sound source object to the listener, going around the obstacle object.

[0545] The diffraction processing unit calculates a path of the diffracted sound from the sound source object to the listener, bypassing the obstacle object, and generates the diffracted sound based on the path, for example, by referring to the audio signal and metadata. The path calculation may use the positions of the sound source object, the listener, and the obstacle object in the three-dimensional sound field (space), as well as the shape and size of the obstacle object.

[0546] When a sound source object is present on the other side of an obstacle object, the occlusion processing unit generates an audio signal of the sound that leaks from the sound source object and passes through the obstacle object based on spatial information and information such as the material of the obstacle object.

[0547] (Example of a Sound Source Object) In the above, the position information assigned to the sound source object indicates a "point" in the virtual space as the position of the sound source object. That is, in the above, the sound source is defined as a "point sound source."

[0548] On the other hand, a sound source in a virtual space may be defined as an object having length, size, shape, etc., i.e., as a spatially extended sound source rather than a point sound source. In this case, the distance between the listener and the sound source and the direction from which the sound is coming are not determined. Therefore, reflected sounds caused by such sound sources may be limited to those selected by the selection unit 1302 without being analyzed by the analysis unit 1301 or regardless of the analysis results. This makes it possible to avoid deterioration in sound quality that may occur when reflected sounds are not selected.

[0549] Alternatively, a representative point such as the center of gravity of the object may be determined, and the processing of the present disclosure may be applied on the assumption that the sound is generated from that representative point. In this case, the threshold may be adjusted according to information on the spatial extent of the sound source.

[0550] (Examples of Direct Sound and Reflected Sound) For example, direct sound is sound that is not reflected by a reflecting object, and reflected sound is sound that is reflected by a reflecting object. Direct sound may be sound that arrives at the listener from a sound source without being reflected by a reflecting object, or reflected sound may be sound that arrives at the listener from a sound source after being reflected by a reflecting object.

[0551] Furthermore, the direct sound and the reflected sound are not limited to sounds that have arrived at the listener, but may be sounds that have not yet arrived at the listener. For example, the direct sound may be sounds output from a sound source, or in other words, sounds from the sound source.

[0552] 28 is a diagram showing sound transmission and diffraction. As shown in FIG. 28, there are cases where direct sound does not reach the listener due to the presence of an obstacle object between the sound source object and the listener. In this case, sound emitted from the sound source object, transmitted through the obstacle object, and reached the listener may be considered as direct sound. Meanwhile, sound emitted from the sound source object, diffracted by the obstacle object, and reached the listener may be considered as reflected sound.

[0553] Furthermore, the two sounds compared in the selection process are not limited to a direct sound and a reflected sound based on a sound emitted from a single sound source. For example, a sound may be selected by comparing two reflected sounds based on a sound emitted from a single sound source. In this case, the direct sound in the present disclosure may be interpreted as the sound that reaches the listener first, and the reflected sound in the present disclosure may be interpreted as the sound that reaches the listener later.

[0554] (Example of Bitstream Structure) A bitstream includes, for example, an audio signal and metadata. The audio signal is sound data that expresses sound, and indicates information about the frequency and intensity of the sound. The metadata includes spatial information about the sound space, which is the space of the sound field.

[0555] For example, the spatial information is information about a space in which a listener who listens to a sound based on an audio signal is located. Specifically, the spatial information is information about a predetermined position (localization position) for localizing a sound image at a predetermined position in a sound space (e.g., a three-dimensional sound field), that is, for allowing the listener to perceive a sound arriving from a direction corresponding to the predetermined position. The spatial information includes, for example, sound source object information and position information indicating the position of the listener.

[0556] The sound source object information is information about a sound source object that generates a sound based on an audio signal. That is, the sound source object information is information about an object (sound source object) that reproduces an audio signal, and is information about a virtual sound source object that is placed in a virtual sound space. Here, the virtual sound space may correspond to a real space in which an object that generates a sound is placed, and the sound source object in the virtual sound space may correspond to an object that generates a sound in the real space.

[0557] The sound source object information may indicate the position of the sound source object arranged in the sound space, the orientation of the sound source object, the directivity of the sound emitted by the sound source object, whether the sound source object belongs to a living thing or not, whether the sound source object is a moving object or not, etc. For example, the audio signal is associated with one or more sound source objects indicated by the sound source object information.

[0558] The bitstream has a data structure that is made up of, for example, metadata (control information) and an audio signal.

[0559] The audio signal and metadata may be contained in a single bitstream or in separate bitstreams, or may be contained in a single file or in separate files.

[0560] A bitstream may exist for each sound source or for each playback time. Even if a bitstream exists for each playback time, multiple bitstreams may be processed in parallel at the same time.

[0561] Metadata may be assigned to each bitstream, or may be assigned to multiple bitstreams together as information for controlling multiple bitstreams. In this case, multiple bitstreams may share the same metadata. Metadata may also be assigned for each playback time.

[0562] When multiple bitstreams or multiple files exist, one or more of the bitstreams or files may contain information indicating the associated bitstreams or files, or alternatively, each of all of the bitstreams or each of all of the files may contain information indicating the associated bitstreams or files.

[0563] Here, the related bitstreams or related files are, for example, bitstreams or files that may be used simultaneously during audio processing, and may also include bitstreams or files that collectively describe information indicating related bitstreams or related files.

[0564] Here, the information indicating the related bitstream or related file may be, for example, an identifier indicating the related bitstream or related file. Alternatively, the information indicating the related bitstream or related file may be, for example, a file name indicating the related bitstream or related file, a URL (Uniform Resource Locator), or a URI (Uniform Resource Identifier).

[0565] In this case, the acquisition unit identifies and acquires the related bitstream or related file based on the information indicating the related bitstream or related file. Alternatively, the bitstream or file may contain information indicating the related bitstream or related file, and another bitstream or another file may contain information indicating the related bitstream or related file.

[0566] Here, the file containing information indicating the associated bitstream or associated file may be a control file such as a manifest file used for content distribution.

[0567] Note that all or part of the metadata may be obtained from sources other than the bitstream of the audio signal. For example, either the metadata for controlling the sound or the metadata for controlling the video may be obtained from sources other than the bitstream, or both may be obtained from sources other than the bitstream.

[0568] Furthermore, metadata for controlling the video may be included in the bitstream acquired by the stereophonic sound reproduction system 1000. In this case, the stereophonic sound reproduction system 1000 may output the metadata for controlling the video to a display device that displays images or a stereophonic video reproduction device that reproduces the stereophonic video.

[0569] (Examples of Information Included in Metadata) Metadata may be information used to describe a scene expressed in sound space, where a scene is a term that refers to a collection of all elements that represent three-dimensional video and sound events in sound space that are modeled by the stereophonic sound reproduction system 1000 using the metadata.

[0570] That is, the metadata may include not only information for controlling audio processing but also information for controlling video processing. The metadata may include only one of information for controlling audio processing and information for controlling video processing, or may include both.

[0571] The stereophonic sound reproduction system 1000 generates virtual sound effects by performing sound processing on audio signals using metadata included in the bitstream and additionally acquired interactive listener position information, etc. Among the sound effects, early reflection processing, obstacle processing, diffraction processing, blocking processing, and reverberation processing may be performed, and other sound processing may be performed using the metadata. For example, sound effects such as distance attenuation, localization, or Doppler effect may be added.

[0572] Furthermore, information on switching on / off all or part of the sound effects, or priority information for multiple sound effect processes may be added to the metadata.

[0573] As an example, the metadata includes information about a sound space including sound source objects and obstacle objects, and information about a positioning position for localizing a sound image at a predetermined position within the sound space (i.e., allowing the listener to perceive sound coming from a predetermined direction).

[0574] Here, an obstacle object is an object that may affect the sound perceived by the listener by, for example, blocking or reflecting the sound emitted by the sound source object before it reaches the listener. Obstacle objects may include not only stationary objects but also moving objects such as animals or machines. The animal may also be a person, etc.

[0575] Furthermore, when multiple sound source objects exist in a sound space, other sound source objects can be obstacle objects for any sound source object. In other words, both non-sound-emitting objects, such as building materials or inanimate objects, which do not emit sound, and sound source objects that emit sound can be obstacle objects.

[0576] The metadata includes information that represents all or part of the shape of the sound space, the shape and position of obstacle objects in the sound space, the shape and position of sound source objects in the sound space, and the position and orientation of the listener in the sound space.

[0577] The sound space may be either a closed space or an open space. The metadata may also include information indicating the reflectance of obstacle objects that may reflect sound in the sound space. For example, the floor, walls, or ceiling that form the boundaries of the sound space may also constitute obstacle objects.

[0578] The reflectance is the ratio of the energy of reflected sound to incident sound, and may be set for each frequency band of sound. Of course, the reflectance may be set uniformly regardless of the frequency band of sound. Note that when the sound space is an open space, parameters such as a uniform attenuation rate, diffracted sound, and early reflected sound may be used.

[0579] The metadata may include information other than reflectance as a parameter related to an obstacle object or a sound source object. For example, the metadata may include information related to the material of the object as a parameter related to both a sound source object and a non-sound-producing object. Specifically, the metadata may include information such as diffusion rate, transmittance, and sound absorption rate.

[0580] The information about the sound source object may include information indicating the volume, radiation characteristics (directivity), playback conditions, the number and type of sound sources in one object, and the sound source area in the object. The playback conditions may determine, for example, whether the sound is a continuous sound or an event-triggering sound. The sound source area in the object may be determined based on the relative relationship between the position of the listener and the position of the object, or may be determined using the object as a reference.

[0581] For example, if the sound source area is defined relative to the position of the listener and the position of the object, it is possible for the listener to perceive sound A coming from the right side of the object and sound B coming from the left side of the object.

[0582] Furthermore, when a sound source region is defined using an object as a reference, it is possible to fix which region of the object will emit which sound. For example, when a listener views an object from the front, it is possible for the listener to perceive a high-pitched sound from the right side of the object and a low-pitched sound from the left side of the object. When a listener views an object from the back, it is possible for the listener to perceive a low-pitched sound from the right side of the object and a high-pitched sound from the left side of the object.

[0583] The spatial metadata may include the time to early reflections, the reverberation time, the ratio of direct sound to diffuse sound, etc. If the ratio of direct sound to diffuse sound is zero, the listener will perceive only direct sound.

[0584] (Determining and Utilizing Threshold) For example, as described above, a threshold for the volume ratio between direct sound and reflected sound is determined from the time difference between the direct sound and reflected sound. The threshold may be determined by the selection unit 1302, the threshold adjustment unit 1304, or another component. For example, the threshold may be determined by a threshold determination unit or a threshold determination device. The threshold determination unit or the threshold determination device may be included in the selection unit 1302 or the threshold adjustment unit 1304, or may be a component or device separate from the selection unit 1302 and the threshold adjustment unit 1304.

[0585] Specifically, the threshold specifying unit or the threshold specifying device may include an acquisition unit and a specification unit. The acquisition unit acquires the time difference between a preceding sound such as a direct sound and a subsequent sound such as a reflected sound. The specification unit specifies a threshold for the volume ratio between the direct sound and the reflected sound from the time difference between the direct sound and the reflected sound, and outputs the threshold. The audio signal processing device 1001 may operate as the threshold specifying device. Furthermore, multiple components included in the threshold specifying unit or the threshold specifying device may be implemented by the processor 1402, the memory 1404, etc.

[0586] The threshold specifying device may specify the threshold using the above-described Equation 1. In this case, the time difference is the time difference between the arrival time of a preceding sound and the arrival time of a following sound that occurs in response to the preceding sound. The threshold corresponds to a boundary between whether or not the following sound is affected by the preceding sound effect exerted by the preceding sound. The specifying unit specifies, as the threshold, a volume ratio corresponding to the time difference on a boundary line defined as a line indicating the boundary in an area having a first axis of time difference and a second axis of volume ratio.

[0587] Here, the domain of the boundary line for the first axis corresponds to the time range in which the precedence effect occurs. When the first axis is the horizontal axis and the second axis is the vertical axis, the boundary line is a curve that slopes downward to the right and is convex upward in the domain.

[0588] 29 is a graph showing thresholds. Specifically, the thresholds specified by the above-described Equation 1 are graphed. The horizontal axis corresponds to the time difference (ms) between the arrival time of a direct sound and the arrival time of its reflected sound, and the vertical axis corresponds to the volume ratio (dB) between the volume of the direct sound and the volume of the reflected sound. The boundary shown in the graph indicates the boundary as to whether or not a subsequent sound is affected by the preceding sound effect exerted by the preceding sound, that is, the boundary (threshold) as to whether or not a reflected sound is perceived.

[0589] For example, the threshold specifying device stores information on the graph shown in Fig. 29 in an internal memory, and specifies a threshold based on the information stored in the internal memory. The threshold can be used for the following purposes. The threshold specifying device may perform the following process to use the threshold, or the audio signal processing device 1001 may perform the following process to use the threshold.

[0590] 30 is a diagram showing a first application example of the threshold. Specifically, reflected sounds located below the boundary line are not selected (i.e., disabled) because they are not perceived. On the other hand, reflected sounds located above the boundary line are selected (i.e., enabled) because they are perceived. This application example corresponds to the example described above using FIG. 14 etc.

[0591] 31 is a diagram showing a second example of how the threshold value is used, in which a threshold value is used to determine which of a plurality of reflected sounds generated in a virtual space is to be selected.

[0592] For example, if the time difference between the i-th reflected sound and its direct sound among multiple reflected sounds is defined as D(i) and their volume ratio is defined as G(i), the distance between point (D(i), G(i)) and the boundary line is calculated. Then, N reflected sounds that are the longest distance upward from the boundary line are selected (i.e., validated). The other reflected sounds are not selected (i.e., invalidated).

[0593] Of the multiple reflected sounds above the boundary line, N reflected sounds may be selected in order of distance. Alternatively, a positive sign may be assigned to the distance when point (D(i), G(i)) is above the boundary line, and a negative sign may be assigned to the distance when point (D(i), G(i)) is below the boundary line, and N reflected sounds may be selected in order of distance.

[0594] The above-described process makes it possible to select reflected sounds in descending order of perceptual priority. Therefore, even if the computational power of the computing means realizing the virtual space is low and the number of reflected sounds to be selected is reduced, reducing the amount of computation required, it is still possible to select reflected sounds that are perceptually effective.

[0595] 32 is a diagram showing a third example of the use of a threshold value. In this example, a threshold value is used to indicate whether or not a reflected sound to be added (selected) has perceptual value when adding (selecting) a reflected sound to be used in the production of a virtual space.

[0596] For example, the time difference between a reflected sound to be added and its direct sound is defined as D(i), and the volume ratio between them is defined as G(i). If point (D(i), G(i)) is located below the boundary line, the reflected sound will not be perceived even if added, and therefore it can be taught that the reflected sound has no value. Alternatively, if the time difference between a reflected sound and its direct sound is specified, the volume of the reflected sound that is effective for that time difference can be taught.

[0597] When a reflected sound is newly added as a sound effect in a virtual space, the audio signal processing device 1001 may, as described above, indicate to the administrator whether the reflected sound to be added has perceptual value. Alternatively, the audio signal processing device 1001 may indicate to the administrator whether a reflected sound generated in accordance with the settings of an object or the like in the virtual space has perceptual value.

[0598] 33 is a diagram showing a fourth example of use of a threshold value. In this example, a threshold value is used to indicate whether a reflected sound to be added (selected) for rendering a virtual space is a reflected sound that compresses the dynamic range.

[0599] For example, if the number of reflected sounds to be added increases, the total amplitude of the multiple sounds generated in the virtual space increases. Typically, the multiple sounds generated in the virtual space are ultimately downmixed into two-channel signals (left and right) and listened to through headphones, etc. Therefore, if multiple reflected sounds are added excessively, the total amplitude of the downmixed signal will overflow.

[0600] By using a threshold value, it becomes possible to quantitatively teach, among multiple reflected sounds that are effective for perception, which reflected sounds compress the dynamic range with only a small number of reflected sounds added, and which reflected sounds do not compress the dynamic range even with a large number of reflected sounds added.

[0601] When a reflected sound is newly added as a sound effect in a virtual space, the audio signal processing device 1001 may indicate to the administrator whether the reflected sound to be added will compress the dynamic range, as described above. Alternatively, the audio signal processing device 1001 may indicate to the administrator whether a reflected sound generated in accordance with the settings of an object or the like in the virtual space will compress the dynamic range.

[0602] 34 is a flowchart showing a first basic operation example of the audio signal processing device 1001. For example, in the audio signal processing device 1001, the processor 1402 operating as a circuit performs the operation shown in FIG.

[0603] Specifically, the processor 1402 acquires sound space information relating to a sound space (S1401). Then, based on the sound space information, the processor 1402 acquires a volume ratio between the volume of a direct sound generated from a sound source in the sound space and the volume of a reflected sound generated in the sound space corresponding to the direct sound (S1402). The processor 1402 also acquires information relating to additional processing (S1403). Then, based on the information relating to additional processing and the volume ratio, the processor 1402 controls whether or not to select a reflected sound (S1404).

[0604] This allows the audio signal processing device 1001 to appropriately control whether to select a reflected sound based on information related to additional processing and the volume ratio between the direct sound and the reflected sound. That is, it becomes possible to appropriately control whether to select a reflected sound generated in response to a direct sound generated from a sound source in a sound space. Therefore, it becomes possible to appropriately reduce the amount of calculation and the calculation load.

[0605] For example, the processor 1402 may adjust the volume ratio according to the information related to the additional processing. Then, the processor 1402 may control whether or not to select a reflected sound based on the adjusted volume ratio. This allows the audio signal processing device 1001 to appropriately control whether or not to select a reflected sound based on the volume ratio that reflects the effect of the additional processing.

[0606] Furthermore, for example, the processor 1402 may calculate the time at which a sound emitted from a sound source arrives as a reflected sound. Then, the processor 1402 may control whether to select a reflected sound based on the calculated time, information related to additional processing, and the volume ratio. This allows the audio signal processing device 1001 to more appropriately select a reflected sound that has a large influence on the listener's perception based on the arrival time of the reflected sound, information related to additional processing, and the volume ratio between the direct sound and the reflected sound.

[0607] Furthermore, for example, the processor 1402 may calculate the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information, and may control whether or not to select the reflected sound based on the calculated time difference, information related to additional processing, and the volume ratio.

[0608] This allows the audio signal processing device 1001 to more appropriately select reflected sounds that have a large influence on the listener's perception, based on the time difference between the arrival time of the direct sound and the arrival time of the reflected sound, information related to additional processing, and the volume ratio between the direct sound and the reflected sound. Therefore, the audio signal processing device 1001 can more appropriately select reflected sounds that have a large influence on the listener's perception, based on the precedence effect.

[0609] Furthermore, for example, the processor 1402 may calculate the time difference between the end time of the direct sound and the arrival time of the reflected sound based on the sound space information, and may control whether or not to select the reflected sound based on the calculated time difference, information related to additional processing, and the volume ratio.

[0610] This allows the audio signal processing device 1001 to more appropriately select reflected sounds that have a large influence on the listener's perception, based on the time difference between the end time of the direct sound and the arrival time of the reflected sound, information related to additional processing, and the volume ratio between the direct sound and the reflected sound. Therefore, the audio signal processing device 1001 can more appropriately select reflected sounds that have a large influence on the listener's perception, based on the post-masking effect.

[0611] Furthermore, for example, the additional processing may be processing that involves increasing or decreasing the amplitude of the reflected sound. The information related to the additional processing may be information indicating the amount of increase or decrease in the amplitude in the processing that involves increasing or decreasing the amplitude of the reflected sound. The processor 1402 may adjust the volume ratio according to the amount of increase or decrease in the amplitude. The processor 1402 may then control whether or not to select the reflected sound based on the adjusted volume ratio.

[0612] This allows the audio signal processing device 1001 to appropriately adjust the volume ratio according to the amplitude that increases or decreases due to the additional processing, and the audio signal processing device 1001 can appropriately control whether or not to select reflected sounds based on the volume ratio adjusted by the additional processing.

[0613] Furthermore, for example, the additional processing may be processing that involves increasing or decreasing the delay of the reflected sound. The information related to the additional processing may be information indicating the amount of increase or decrease in the delay in the processing that involves increasing or decreasing the delay of the reflected sound. The processor 1402 may adjust the time difference according to the amount of increase or decrease in the delay. The processor 1402 may then control whether or not to select the reflected sound based on the adjusted time difference and volume ratio.

[0614] This allows the audio signal processing device 1001 to appropriately adjust the time difference according to the delay that increases or decreases due to the additional processing, and the audio signal processing device 1001 can appropriately control whether or not to select reflected sounds based on the time difference adjusted by the additional processing.

[0615] Furthermore, for example, the additional processing may be processing that is not applied to the direct sound. This allows the audio signal processing device 1001 to apply the additional processing to the reflected sound without applying it to the direct sound. In other words, the audio signal processing device 1001 can apply the additional processing that is not applied to the direct sound to the reflected sound. Therefore, the audio signal processing device 1001 can apply the additional processing that is suitable for the reflected sound to the reflected sound.

[0616] Furthermore, for example, the additional processing may be a filter process that auditorily emphasizes the diffusion of sound. This allows the audio signal processing device 1001 to appropriately control whether to select a reflected sound based on information related to the filter process that auditorily emphasizes the diffusion of sound and the volume ratio between the direct sound and the reflected sound.

[0617] Furthermore, for example, if the information on the additional processing indicates that the additional processing is not to be performed, the volume ratio does not need to be adjusted. This allows the audio signal processing device 1001 to appropriately control whether to adjust the volume ratio depending on whether to perform the additional processing.

[0618] Furthermore, for example, if the information on the additional processing indicates that the additional processing is not to be performed, the time difference does not need to be adjusted. This allows the audio signal processing device 1001 to appropriately control whether to adjust the time difference depending on whether to perform the additional processing.

[0619] 35 is a flowchart showing a second basic operation example of the audio signal processing device 1001. For example, in the audio signal processing device 1001, the processor 1402 operating as a circuit performs the operation shown in FIG.

[0620] Specifically, the processor 1402 acquires sound space information relating to a sound space (S1501). Then, the processor 1402 determines whether or not a sound source in the sound space is a point sound source based on the sound space information (S1502).

[0621] If the sound source is determined to be a point sound source (Yes in S1502), the processor 1402 acquires the volume ratio between the volume of the direct sound generated from the sound source and the volume of the reflected sound generated in the sound space corresponding to the direct sound based on the sound space information (S1503). The processor 1402 also calculates the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information (S1504). The processor 1402 then determines whether to select the reflected sound based on the volume ratio and the time difference (S1505).

[0622] On the other hand, if it is determined that the sound source is not a point sound source (No in S1502), the processor 1402 selects the reflected sound regardless of the volume ratio and time difference (S1506).

[0623] As a result, even if the distance between the listener and the sound source is not determined and it is difficult to appropriately calculate the volume ratio and time difference between the direct sound and the reflected sound, the audio signal processing device 1001 can select the reflected sound regardless of the volume ratio and time difference. Therefore, the audio signal processing device 1001 can suppress deterioration of sound quality that can occur when the sound source is not a point sound source.

[0624] 36 is a flowchart showing a third basic operation example of the audio signal processing device 1001. For example, in the audio signal processing device 1001, the processor 1402 operating as a circuit performs the operation shown in FIG.

[0625] Specifically, the processor 1402 acquires the time difference between the arrival time of the preceding sound and the arrival time of the subsequent sound generated in response to the preceding sound (S1601). Then, based on the time difference, the processor 1402 identifies a threshold value for the volume ratio of the volume of the subsequent sound to the volume of the preceding sound, which corresponds to the boundary between whether the subsequent sound is affected by the preceding sound effect of the preceding sound, and outputs the threshold value (S1602).

[0626] In identifying the threshold, a volume ratio corresponding to the acquired time difference is identified as a threshold on a boundary line defined as a line indicating whether or not a following sound is affected by the precedence effect in an area having a first axis of time difference and a second axis of volume ratio. Here, the domain of the boundary line for the first axis is the time range in which the precedence effect occurs.

[0627] This allows the audio signal processing device 1001 to appropriately specify a threshold value corresponding to the boundary of the precedence effect based on the relationship between the time difference between the leading sound and the following sound and the volume ratio between the leading sound and the following sound. Furthermore, the audio signal processing device 1001 can efficiently specify a threshold value for the time range in which the precedence effect occurs.

[0628] For example, when the first axis is the horizontal axis and the second axis is the vertical axis, the boundary line may be a curve that slopes downward to the right and convex upward in the domain. This allows the audio signal processing device 1001 to appropriately specify the threshold value based on the boundary characteristics that the larger the time difference, the smaller the volume ratio and the larger the rate of decrease in the volume ratio relative to an increase in the time difference. Therefore, the audio signal processing device 1001 can provide an appropriate threshold value for determining whether a subsequent sound is affected by the precedence effect.

[0629] Furthermore, for example, the boundary line may be defined by a mathematical formula having a time difference as a variable. Then, the threshold may be calculated based on the mathematical formula. This allows the audio signal processing device 1001 to efficiently identify the threshold based on the mathematical formula. Therefore, the audio signal processing device 1001 can efficiently provide the threshold.

[0630] Here, in the audio signal processing device 1001, the processor 1402 operating as a circuit performs the operations shown in Fig. 36 using the memory 1404, but in other devices such as a threshold value specifying device, a circuit may perform the operations shown in Fig. 36 using a memory. Then, the specified threshold value may be provided to the audio signal processing device 1001 or other devices.

[0631] In the above-described examples, the memory 1404 may store a program, a bitstream, sound space information, threshold data (boundary data), etc. The processor 1402 may refer to the information stored in the memory 1404 during operation, or may acquire information from the memory 1404. The processor 1402 may also store the information obtained through operation in the memory 1404.

[0632] (Supplementary Note) The aspects grasped based on the present disclosure are not limited to the embodiments, and may be implemented with various modifications.

[0633] For example, a process performed by a specific component in the embodiment may be performed by another component instead of the specific component. Also, the order of multiple processes may be changed, or multiple processes may be performed in parallel.

[0634] Furthermore, ordinal numbers such as first and second used in the description may be rearranged, removed, or newly added as appropriate. These ordinal numbers do not necessarily correspond to a meaningful order, but may be used to identify elements.

[0635] Furthermore, for example, in comparison with a threshold, "greater than or equal to the threshold" and "greater than the threshold" may be interpreted interchangeably. Similarly, "equal to or less than the threshold" and "smaller than the threshold" may be interpreted interchangeably. Furthermore, for example, "time" and "hour" may be interpreted interchangeably.

[0636] Furthermore, in the process of selecting one or more processing target sounds from a plurality of sounds, if there is no sound that satisfies the conditions, then none of the sounds may be selected as the processing target sounds. In other words, the process of selecting one or more processing target sounds from a plurality of sounds may include cases in which no processing target sounds are selected.

[0637] Also, for example, reference to at least one of a first element, a second element, and a third element may correspond to the first element, the second element, the third element, or any combination thereof.

[0638] Furthermore, for example, in the embodiments, the cases where the aspects understood based on the present disclosure are implemented as an audio processing device, an encoding device, or a decoding device are described. However, the aspects understood based on the present disclosure are not limited to these, and may be implemented as software for executing an audio processing method, an encoding method, or a decoding method.

[0639] For example, a program for executing the above-described acoustic processing method, encoding method, or decoding method may be stored in advance in a ROM, and the CPU may operate in accordance with the program.

[0640] Furthermore, a program for executing the above-described acoustic processing method, encoding method, or decoding method may be stored in a computer-readable recording medium, and the computer may then record the program stored in the recording medium into its RAM and operate in accordance with the program.

[0641] Each of the above components may be realized as an LSI, which is typically an integrated circuit having input and output terminals. These may be individually integrated into a single chip, or a single chip may include all or some of the components of the embodiments. The LSI may be expressed as an IC, a system LSI, a super LSI, or an ultra LSI depending on the degree of integration.

[0642] Furthermore, the present invention is not limited to LSIs, and dedicated circuits or general-purpose processors may also be used. Furthermore, FPGAs, which can be programmed after LSI manufacturing, or reconfigurable processors, which allow the connection or settings of circuit cells within the LSI to be reconfigured, may also be used. Furthermore, if an integrated circuit technology that replaces LSIs emerges due to advances in semiconductor technology or other derived technologies, that technology may naturally be used to integrate components. The application of biotechnology, etc., is also a possibility.

[0643] Furthermore, the FPGA, CPU, etc. may download all or part of the software for realizing the acoustic processing method, encoding method, or decoding method described in the present disclosure via wireless or wired communication. Furthermore, all or part of the software for updating may be downloaded via wireless or wired communication. The FPGA, CPU, etc. may then store the downloaded software in memory and operate based on the stored software to perform the digital signal processing described in the present disclosure.

[0644] In this case, the device equipped with the FPGA or CPU may be connected to the signal processing device wirelessly or via a wire, or may be connected to the signal processing server via a network, and this device and the signal processing device or the signal processing server may perform the acoustic processing method, encoding method, or decoding method described in the present disclosure.

[0645] For example, the sound processing device, encoding device, or decoding device in the present disclosure may include an FPGA, a CPU, etc. Furthermore, the sound processing device, encoding device, or decoding device may include an interface for externally obtaining software for operating the FPGA, CPU, etc., and a memory for storing the obtained software. Then, the FPGA, CPU, etc. may operate based on the stored software to perform the signal processing described in the present disclosure.

[0646] A server may provide software related to the acoustic processing, encoding processing, or decoding processing of the present disclosure. Then, a terminal or device may operate as the acoustic processing device, encoding device, or decoding device described in the present disclosure by installing the software. Note that the terminal or device may connect to the server via a network and install the software.

[0647] Furthermore, a device other than the terminal or device may connect to a server via a network to acquire data for installing the software, and the other device may provide the data for installing the software to the terminal or device, thereby installing the software in the terminal or device. Note that an example of the software may be VR software or AR software for causing a terminal or device to execute the acoustic processing method described using the embodiment.

[0648] In the above-described embodiments, each component may be configured with dedicated hardware, or may be realized by executing a software program suitable for each component. Each component may be realized by a program execution unit such as a CPU or processor reading and executing a software program recorded on a recording medium such as a hard disk or semiconductor memory.

[0649] Although the devices and the like according to one or more aspects have been described above based on the embodiments, the aspects grasped based on the present disclosure are not limited to the embodiments. As long as they do not deviate from the spirit of the present disclosure, forms obtained by applying various modifications conceivable by a person skilled in the art to the embodiments, and forms constructed by combining components in different modifications, may also be included within the scope of one or more aspects.

[0650] (Additional Notes) The above description of the embodiments discloses the following techniques.

[0651] (Technology 1) A sound processing device comprising a circuit and a memory, wherein the circuit uses the memory to acquire sound space information relating to a sound space, acquires characteristics relating to a first sound generated from a sound source in the sound space based on the sound space information, and controls whether to select a second sound generated in the sound space corresponding to the first sound based on the characteristics relating to the first sound.

[0652] (Technology 2) The sound processing device according to Technology 1, wherein the first sound is a direct sound and the second sound is a reflected sound.

[0653] (Technology 3) The characteristic related to the first sound is a volume ratio between the volume of the direct sound and the volume of the reflected sound, and the circuit calculates the volume ratio based on the sound space information and controls whether or not to select the reflected sound based on the volume ratio. This is an audio processing device described in Technology 2.

[0654] (Technology 4) An audio processing device according to Technology 3, wherein when the reflected sound is selected, the circuit applies binaural processing to the reflected sound and the direct sound to generate sounds that arrive at each of the listener's ears.

[0655] (Technology 5) An audio processing device described in Technology 3 or 4, wherein the circuit calculates the time difference between the end time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference and the volume ratio.

[0656] (Technology 6) An audio processing device as described in Technology 5, wherein the circuit selects the reflected sound when the volume ratio is greater than or equal to a threshold, and a first threshold used as the threshold when the time difference is a first value is greater than a second threshold used as the threshold when the time difference is a second value greater than the first value.

[0657] (Technology 7) An acoustic processing device described in Technology 3 or 4, in which the circuit calculates the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference and the volume ratio.

[0658] (Technology 8) An audio processing device as described in Technology 7, wherein the circuit selects the reflected sound when the volume ratio is greater than or equal to a threshold, and a first threshold used as the threshold when the time difference is a first value is greater than a second threshold used as the threshold when the time difference is a second value greater than the first value.

[0659] (Technology 9) The sound processing device according to Technology 8, wherein the circuit adjusts the threshold value based on the direction from which the direct sound arrives and the direction from which the reflected sound arrives.

[0660] (Technology 10) The sound processing device according to any one of technologies 2 to 9, wherein the circuit corrects the volume of the direct sound based on the volume of the reflected sound when the reflected sound is not selected.

[0661] (Technology 11) The sound processing device according to any one of Techniques 2 to 9, wherein the circuit synthesizes the reflected sound with the direct sound when the reflected sound is not selected.

[0662] (Technology 12) An audio processing device described in any of Technologies 3 to 9, wherein the volume ratio is the volume ratio between the volume of the direct sound at a first time and the volume of the reflected sound at a second time different from the first time.

[0663] (Technology 13) The sound processing device described in Technology 1 or 2, wherein the circuit sets a threshold based on characteristics related to the first sound and controls whether or not to select the second sound based on the threshold.

[0664] (Technology 14) An audio processing device described in any one of technologies 1, 2, and 13, wherein the characteristic related to the first sound is any one of the volume of the sound source, the visibility of the sound source, and the localization of the sound source, or a combination of any two or more of these.

[0665] (Technology 15) The sound processing device according to any one of technologies 1, 2, and 13, wherein the characteristic related to the first sound is a frequency characteristic of the first sound.

[0666] (Technology 16) A sound processing device according to any one of technologies 1, 2, and 13, wherein the characteristic related to the first sound is a characteristic indicating the intermittency of the amplitude of the first sound.

[0667] (Technology 17) An audio processing device described in any one of technologies 1, 2, 13, and 16, wherein the characteristic related to the first sound is a characteristic indicating the duration of a sound portion of the first sound or the duration of a silent portion of the first sound.

[0668] (Technology 18) An audio processing device described in any one of technologies 1, 2, 13, 16 and 17, wherein the characteristic related to the first sound is a characteristic indicating the duration of the sound portion of the first sound and the duration of the silent portion of the first sound in time series.

[0669] (Technology 19) A sound processing device according to any one of technologies 1, 2, 13, and 15, wherein the characteristic related to the first sound is a characteristic indicating a fluctuation in the frequency characteristics of the first sound.

[0670] (Technology 20) A sound processing device according to any one of technologies 1, 2, 13, 15, and 19, wherein the characteristic related to the first sound is a characteristic indicating the constancy of the frequency characteristics of the first sound.

[0671] (Technology 21) An audio processing device according to any one of technologies 1, 2, and 13 to 20, wherein the characteristics related to the first sound are acquired from a bitstream.

[0672] (Technology 22) An audio processing device described in any of Technologies 1, 2, and 13 to 21, wherein the circuit calculates characteristics related to the second sound and controls whether or not to select the second sound based on the characteristics related to the first sound and the characteristics related to the second sound.

[0673] (Technology 23) The circuit acquires a threshold value indicating a volume corresponding to the boundary between whether a sound can be heard or not, and controls whether to select the second sound based on characteristics related to the first sound, characteristics related to the second sound, and the threshold value.

[0674] (Technology 24) The sound processing device according to Technology 23, wherein the characteristic related to the second sound is the volume of the second sound.

[0675] (Technology 25) An audio processing device described in Technology 1 or 2, wherein the sound space information includes information on the listener's position in the sound space, the second sounds are each of a plurality of second sounds that occur in the sound space corresponding to the first sound, and the circuit selects one or more processing target sounds to which binaural processing is applied from among the plurality of second sounds by controlling whether or not to select each of the plurality of second sounds based on characteristics related to the first sound.

[0676] (Technology 26) The timing for acquiring the characteristics related to the first sound is at least one of when the sound space is created, when processing of the sound space begins, and when an information update thread occurs during processing of the sound space, in the sound processing device described in Technology 25.

[0677] (Technology 27) The sound processing device according to Technology 26, wherein the characteristics related to the first sound are acquired periodically after processing of the sound space begins.

[0678] (Technology 28) An audio processing device described in Technology 1 or 2, wherein the characteristic related to the first sound is the volume of the first sound, and the circuit calculates an evaluation value of the second sound based on the volume of the first sound, and controls whether or not to select the second sound based on the evaluation value.

[0679] (Technology 29) The sound processing device described in Technology 28, wherein the volume of the first sound has a transition.

[0680] (Technology 30) The sound processing device described in Technology 28 or 29, wherein the circuit calculates the evaluation value such that the second sound is more likely to be selected as the volume of the first sound increases.

[0681] (Technology 31) The sound space information is scene information including information on the sound source in the sound space and information on the position of the listener in the sound space, the second sound is each of a plurality of second sounds that occur in the sound space corresponding to the first sound, and the circuit acquires a signal of the first sound, calculates the plurality of second sounds based on the scene information and the signal of the first sound, acquires characteristics related to the first sound from the information on the sound source, and selects one or more second sounds to which binaural processing is not applied from among the plurality of second sounds by controlling whether or not to select each of the plurality of second sounds as a sound to which binaural processing is not applied based on the characteristics related to the first sound.

[0682] (Technology 32) The sound processing device described in Technology 31, wherein the scene information is updated based on input information, and the characteristics related to the first sound are acquired in response to the update of the scene information.

[0683] (Technology 33) An audio processing device described in Technology 31 or 32, wherein the scene information and characteristics related to the first sound are obtained from metadata included in a bitstream.

[0684] (Technology 34) An acoustic processing method including the steps of: acquiring sound space information relating to a sound space; acquiring characteristics relating to a first sound generated from a sound source in the sound space based on the sound space information; and controlling whether or not to select a second sound generated in the sound space corresponding to the first sound based on the characteristics relating to the first sound.

[0685] (Technology 35) A program for causing a computer to execute the acoustic processing method described in Technology 34.

[0686] (Technology 36) An audio processing device described in any of Technologies 3 to 9 and 12, comprising an addition means for applying additional processing to the second sound, an addition increase / decrease amount acquisition means for acquiring an increase / decrease amount of amplitude in the addition means, and a volume ratio adjustment means for adjusting the volume ratio according to the increase / decrease amount acquired by the addition increase / decrease amount acquisition means.

[0687] (Technology 37) An audio processing device described in any of Technologies 5 to 9, comprising an addition means for applying additional processing to the second sound, an addition delay amount acquisition means for acquiring the amount of increase or decrease in the delay in the addition means, and a time difference adjustment means for adjusting the time difference according to the amount of increase or decrease acquired by the addition delay amount acquisition means.

[0688] (Technology 38) The sound processing device described in Technology 36 or 37, wherein the additional processing is processing that is not applied to the first sound.

[0689] (Technology 39) The sound processing device according to Technology 36 or 37, wherein the adding means is a filter means for audibly emphasizing the diffusion of sound.

[0690] (Technology 40) An acoustic processing device comprising a circuit and a memory, wherein the circuit uses the memory to acquire sound space information relating to a sound space, and based on the sound space information, acquires a volume ratio between the volume of a direct sound generated from a sound source in the sound space and the volume of a reflected sound generated in the sound space corresponding to the direct sound, acquires information relating to additional processing, and controls whether or not to select the reflected sound based on the information relating to the additional processing and the volume ratio.

[0691] (Technology 41) The sound processing device described in Technology 40, wherein the circuit adjusts the volume ratio according to information regarding the additional processing and controls whether or not to select the reflected sound based on the adjusted volume ratio.

[0692] (Technology 42) An audio processing device described in Technology 40 or 41, in which the circuit calculates the time at which the sound emitted from the sound source arrives as the reflected sound, and controls whether or not to select the reflected sound based on the time, information regarding the additional processing, and the volume ratio.

[0693] (Technology 43) An audio processing device described in any of Technologies 40 to 42, wherein the circuit calculates the time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference, information regarding the additional processing, and the volume ratio.

[0694] (Technology 44) An audio processing device described in any of Technologies 40 to 42, wherein the circuit calculates the time difference between the end time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference, information regarding the additional processing, and the volume ratio.

[0695] (Technology 45) An audio processing device described in any of Technologies 40 to 44, wherein the additional processing involves increasing or decreasing the amplitude of the reflected sound, the information regarding the additional processing is information indicating the amount of increase or decrease in amplitude in the processing involving an increase or decrease in the amplitude of the reflected sound, and the circuit adjusts the volume ratio according to the amount of increase or decrease in amplitude, and controls whether or not to select the reflected sound based on the adjusted volume ratio.

[0696] (Technology 46) An audio processing device described in Technology 43 or 44, wherein the additional processing is processing that involves increasing or decreasing the delay of the reflected sound, the information regarding the additional processing is information indicating the amount of increase or decrease in the delay in processing that involves increasing or decreasing the delay of the reflected sound, and the circuit adjusts the time difference according to the amount of increase or decrease in the delay, and controls whether or not to select the reflected sound based on the adjusted time difference and the volume ratio.

[0697] (Technology 47) A sound processing device according to any one of Techniques 40 to 46, wherein the additional processing is processing that is not applied to the direct sound.

[0698] (Technology 48) An audio processing device according to any one of techniques 40 to 47, wherein the additional processing is a filter process that auditorily emphasizes the diffusion of sound.

[0699] (Technology 49) An audio processing device according to Technology 41 or 45, wherein the volume ratio is not adjusted if the information regarding the additional processing indicates that the additional processing is not to be performed.

[0700] (Technology 50) An audio processing device according to Technology 46, wherein the time difference is not adjusted if the information regarding the additional processing indicates that the additional processing is not to be performed.

[0701] (Technology 51) A sound processing device comprising a circuit and a memory, wherein the circuit uses the memory to acquire sound space information relating to a sound space, and determines whether a sound source in the sound space is a point sound source based on the sound space information, and if it is determined that the sound source is the point sound source, (i) based on the sound space information, acquires a volume ratio between the volume of a direct sound generated from the sound source and the volume of a reflected sound generated in the sound space corresponding to the direct sound, (ii) based on the sound space information, calculates a time difference between the arrival time of the direct sound and the arrival time of the reflected sound, and (iii) based on the volume ratio and the time difference, controls whether to select the reflected sound, and if it is determined that the sound source is not the point sound source, selects the reflected sound regardless of the volume ratio and the time difference.

[0702] (Technology 52) A threshold specification device comprising a circuit and a memory, wherein the circuit uses the memory to obtain a time difference between the arrival time of a preceding sound and the arrival time of a following sound that occurs in response to the preceding sound, and specifies a threshold for a volume ratio of the volume of the following sound to the volume of the preceding sound based on the time difference, the threshold corresponding to a boundary between whether or not the following sound is affected by a preceding sound effect exerted by the preceding sound, and outputs the threshold, wherein the threshold specification is performed by specifying, in an area having a first axis of the time difference and a second axis of the volume ratio, the volume ratio corresponding to the time difference on a boundary line defined as a line indicating the boundary, and the domain of the boundary line with respect to the first axis is a time range in which the precedence sound effect occurs.

[0703] (Technology 53) A threshold identification device described in Technology 52, wherein when the first axis is the horizontal axis and the second axis is the vertical axis, the boundary line is a curve that slopes downward to the right and is convex upward in the domain.

[0704] (Technique 54) A threshold specifying device according to Technique 52 or 53, wherein the boundary line is defined by a mathematical formula having the time difference as a variable, and the threshold is calculated based on the mathematical formula.

[0705] (Technology 55) An acoustic processing method including the steps of: acquiring sound space information relating to a sound space; acquiring, based on the sound space information, a volume ratio between the volume of a direct sound generated from a sound source in the sound space and the volume of a reflected sound generated in the sound space corresponding to the direct sound; acquiring information relating to additional processing; and controlling whether or not to select the reflected sound based on the information relating to the additional processing and the volume ratio.

[0706] (Technology 56) A program for causing a computer to execute the acoustic processing method described in Technology 55.

[0707] The present disclosure includes aspects that are applicable to, for example, an audio processing device, an encoding device, a decoding device, or a terminal or device that includes any of these devices.

[0708] 1000 Stereophonic sound reproduction system 1001 Audio signal processing device (audio processing device) 1002 Audio presentation device 1100, 1120, 1500 Encoding device 1101, 1113 Input data 1102 Encoder 1103 Encoded data 1104, 1114, 1404, 1503 Memory 1110, 1130 Decoding device 1111 Audio signal 1112, 1200, 1210 Decoder 1121 Transmitting unit 1122 Transmitted signal 1131 Receiving unit 1132 Received signal 1201, 1211 Spatial information management unit 1202 Audio data decoder 1203, 1213, 1300 Rendering unit 1301 Analysis unit 1302, 1314 Selection unit 1303 Synthesis unit 1304 Threshold adjustment unit 1311 Reverberation processing unit 1312 Early reflection processing unit 1313 Distance attenuation processing unit 1315 Generation unit 1316 Binaural processing unit 1401 Speaker 1402, 1501 Processor 1403, 1502 Communication IF 1405 Sensor

Claims

1. A sound processing device comprising a circuit and a memory, the circuit using the memory to acquire sound space information relating to a sound space, acquire a volume ratio between a volume of a direct sound generated from a sound source in the sound space and a volume of a reflected sound generated in the sound space corresponding to the direct sound based on the sound space information, acquire information relating to additional processing, and control whether or not to select the reflected sound based on the information relating to the additional processing and the volume ratio.

2. The sound processing device according to claim 1, wherein the circuit adjusts the volume ratio in accordance with the information on the additional processing, and controls whether or not to select the reflected sound based on the adjusted volume ratio.

3. The sound processing device according to claim 1, wherein the circuit calculates the time at which the sound emitted from the sound source arrives as the reflected sound, and controls whether or not to select the reflected sound based on the time, information related to the additional processing, and the volume ratio.

4. The sound processing device of claim 1, wherein the circuit calculates a time difference between the arrival time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference, information regarding the additional processing, and the volume ratio.

5. The sound processing device of claim 1, wherein the circuit calculates the time difference between the end time of the direct sound and the arrival time of the reflected sound based on the sound space information, and controls whether or not to select the reflected sound based on the time difference, information regarding the additional processing, and the volume ratio.

6. The sound processing device of claim 1, wherein the additional processing is processing involving an increase or decrease in the amplitude of the reflected sound, the information relating to the additional processing is information indicating an amount of increase or decrease in amplitude in the processing involving an increase or decrease in the amplitude of the reflected sound, and the circuit adjusts the volume ratio according to the amount of increase or decrease in amplitude, and controls whether or not to select the reflected sound based on the adjusted volume ratio.

7. The sound processing device described in claim 4 or 5, wherein the additional processing is processing involving an increase or decrease in the delay of the reflected sound, the information relating to the additional processing is information indicating an amount of increase or decrease in the delay in the processing involving an increase or decrease in the delay of the reflected sound, and the circuit adjusts the time difference according to the amount of increase or decrease in the delay, and controls whether or not to select the reflected sound based on the adjusted time difference and the volume ratio.

8. The sound processing device according to claim 1, wherein the additional processing is processing that is not applied to the direct sound.

9. The sound processing device according to claim 1, wherein the additional processing is a filter process that auditorily emphasizes the diffusion of sound.

10. The sound processing device according to claim 2 or 6, wherein the volume ratio is not adjusted if the information regarding the additional processing indicates that the additional processing is not to be executed.

11. The sound processing device according to claim 7, wherein the time difference is not adjusted if the information regarding the additional processing indicates that the additional processing is not to be executed.

12. An acoustic processing device comprising a circuit and a memory, wherein the circuit uses the memory to obtain sound space information relating to a sound space, and determines whether a sound source in the sound space is a point sound source based on the sound space information, and if it is determined that the sound source is the point sound source, (i) obtains, based on the sound space information, a volume ratio between a volume of a direct sound generated from the sound source and a volume of a reflected sound generated in the sound space corresponding to the direct sound, (ii) calculates, based on the sound space information, a time difference between the arrival time of the direct sound and the arrival time of the reflected sound, and (iii) controls whether or not to select the reflected sound based on the volume ratio and the time difference, and if it is determined that the sound source is not the point sound source, selects the reflected sound regardless of the volume ratio and the time difference.

13. A threshold identification device comprising a circuit and a memory, wherein the circuit uses the memory to obtain a time difference between an arrival time of a preceding sound and an arrival time of a following sound occurring in response to the preceding sound, and identifies a threshold of a volume ratio of the volume of the following sound to the volume of the preceding sound based on the time difference, the threshold corresponding to a boundary as to whether or not the following sound is affected by the preceding sound effect exerted by the preceding sound, and outputs the threshold, wherein in identifying the threshold, the volume ratio corresponding to the time difference is identified as the threshold on a boundary line defined as a line indicating the boundary in an area having a first axis of the time difference and a second axis of the volume ratio, and a domain of the boundary line with respect to the first axis is a time range in which the preceding sound effect occurs.

14. The threshold specifying device according to claim 13, wherein when the first axis is a horizontal axis and the second axis is a vertical axis, the boundary line is a right-sloping, upwardly convex curve in the domain.

15. The threshold determination device according to claim 13 or 14, wherein the boundary line is defined by a mathematical formula having the time difference as a variable, and the threshold is calculated based on the mathematical formula.

16. An acoustic processing method comprising the steps of: acquiring sound space information relating to a sound space; acquiring a volume ratio between a volume of a direct sound generated from a sound source in the sound space and a volume of a reflected sound generated in the sound space corresponding to the direct sound based on the sound space information; acquiring information relating to additional processing; and controlling whether or not to select the reflected sound based on the information relating to the additional processing and the volume ratio.

17. A program for causing a computer to execute the acoustic processing method according to claim 16.

Images