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

Abstract

This audio signal processing method is executed by an audio signal processing device, and includes: an acquisition step for acquiring an audio signal which is an audio signal and which includes attribute information for specifying an attribute of the audio signal; a determination step for, when the attribute specified by the attribute information included in the acquired audio signal is information indicating an indirect sound, performing a first determination process for determining whether or not the acquired audio signal satisfies a first condition and a second determination process for determining whether or not same satisfies a second condition different from the first condition; and a reproduction step for outputting an output signal based on the acquired audio signal, when the acquired audio signal satisfies the first condition and the second condition.

Description

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

[0001] The present disclosure relates to an audio signal processing method 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, audio signal 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 has become increasingly important.

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

[0004] Japanese Patent No. 6288100 JP 2019-22049 A International Publication No. 2021 / 180938

[0005] B. C. J. Moore, "Introduction to Auditory Psychology," Seishin Shobo, April 20, 1994, Chapter 6: Spatial Perception, p. 225

[0006] However, with the technology disclosed in Patent Document 1, it may be difficult to appropriately reduce the amount of calculation and the calculation load.

[0007] Therefore, an object of the present disclosure is to provide an audio signal processing method and the like that can appropriately reduce the amount of calculation and the calculation load.

[0008] An audio signal processing method according to one aspect of the present disclosure is an audio signal processing method executed by an audio signal processing device, and includes: an acquisition step of acquiring an audio signal including attribute information that identifies an attribute of the audio signal; a determination step of performing a first determination process of determining whether the acquired audio signal satisfies a first condition and a second determination process of determining whether the acquired audio signal satisfies a second condition different from the first condition, if the attribute identified by the attribute information included in the acquired audio signal is information indicating indirect sound; and a reproduction step of outputting an output signal based on the acquired audio signal, if the acquired audio signal satisfies the first condition and the second condition.

[0009] An audio signal processing method according to one aspect of the present disclosure is an audio signal processing method executed by an audio signal processing device, the method including: an acquisition step of acquiring an audio signal indicating indirect sound; a renderer pipeline step of performing one or more first processes, a first determination process, a second determination process, and one or more second processes different from the one or more first processes on the acquired audio signal; a reproduction step of outputting an output signal based on the acquired audio signal; a first gain accumulation step of calculating a first increase or decrease by accumulating increase or decrease amounts determined by each of the one or more first processes and used to amplify the amplitude of the acquired audio signal; and a second gain accumulation step of calculating a first increase or decrease amount by accumulating increase or decrease amounts determined by each of the one or more second processes and used to amplify the amplitude of the acquired audio signal. and a second gain accumulation step of accumulating the amounts of increase or decrease caused by the calculation to calculate a second amount of increase or decrease, wherein in the first determination process, if the calculated first amount of increase or decrease is equal to or greater than a first threshold, it is determined that the acquired audio signal satisfies a first condition, and if it is determined that the acquired audio signal does not satisfy the first condition, the second determination process is not performed, and if it is determined that the acquired audio signal satisfies the first condition, the second determination process is performed, and in the second determination process, if a value corresponding to the calculated second amount of increase or decrease is equal to or greater than a second threshold different from the first threshold, it is determined that the acquired audio signal satisfies the second condition, and if it is determined that the acquired audio signal satisfies the second condition, the reproduction step is performed.

[0010] Furthermore, a computer program according to one aspect of the present disclosure causes a computer to execute the above-described audio signal processing method.

[0011] In addition, an audio signal processing device according to one aspect of the present disclosure includes an acquisition unit that acquires an audio signal, the audio signal including attribute information that identifies an attribute of the audio signal; a judgment unit that, when the attribute identified by the attribute information included in the acquired audio signal is information indicating indirect sound, performs a first judgment process that determines whether the acquired audio signal satisfies a first condition and a second judgment process that determines whether the acquired audio signal satisfies a second condition different from the first condition; and a playback unit that, when the acquired audio signal satisfies the first condition and the second condition, outputs an output signal based on the acquired audio signal.

[0012] 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 of a system, an apparatus, a method, an integrated circuit, a computer program, and a recording medium.

[0013] According to the audio signal processing method and the like according to one aspect of the present disclosure, the amount of calculation and the calculation load can be appropriately reduced.

[0014] 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 Embodiment 1. FIG. 3A is a block diagram showing an example of the configuration of an encoding device according to Embodiment 1. FIG. 3B is a block diagram showing an example of the configuration of a decoding device according to Embodiment 1. FIG. 3C is a block diagram showing another example of the configuration of an encoding device according to Embodiment 1. FIG. 3D is a block diagram showing another example of the configuration of a decoding device according to Embodiment 1. FIG. 4A is a block diagram showing an example of the configuration of a decoder according to Embodiment 1. FIG. 4B is a block diagram showing another example of the configuration of a decoder according to Embodiment 1. FIG. 5 is a diagram showing an example of the physical configuration of an audio signal processing device according to Embodiment 1. FIG. 6 is a diagram showing an example of the physical configuration of an encoding device according to Embodiment 1. FIG. 7 is a block diagram showing an example of the configuration of a rendering unit according to Embodiment 1. FIG. 8 is a flowchart showing an example of the operation of the audio signal processing device according to Embodiment 1. FIG. 9 is a diagram showing a positional relationship between a listener and an obstacle object when they are relatively far apart. FIG. 10 is a diagram showing a positional relationship between a listener and an obstacle object when they are relatively close together. 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 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 a selection process. FIG. 15 is a diagram showing the relationship between the direction of a direct sound, the direction of a reflected sound, the time difference, and the threshold. FIG. 16 is a diagram showing the relationship between the angle difference, the time difference, and the threshold. FIG. 17 is a block diagram showing another example of the configuration of a rendering unit. FIG. 18 is a flowchart showing another example of the selection process. FIG. 19 is a flowchart showing yet another example of the selection process. FIG. 20 is a flowchart showing a first modified example of the operation of the audio signal processing device according to Embodiment 1. FIG. 21 is a flowchart showing a second modified example of the operation of the audio signal processing device according to Embodiment 1. FIG. 22 is a diagram showing an example of the arrangement of an avatar, a sound source object, and an obstacle object. FIG. 23 is a flowchart showing yet another example of the selection process.FIG. 24 is a block diagram showing an example of a configuration for a rendering unit to perform pipeline processing. FIG. 25 is a diagram showing transmission and diffraction of sound. FIG. 26 is a block diagram showing an example of a configuration of a rendering unit according to embodiment 2. FIG. 27 is a flowchart showing an example of an operation of an audio signal processing device according to embodiment 2. FIG. 28 is a graph showing threshold data indicating a first threshold according to embodiment 2. FIG. 29 is a block diagram showing an example of a configuration of a rendering unit according to embodiment 3. FIG. 30 is a block diagram showing an example of a configuration of a rendering unit according to embodiment 4. FIG. 31 is a diagram explaining energy of a head-related transfer function according to embodiment 4. FIG. 32 is a diagram explaining energy of a head-related transfer function according to embodiment 4. FIG. 33 is a diagram explaining energy of a head-related transfer function according to embodiment 4. FIG. 34 is a diagram explaining energy of a head-related transfer function according to embodiment 4. FIG. 35 is a flowchart showing an example of an operation of an audio signal processing device according to embodiment 4. FIG. 36 is a block diagram showing an example of a configuration of a rendering unit according to embodiment 5. FIG. 37 is a diagram showing a table for explaining influences of change of the first threshold and invalidation processing according to embodiment 5.

[0015] (Knowledge forming the basis of the present disclosure) Conventionally, audio signal processing techniques have been studied that provide immersive audio to listeners in a virtual or real space by adding acoustic effects that arise according to the environment of the space to sounds emitted by a virtual sound source.

[0016] Such an audio signal processing technology is disclosed in Patent Document 1. More specifically, Patent Document 1 discloses a technology for detecting the importance of an audio signal (voice signal) and not outputting an audio signal with a detected low importance. By not outputting an audio signal with a low importance in this way, the audio signal processing technology is expected to appropriately reduce the amount of calculation and the calculation load.

[0017] Incidentally, in a sound space (virtual space or real space), reflected sound can be important.

[0018] 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).

[0019] 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.

[0020] 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.

[0021] 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.

[0022] 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.

[0023] 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.

[0024] Furthermore, in some sound reproduction systems, degrees of freedom such as 6 DoF (6 Degrees of Freedom) are permitted for the position and orientation of the listener (i.e., the listening position where the listener is located). 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, reflected sounds are also not determined until playback. Therefore, it is difficult to determine the reflected sounds to be processed in advance.

[0025] Therefore, appropriately selecting and outputting (playing back) one or more reflected sounds that are to be processed or not to be processed from among the multiple reflected sounds that occur in the sound space during playback is useful for appropriately reducing the amount of calculation and the calculation load.

[0026] Note that controlling whether to select a sound corresponds to determining whether to select a sound, and more specifically, to determining whether to select a sound and output (play) it. 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.

[0027] However, in Patent Document 1, the importance of an audio signal, more specifically, the importance of a direct sound indicated by the audio signal is detected, but the importance of a reflected sound is not considered. Therefore, when an indirect sound such as a reflected sound occurs as shown in Figure 1, the amount of calculation and the calculation load increase, which means that it may be difficult to appropriately reduce the amount of calculation and the calculation load.

[0028] Therefore, there is a demand for an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load in the sound space.

[0029] Therefore, an audio signal processing method according to a first aspect of the present disclosure is an audio signal processing method executed by an audio signal processing device, and includes: an acquisition step of acquiring an audio signal including attribute information that identifies an attribute of the audio signal; a determination step of performing a first determination process of determining whether the acquired audio signal satisfies a first condition and a second determination process of determining whether the acquired audio signal satisfies a second condition different from the first condition, when the attribute identified by the attribute information included in the acquired audio signal is information indicating indirect sound; and a reproduction step of outputting an output signal based on the acquired audio signal when the acquired audio signal satisfies the first condition and the second condition.

[0030] As a result, the first determination process and the second determination process are performed on an audio signal whose attribute is indirect sound, and an output signal based on the acquired audio signal is output if the audio signal satisfies the first condition and the second condition. That is, it is appropriately determined whether or not an output signal based on an audio signal whose attribute is indirect sound is to be output. If an output signal is not to be output, the amount of calculation and the calculation load are reduced. That is, it is possible to realize an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load.

[0031] An audio signal processing method according to a second aspect of the present disclosure is the audio signal processing method according to the first aspect, wherein the indirect sound is a reflected sound.

[0032] This realizes an audio signal processing method that can use reflected sound as indirect sound.

[0033] In the audio signal processing method according to the third aspect of the present disclosure, in the audio signal processing method according to the first or second aspect, if the attribute identified by the attribute information contained in the acquired audio signal is information indicating a predetermined sound different from the indirect sound, the second judgment process is not performed in the judgment step.

[0034] As a result, the second determination process is not performed on the audio signal whose attribute is a predetermined sound, and the amount of calculation and the calculation load are further reduced. In other words, it is possible to realize an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load.

[0035] An audio signal processing method according to a fourth aspect of the present disclosure is the audio signal processing method according to the third aspect, wherein the predetermined sound is a direct sound.

[0036] This realizes an audio signal processing method that can use direct sound as the predetermined sound.

[0037] An audio signal processing method according to a fifth aspect of the present disclosure is an audio signal processing method according to any one of the first to fourth aspects, wherein, in the first determination process, if the amplitude value of the acquired audio signal is equal to or greater than a first threshold, it is determined that the acquired audio signal satisfies the first condition, and in the second determination process, if the volume ratio between the direct sound related to the indirect sound and the indirect sound when the direct sound and the indirect sound arrive at a listening position where a listener is located is equal to or greater than a second threshold determined according to the time difference between the arrival of the direct sound and the indirect sound, it is determined that the acquired audio signal satisfies the second condition.

[0038] As a result, in the first determination process, the audio signal is determined to satisfy the first condition if the amplitude value is equal to or greater than a first threshold, and in the second determination process, the audio signal is determined to satisfy the second condition if the volume ratio is equal to or greater than a second threshold. An output signal based on the acquired audio signal is output when such a second condition is satisfied. In other words, it is more appropriately determined whether or not an output signal based on the audio signal is to be output. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0039] An audio signal processing method according to a sixth aspect of the present disclosure is the audio signal processing method according to any one of the first to fifth aspects, wherein the reproduction step includes a first reproduction step of outputting a first output signal based on the audio signal whose attribute is a predetermined sound different from the indirect sound, and a second reproduction step of outputting a second output signal based on the audio signal whose attribute is the indirect sound, wherein the second reproduction step outputs the second output signal by performing a diffusion filter process on the acquired audio signal to diffuse the indirect sound and thereby improve the realism of the indirect sound, and the first reproduction step outputs the first output signal without performing the diffusion filter process on the acquired audio signal.

[0040] As a result, the audio signal whose attribute is indirect sound is subjected to diffusion filtering before being output, allowing the listener to hear indirect sound with higher sound quality. Also, the audio signal whose attribute is predetermined sound is not subjected to diffusion filtering, further reducing the amount of calculation and the calculation load. In other words, it is possible to realize an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load.

[0041] An audio signal processing method according to a seventh aspect of the present disclosure is an audio signal processing method according to any one of the first to sixth aspects, in which the second determination process is performed after the first determination process is performed.

[0042] This realizes an audio signal processing method that can perform the second determination process after the first determination process has been performed.

[0043] Performing the determination processes in this order produces the following special effects. The first determination process is primarily a process of determining whether the amplitude value of the target audio signal is greater than or less than a predetermined threshold, and therefore the computational load is extremely light. On the other hand, the second determination process requires calculation of the volume ratio and arrival time difference between the target indirect sound and the direct sound associated with the indirect sound, and therefore the computational load is significantly greater than that of the first determination process. Therefore, in the first determination process, which has a light computational load, multiple input signals are first sieved, and only the remaining signals are subjected to determination in the second determination process. This significantly reduces the computational load of the entire determination step, and therefore this order of processing is extremely important in the present disclosure, whose original purpose is to reduce the computational load of audio signal processing.

[0044] An audio signal processing method according to an eighth aspect of the present disclosure is an audio signal processing method executed by an audio signal processing device, the audio signal processing method including: an acquisition step of acquiring an audio signal indicating indirect sound; a renderer pipeline step of performing one or more first processes, a first determination process, a second determination process, and one or more second processes different from the one or more first processes on the acquired audio signal; a reproduction step of outputting an output signal based on the acquired audio signal; a first gain accumulation step of calculating a first increase or decrease by accumulating increase or decrease amounts determined by each of the one or more first processes and by which the amplitude of the acquired audio signal is amplified; and and a second gain accumulation step of accumulating the amounts of increase or decrease caused by the acquired audio signal to calculate a second amount of increase or decrease, wherein in the first determination process, if the calculated first amount of increase or decrease is equal to or greater than a first threshold, it is determined that the acquired audio signal satisfies a first condition, and if it is determined that the acquired audio signal does not satisfy the first condition, the second determination process is not performed, and if it is determined that the acquired audio signal satisfies the first condition, the second determination process is performed, and in the second determination process, if a value corresponding to the calculated second amount of increase or decrease is equal to or greater than a second threshold different from the first threshold, it is determined that the acquired audio signal satisfies the second condition, and if it is determined that the acquired audio signal satisfies the second condition, the reproduction step is performed.

[0045] As a result, the first determination process and the second determination process are performed on the audio signal indicating indirect sound, and an output signal based on the acquired audio signal is output if the audio signal satisfies the first condition and the second condition. That is, it is appropriately determined whether or not an output signal based on the audio signal indicating indirect sound is to be output. If an output signal is not to be output, the amount of calculation and the calculation load are reduced. In other words, it is possible to realize an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load.

[0046] An audio signal processing method according to a ninth aspect of the present disclosure is the audio signal processing method according to the eighth aspect, wherein the second increase / decrease amount calculated in the second gain accumulation step is an increase / decrease amount when a diffusion filter process is performed to improve the realism of the indirect sound by diffusing the indirect sound, and includes an increase / decrease amount that amplifies the amplitude of the acquired audio signal, the value being a ratio between the volume of a direct sound related to the indirect sound and the second increase / decrease amount calculated in the second gain accumulation step, and the second threshold is determined according to the time difference between the arrival of the direct sound and the indirect sound.

[0047] As a result, in the second determination process, if the ratio is equal to or greater than the second threshold, it is determined that the audio signal indicating indirect sound satisfies the second condition, and an output signal based on the acquired audio signal when such second condition is satisfied is output. In other words, it is more appropriately determined whether or not an output signal based on an audio signal indicating indirect sound is to be output. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0048] An audio signal processing method according to a tenth aspect of the present disclosure is the audio signal processing method according to the eighth aspect, wherein the second amount of increase or decrease calculated in the second gain accumulation step includes an amount of increase or decrease in amplitude due to a sound quality adjustment function set by a listener's selection, the value is a ratio between the volume of a direct sound related to the indirect sound and the second amount of increase or decrease calculated in the second gain accumulation step, and the second threshold is determined according to the time difference between the arrival of the direct sound and the indirect sound.

[0049] This allows the listener to select the sound quality that suits their taste, and thus to listen to the sound with the sound quality that they prefer.

[0050] An audio signal processing method according to an eleventh aspect of the present disclosure is the audio signal processing method according to any one of the eighth to tenth aspects, wherein the first threshold value is a value related to volume.

[0051] This makes it possible to realize an audio signal processing method that can use a value related to the volume as the first threshold value.

[0052] An audio signal processing method according to a twelfth aspect of the present disclosure is an audio signal processing method according to any one of the eighth to eleventh aspects, wherein in the second gain accumulation step, a parameter is acquired indicating the amount of increase or decrease determined by each of the one or more second processes, which amount of increase or decrease amplifies the amplitude of the acquired audio signal, and the second amount of increase or decrease is calculated according to the acquired multiple parameters.

[0053] This allows the second increase or decrease amount to be calculated according to the parameter, thereby more appropriately determining whether or not to output an output signal based on an audio signal whose attribute is indirect sound. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0054] An audio signal processing method according to a thirteenth aspect of the present disclosure is an audio signal processing method according to any one of the eighth to twelfth aspects, which includes a change step of setting the first threshold and an invalidation step of performing an invalidation process of invalidating the second judgment process, and in the first judgment process, if the calculated first increase / decrease amount is equal to or greater than the set first threshold, it is judged that the acquired audio signal satisfies the first condition, and if the invalidation process is performed, the reproduction step is performed when it is judged that the acquired audio signal satisfies the first condition.

[0055] As a result, the second determination process is not performed due to the invalidation process, and the amount of calculation and the calculation load required to perform the second determination process can be reduced. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0056] An audio signal processing method according to a fourteenth aspect of the present disclosure is the audio signal processing method according to the thirteenth aspect, wherein the storage area for storing the value indicating the first threshold set in the change step and the storage area for storing the signal instructing to perform the invalidation processing are adjacent areas.

[0057] This not only makes it easier for the administrator or listener of the virtual space to visually grasp the set value, but also has the special effect of allowing both values ​​to be set simultaneously (in a single memory access process) because the memory areas are adjacent to each other. In other words, the value indicating the first threshold and the signal instructing the execution of the invalidation process are linked and arranged in the upper and lower bit fields of an area accessible by a single address, and a series of data arranged in this way is written in a single memory access, allowing the two data to be set simultaneously.

[0058] An audio signal processing method according to a fifteenth aspect of the present disclosure is the audio signal processing method according to the thirteenth aspect, wherein the first threshold value set when the invalidation processing is performed is greater than the first threshold value set when the invalidation processing is not performed.

[0059] This makes it possible to realize an audio signal processing method that can set the magnitude of the first threshold depending on whether or not the invalidation process is performed.

[0060] A computer program according to a sixteenth aspect of the present disclosure is a computer program for causing a computer to execute the audio signal processing method according to any one of the first to fifteenth aspects.

[0061] This allows the computer to execute the above-described audio signal processing method in accordance with the computer program.

[0062] An audio signal processing device according to a seventeenth aspect of the present disclosure includes an acquisition unit that acquires an audio signal, the audio signal including attribute information that identifies an attribute of the audio signal; a judgment unit that, when the attribute identified by the attribute information included in the acquired audio signal is information indicating indirect sound, performs a first judgment process that determines whether the acquired audio signal satisfies a first condition and a second judgment process that determines whether the acquired audio signal satisfies a second condition different from the first condition; and a playback unit that, when the acquired audio signal satisfies the first condition and the second condition, outputs an output signal based on the acquired audio signal.

[0063] As a result, the first determination process and the second determination process are performed on an audio signal whose attribute is indirect sound, and an output signal based on the acquired audio signal is output if the audio signal satisfies the first condition and the second condition. That is, it is appropriately determined whether or not an output signal based on an audio signal whose attribute is indirect sound is to be output. If an output signal is not to be output, the amount of calculation and the calculation load are reduced. That is, it is possible to realize an audio signal processing device that can appropriately reduce the amount of calculation and the calculation load.

[0064] (Embodiment 1) (Example of a stereophonic sound reproduction system) Fig. 2 is a diagram showing an example of a stereophonic sound reproduction system 1000. Specifically, Fig. 2 shows the 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.

[0065] 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.

[0066] 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.

[0067] 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.

[0068] 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.

[0069] 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.

[0070] 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, but the stereophonic sound reproduction system 1000 to which the audio processing method (audio signal 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.

[0071] 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.

[0072] 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 audio processing. Therefore, the audio signal processing device 1001 may be referred to as a decoding device.

[0073] (Example of Encoding Device) Fig. 3A is a block diagram showing an example configuration of encoding device 1100. Specifically, Fig. 3A shows the configuration of encoding device 1100, which is an example of an encoding device of the present disclosure.

[0074] 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 described later.

[0075] 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.

[0076] 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.

[0077] 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.

[0078] 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.

[0079] 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.

[0080] (Example of Decoding Device) Fig. 3B is a block diagram showing an example configuration of the decoding device 1110. Specifically, Fig. 3B shows the configuration of the decoding device 1110, which is an example of a decoding device of the present disclosure.

[0081] 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.

[0082] 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.

[0083] 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.

[0084] Decoder 1112 decodes input data 1113 to produce an audio signal 1111 representing the audio to be presented to the listener.

[0085] (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.

[0086] 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.

[0087] 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.

[0088] (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.

[0089] 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.

[0090] 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.

[0091] 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.

[0092] (Example of Decoder) Fig. 4A is a block diagram showing an example of the configuration of the decoder 1200. Specifically, Fig. 4A shows the configuration of the decoder 1200, which is an example of the decoder 1112 in Fig. 3B or 3D.

[0093] 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.

[0094] 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.

[0095] 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.

[0096] 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.

[0097] 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.

[0098] 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.

[0099] 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.

[0100] 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.

[0101] That is, the spatial information management unit 1201 may communicate with an external system or server to acquire spatial information and listener positions (i.e., listening 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.

[0102] 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.

[0103] The audio data decoder 1202 decodes the encoded audio data included in the input data 1113 to obtain an audio signal.

[0104] 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.

[0105] 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).

[0106] 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.

[0107] 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).

[0108] 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.

[0109] 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 .

[0110] 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.

[0111] 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.

[0112] 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.

[0113] 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.

[0114] 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.

[0115] The role of the information update thread that executes the spatial information update process is, for example, to update the position or orientation of the listener's avatar placed in the virtual space based on the position or orientation of the VR goggles worn by the listener, and to update the position of objects moving in the virtual space. These tasks are handled within a processing thread that runs relatively infrequently, on the order of several tens of Hz. Processing that reflects the properties of direct sound may be performed in such an infrequently occurring processing thread. This is because the properties of direct sound change less frequently than the frequency with which audio processing frames for audio output occur. Doing so can actually reduce the computational load of the process relatively, and can also avoid the risk of pulsive noise occurring when information is updated at an unnecessarily fast frequency.

[0116] 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.

[0117] 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.

[0118] 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.

[0119] The rendering unit 1213 is the same as the rendering unit 1203 in FIG. 4A, and therefore a description thereof will be omitted.

[0120] 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.

[0121] (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.

[0122] 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 .

[0123] 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.

[0124] 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 and 1211, and threshold data, which will be described later.

[0125] 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.

[0126] 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.

[0127] 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).

[0128] 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.

[0129] 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.

[0130] 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.

[0131] 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.

[0132] 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.

[0133] 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.

[0134] 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.

[0135] 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.

[0136] 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.

[0137] Here, an example has been given in which the audio signal processing device 1001 is provided with a speaker 1401 and an audio signal after acoustic processing is presented via the speaker 1401, but the means for presenting the audio signal is not limited to the above configuration.

[0138] 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.

[0139] 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.

[0140] (Physical Configuration of Encoding Apparatus) Fig. 6 is a diagram showing an example of the physical configuration of encoding apparatus 1500. 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.

[0141] The encoding device 1500 in FIG. 6 includes a processor 1501 , a memory 1503 , and a communication IF 1502 .

[0142] 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.

[0143] 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.

[0144] 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.

[0145] 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).

[0146] The communication module is composed of, for example, a signal processing circuit and an antenna corresponding to the communication method. In the above example, Bluetooth (registered trademark) or WIGIG (registered trademark) was used as an example of the communication method, but the communication method may also be compatible with communication methods such as LTE (Long Term Evolution), NR (New Radio), or Wi-Fi (registered trademark). Furthermore, the communication IF may be a wired communication method such as Ethernet (registered trademark), USB (Universal Serial Bus), or HDMI (registered trademark) (High-Definition Multimedia Interface) instead of the wireless communication method described above.

[0147] [Configuration of Rendering Unit] Fig. 7 is a block diagram showing an example configuration of the rendering unit 1300. Specifically, Fig. 7 shows an example detailed configuration of the rendering unit 1300 corresponding to the rendering units 1203 and 1213 in Figs. 4A and 4B.

[0148] The rendering unit 1300 is composed of an analysis unit 1301, a determination unit 1302, and a reproduction unit 1303, and applies acoustic processing to sound data contained in an input signal and outputs the result.

[0149] 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.

[0150] 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.

[0151] 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.

[0152] 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.

[0153] 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.

[0154] The shape information may include information about the surface material.

[0155] 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.

[0156] 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.

[0157] 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.

[0158] 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.

[0159] 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 playback 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.

[0160] 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.

[0161] 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).

[0162] 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.

[0163] 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.

[0164] 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.

[0165] 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.

[0166] 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.

[0167] 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.

[0168] 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.

[0169] 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.

[0170] 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 calculation process for the volume of direct sound or reflected sound to be perceived by the listener, or may be used in a selection process (also called a determination process) for whether or not to allow the listener 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.

[0171] It should be noted that the reflected sound according to this embodiment is an example of an indirect sound. The indirect sound may be a reflected sound, a diffracted sound, or the like. The direct sound according to this embodiment is an example of a predetermined sound different from the indirect sound. The predetermined sound may be a direct sound or a High Order Ambisonics sound (HOA). It is also possible to create a representative sound of the multiple sounds represented by the multiple audio signals by bundling them together (by mixing, etc.), and use this representative sound as the predetermined sound. In this case, this representative sound may be called a representative sound.

[0172] In this embodiment, we will explain using reflected sound, which is an example of indirect sound, and direct sound, which is an example of specified sound, but the same processing is performed even if indirect sound is used instead of reflected sound, and specified sound is used instead of direct sound.

[0173] 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.

[0174] 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).

[0175] The position information and orientation information of the listener may change over time, and if so, is transmitted to the rendering unit 1300 .

[0176] 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.

[0177] 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.

[0178] The analysis unit 1301 analyzes the audio signal contained in the input signal and the spatial information received from the spatial information management units 1201 and 1211, and calculates the information necessary for generating direct sound and reflected sound in the playback unit 1303, as well as the information necessary for determining (selecting) whether or not to generate reflected sound.

[0179] The information required to generate direct sound and reflected sound is, for example, values ​​relating to the path taken by each of the direct sound and the reflected sound to reach the listening position, the time it takes for the sound to reach the listening position, the volume at the time of arrival, etc. The values ​​relating to the path taken by each of the direct sound and the reflected sound to reach the listening position, the time it takes for the sound to reach the listening position, the volume at the time of arrival, etc. are, for example, values ​​indicating the path taken by each of the direct sound and the reflected sound to reach the listening position, the time it takes for the sound to reach the listening position, and the volume at the time of arrival, respectively.

[0180] 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 relating to the time difference between the direct sound and the reflected sound, and a value relating to the volume ratio between the direct sound and the reflected sound at the listening position. The value relating to the time difference between the direct sound and the reflected sound and the value relating to the volume ratio between the direct sound and the reflected sound at the listening position are, for example, a value indicating the time difference between the direct sound and the reflected sound and a value indicating the volume ratio between the direct sound and the reflected sound at the listening position, respectively.

[0181] 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.

[0182] 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.

[0183] 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.

[0184] 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.

[0185] The time difference between the arrival of direct sound and reflected sound is, for example, the time difference between the arrival time of direct sound (arrival time) and the arrival time of reflected sound (arrival time). For simplicity, the time difference between the arrival of direct sound and reflected sound may be referred to as the time difference between direct sound and reflected sound. The time difference between direct sound and reflected sound may be the time difference between the times when the direct sound and reflected sound arrive at the listening position, the difference in the time it takes for the direct sound and 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. The calculation method for these values ​​will be described later.

[0186] The determination unit 1302 performs at least one of a first determination process using a first threshold value and a second determination process using a second threshold value. In this embodiment, the determination unit 1302 performs the second determination process on reflected sound (i.e., an audio signal indicating reflected sound). The second determination process will be described in more detail below.

[0187] The determination unit 1302 determines whether or not the reproduction unit 1303 will generate a reflected sound, using the information calculated by the analysis unit 1301 and threshold data indicating the second threshold. In other words, the determination unit 1302 determines whether or not to select a reflected sound as a reflected sound to be generated. In other words, the determination unit 1302 selects which of the multiple reflected sounds the reproduction unit 1303 will generate. Note that hereinafter, the determination by the determination unit 1302 that the reproduction unit 1303 will generate a reflected sound may also be described as the determination unit 1302 selecting a reflected sound or the determination unit 1302 selecting to generate a reflected sound.

[0188] The threshold data is represented, for example, as a graph having the value of the time difference between direct sound and reflected sound on the horizontal axis and the volume ratio between direct sound and reflected sound on the vertical axis, as a boundary (threshold) between whether the reflected sound is perceived or not. The threshold data may be expressed as an approximation formula having the value of the time difference between direct sound and reflected sound as a variable, or may be expressed as an array having the value of the time difference between direct sound and reflected sound as an index and corresponding thresholds. Note that in the first embodiment, the second threshold may be simply referred to as a threshold.

[0189] The determination 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. Note that the volume at the time of arrival means the volume of the sound when it reaches the listening position.

[0190] 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.

[0191] 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.

[0192] The reproduction unit 1303 synthesizes the audio signal of the direct sound with the audio signal of the reflected sound that the determination unit 1302 has selected to generate.

[0193] Specifically, the reproduction 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 reproduction 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 determination unit 1302. The reproduction unit 1303 then synthesizes and outputs the generated direct sound and reflected sound.

[0194] [Example of 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.

[0195] 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 determination unit 1302 as the reflected sound that will ultimately be generated by the reproduction 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.

[0196] 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.

[0197] 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 A1 (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 A1 (xa, ya, za) by the speed of sound (approximately 340 m / s).

[0198] 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.

[0199] The volume N at the sound source position may be the reference volume described above.

[0200] 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 A1 (xa, ya, za).

[0201] 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.

[0202] Fig. 9 is a diagram showing a positional relationship between a listener and an obstacle object where the listener is relatively far away. Fig. 10 is a diagram showing a positional relationship between a listener and an obstacle object where the listener 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 reflected sound arrival time can be determined in the same way as the method for calculating the direct sound arrival time.

[0203] 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 A1 (xa, ya, za) by the speed of sound (approximately 340 m / sec). The volume attenuates in inverse proportion 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.

[0204] 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.

[0205] 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.

[0206] The volume ratio (L), which is the ratio of the volume (ld) when direct sound arrives to the volume (lr) when direct sound arrives, is, for example, the value obtained by dividing the volume (lr) when reflected sound arrives by the volume (ld) when direct sound arrives, and is calculated as follows: L = (N * G * U / Y) / (N * U / X) = G * X / Y. Because the value to be calculated is the volume ratio, the values ​​of N and U may be any predetermined values.

[0207] 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 arrive at the listening position. For example, the time difference (T) between the direct sound and the reflected sound to arrive at the listening position can be calculated as T = tr - td.

[0208] 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.

[0209] Next, in the reflected sound selection process (S102 in FIG. 8), the determination unit 1302 selects whether or not the reproduction unit 1303 will generate the reflected sound calculated by the analysis unit 1301. In other words, the determination unit 1302 determines whether or not to select the reflected sound as a reflected sound to be generated. When there are multiple reflected sounds, the determination unit 1302 selects whether or not to generate each reflected sound. As a result of selecting whether or not to generate each reflected sound, the determination unit 1302 may select one or more reflected sounds to be generated from among the multiple reflected sounds, or may not select any reflected sounds to be generated.

[0210] The determination unit 1302 may select reflected sounds to which other processing is to be applied, not limited to generation processing. For example, the determination unit 1302 may select reflected sounds to which binaural processing is to be applied. Furthermore, the determination unit 1302 basically selects only one or more reflected sounds to be processed. However, the determination 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.

[0211] 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.

[0212] Specifically, the selection of whether to generate reflected sound is made by comparing 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 of the direct sound (Id) and the volume of the reflected sound (lr).

[0213] 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.

[0214] 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.

[0215] 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 determination 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.

[0216] 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, setting method, etc. of the threshold data will be described in detail later.

[0217] Next, in the process of generating direct sound and reflected sound (S103 in FIG. 8), the reproduction unit 1303 generates and synthesizes an audio signal of the direct sound and an audio signal of the reflected sound selected by the determination unit 1302 as the reflected sound to be generated.

[0218] The audio signal of the direct sound is generated by applying the direct sound arrival time (td) and the volume at direct sound arrival (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 direct sound arrival time (td) and multiplied by the volume at direct sound arrival (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, such as that disclosed in Patent Document 2, may be applied.

[0219] The audio signal of the reflected sound is generated by applying the reflected sound arrival time (tr) and the volume at the time of arrival of the reflected sound (Ir) calculated by the analysis unit 1301 to the sound data of the sound source object, just like the direct sound.

[0220] However, unlike the volume of the direct sound when it arrives, the volume of the reflected sound when it is generated (lr) is a value to which the attenuation rate G of the volume of the reflection is applied. G may be an attenuation rate that is applied to all frequency bands at once. Alternatively, a reflectance 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 the reflected sound when it arrives (lr) may be performed as a frequency equalizer process that multiplies each band by an attenuation rate.

[0221] 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 arrival volumes 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.

[0222] 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.

[0223] 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.

[0224] (Details of Selection Process) Details of the selection process of whether or not to generate reflected sound will be described.

[0225] 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.

[0226] 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.

[0227] 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.

[0228] 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.

[0229] 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.

[0230] 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 direct sound and the reflected 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.

[0231] 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).

[0232] 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.

[0233] The determination 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.

[0234] 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.

[0235] 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.

[0236] 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.

[0237] 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.

[0238] 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.

[0239] 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.

[0240] The transition information may be expressed as data on the duration for which the frequency characteristics of the sound signal can be considered 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 stationary and the frequency characteristics during those periods.

[0241] 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.

[0242] 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.

[0243] 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.

[0244] 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.

[0245] 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.

[0246] 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.

[0247] 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.

[0248] 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.

[0249] 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.

[0250] 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.

[0251] 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.

[0252] 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.

[0253] 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.

[0254] 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.

[0255] Note that setting the threshold value may include adjusting the threshold value, changing the threshold value, and the like.

[0256] [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.

[0257] 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.

[0258] 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.

[0259] Fig. 13 is a diagram showing an example of a method for setting a threshold. 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.

[0260] 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.

[0261] 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.

[0262] 14 is a flowchart showing an example of the selection process. First, the determination unit 1302 specifies the reflected sound detected by the analysis unit 1301 (S201). Then, the determination 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).

[0263] 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.

[0264] Specifically, the determination 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 determination 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.

[0265] 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 determination unit 1302 detects the volume ratio by calculating the ratio between these volumes.

[0266] The determination unit 1302 also uses the threshold data to identify a threshold value corresponding to the time difference (T) (S204), and determines whether the detected volume ratio (L) is equal to or greater than the threshold value (S205).

[0267] If the volume ratio (L) is equal to or greater than the threshold (Yes in S205), the determination 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 determination unit 1302 does not select the reflected sound as the reflected sound to be generated (S207). That is, in this case, the determination unit 1302 determines the reflected sound as a reflected sound not to be generated.

[0268] Thereafter, the determination unit 1302 determines whether or not there is an unspecified reflected sound (S208). If there is an unspecified reflected sound (Yes in S208), the determination unit 1302 repeats the above-described processing (S201 to S207). If there is no unspecified reflected sound (No in S208), the determination unit 1302 ends the processing.

[0269] 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.

[0270] [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 stored threshold data may be in any format and of any type. When multiple formats and 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 sounds. A method for determining which threshold data to use in the selection process will be described later.

[0271] Furthermore, threshold data in a plurality of formats and of a plurality of types may be stored in combination. The combined threshold data may be read from the spatial information management units 1201 and 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 and 1211.

[0272] 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.

[0273] 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.

[0274] For example, the threshold data may be expressed by the following formula, where the time difference (T) is timeDiff and the threshold is gainThresh.

[0275]

[0276] The threshold is defined only within the time range in which the precedence effect is expected to occur. For time differences outside this time range (values ​​of 1 ms or less or 40 ms or more in the above formula), the determination using gainThresh may not be performed, and the determination may be performed only based on a threshold indicating the minimum volume reproduced in the virtual space, as described below.

[0277] 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 is believed to occur. The above formula is an example of an approximation formula generated based on the experiments.

[0278] 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.

[0279] 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.

[0280] 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.

[0281] 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.

[0282] 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.

[0283] 15 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. 15, 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.

[0284] 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 time when the direct sound arrives at the listening position and the time when the reflected sound arrives. The volume ratio (L) corresponds to the volume ratio between the volume when the direct sound arrives and the volume when the reflected sound arrives.

[0285] Of course, the thresholds shown in Fig. 15 are merely an example, and the thresholds are not limited to the example of Fig. 15. Also, Fig. 15 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.

[0286] In the above description, the thresholds are stored in an array having the direction of the direct sound (θ) (more specifically, the angle (θ) of the direction from which the direct sound arrives) and the direction of the reflected sound (γ) (more specifically, the angle (γ) of the direction from which the reflected sound arrives) as independent variables or indexes. However, 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 do not have to be used as independent variables.

[0287] 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.

[0288] Fig. 16 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. 16. Of course, the threshold shown in Fig. 16 is just an example, and the threshold is not limited to the example of Fig. 16.

[0289] 16, 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.

[0290] 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.

[0291] A method for selecting reflected sounds using a threshold value according to the direction of arrival will be described later.

[0292] 12A , 12B, and 12C , multiple formats and multiple types of thresholds may be stored in the spatial information management units 1201 and 1211. Then, it may be determined which format and which type of threshold to use in the reflected sound selection process from the multiple formats and multiple types of thresholds. 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.

[0293] 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.

[0294] [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.

[0295] Fig. 17 is a block diagram showing another example configuration of the rendering unit 1300 shown in Fig. 7. The rendering unit 1300 in Fig. 17 differs from the rendering unit 1300 in Fig. 7 in that it includes a threshold adjustment unit 1304. The description of the components other than the threshold adjustment unit 1304 is omitted because they are the same as those described in Fig. 7.

[0296] The threshold adjustment unit 1304 selects a threshold to be used by the determination 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.

[0297] 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.

[0298] 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.

[0299] 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 determination 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.

[0300] Fig. 18 is a flowchart showing another example of the selection process. Fig. 19 is a flowchart showing yet another example of the selection process. In Fig. 18 and Fig. 19, a threshold is set according to the properties of the direct sound. Specifically, in Fig. 18, 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. 19, 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.

[0301] The operation of each example will be described below, with the explanation of the processes common to the example in FIG.

[0302] First, an example of processing shown in Fig. 18 will be described. Here, threshold data for each property of 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.

[0303] 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).

[0304] As shown in FIG. 19, the threshold value adjusting unit 1304 may adjust the threshold value determined by the determining unit 1302 based on the properties of the direct sound (S221).

[0305] 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.

[0306] 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.

[0307] In the examples of Figures 18 and 19, 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 18, or the threshold value may be adjusted according to the properties of the audio signal, as in Figure 19. Furthermore, the parameters of the threshold data may be adjusted according to the properties of the audio signal.

[0308] The operation performed by the threshold adjustment unit 1304 may be performed by the analysis unit 1301 or the determination unit 1302. For example, the analysis unit 1301 may acquire the properties of the audio signal. Alternatively, the determination unit 1302 may set a threshold according to the properties of the audio signal.

[0309] Next, the relationship between the characteristics of the audio signal and the threshold will be described.

[0310] 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.

[0311] 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.

[0312] 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.

[0313] 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 judgment unit 1302.

[0314] 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.

[0315] 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.

[0316] 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.

[0317] 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.

[0318] 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.

[0319] 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).

[0320] 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.

[0321] 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.

[0322] 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).

[0323] 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.

[0324] 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.

[0325] 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.

[0326] 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.

[0327] 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.

[0328] 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.

[0329] 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.

[0330] 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.

[0331] 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.

[0332] 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.

[0333] 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.

[0334] 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.

[0335] 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.

[0336] 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.

[0337] 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.

[0338] 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.

[0339] 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.

[0340]

[0341] 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.

[0342] 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.

[0343] Furthermore, information I(n) indicating the properties of the direct sound may be calculated according to the following formula:

[0344]

[0345] 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.

[0346] 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.

[0347] The above formulas 1 and 2 can be considered as filters in which E(n) is the input signal and I(n) is the output signal. In this case, formula 1 is a moving average (MA) model filter, and formula 2 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.

[0348] 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.

[0349] 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.

[0350] 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 1.

[0351] If the energy of the low frequency component of the n-th analysis frame is expressed as EL(n), the information I(n) indicating the properties of the direct sound can be calculated according to the following equation.

[0352]

[0353] 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.

[0354] 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.

[0355] Similarly to Equation 2, information I(n) indicating the properties of the direct sound may be calculated according to the following equation:

[0356]

[0357] 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.

[0358] 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.

[0359] The above equations 3 and 4 can be considered as filters in which E(n) is the input signal and I(n) is the output signal. In this case, equation 3 is a moving average (MA) model filter, and equation 4 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.

[0360] 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.

[0361] 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.

[0362] The relationship between the analysis frame length and the interval at which the information update thread occurs may be as follows:

[0363] 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 1) and (Equation 3) 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 2) and (Equation 4) for the AR filter may be values ​​such that the time constant of the filter is approximately TU (msec).

[0364] The reason for the above setting is that the filter is expected to converge within the interval period of information update.

[0365] 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.

[0366] 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.

[0367] 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.

[0368] 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.

[0369] Here, in the information update thread, processing for updating the spatial information managed by the spatial information management units 1201 and 1211 is carried out.

[0370] 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.

[0371] 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.

[0372] [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. As a result, the volume of more reflected sounds becomes lower than the threshold, making it possible to reduce the amount of reflected sounds that are subjected to binaural processing, and thereby reducing the amount of computation.

[0373] 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.

[0374] [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.

[0375] 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.

[0376] [First Modification of Operation of Rendering Unit] Fig. 20 is a flowchart showing a first modification of the operation of the audio signal processing device 1001. Fig. 20 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.

[0377] For example, the analysis unit 1301 acquires data (input signal) (S301). Next, the analysis unit 1301 analyzes the data (S302). Next, the determination unit 1302 determines whether or not to select reflected sound based on the analysis result (S303). Next, the playback unit 1303 performs volume compensation processing based on the reflected sound that is not selected (S304). Next, the playback unit 1303 performs acoustic processing on the direct sound and reflected sound (S305). Then, the playback unit 1303 outputs the direct sound and reflected sound as audio (S306).

[0378] 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.

[0379] 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.

[0380] First, a method for compensating for the sense of volume by increasing the volume of the direct sound will be described. The reproduction 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.

[0381] When increasing the volume, the playback 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.

[0382] Next, a method for compensating for the perceived loudness by synthesizing reflected sounds with direct sounds will be described. In this method, the playback unit 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.

[0383] 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.

[0384] 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.

[0385] 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.

[0386] [Second Modification of Operation of Rendering Unit] Fig. 21 is a flowchart showing a second modification of the operation of the audio signal processing device 1001. Fig. 21 shows the processing executed mainly by the rendering unit 1300 of the audio signal processing device 1001. In this modification, left-right volume difference adjustment processing is added to the operation of the rendering unit 1300.

[0387] 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 determination unit 1302 adjusts the difference in volume between the sounds perceived by the left and right ears (S403). The determination unit 1302 also adjusts the difference in arrival time (delay) between the sounds perceived by the left and right ears (S404). The determination unit 1302 determines whether to select a reflected sound based on the adjusted sound information (S405).

[0388] Of the above processes (S401 to S405), the processes other than the left-right volume difference adjustment process (S403) and the delay adjustment process (S404) are common to the other examples described above, and therefore descriptions thereof will be omitted.

[0389] Fig. 22 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 incoming direction of the direct sound (θ) and the incoming direction of the reflected sound (γ) (direction of the reflected sound (γ)) is different, as shown in Fig. 22, the volume difference occurring between the two ears is corrected.

[0390] 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 determination unit 1302 performs left-right volume difference adjustment processing (S403) to adjust the volume of the direct sound according to the position of the ear that primarily perceives the reflected sound. For example, the determination 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).

[0391] The determination 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.

[0392] Furthermore, in addition to the left-right volume difference adjustment process (S403), the determination unit 1302 may also perform a delay adjustment process (S404) in which the direct sound arrival time is delayed in accordance with the position of the ear that perceives the reflected sound. Specifically, the determination unit 1302 may delay the direct sound arrival time 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 direct sound arrival time.

[0393] [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.

[0394] Fig. 23 is a flowchart showing yet another example of the selection process. A description of the process common to the example of Fig. 14 will be omitted. In the example of Fig. 23, the determination unit 1302 selects a reflected sound using a threshold value according to the arrival direction.

[0395] Specifically, the determination unit 1302 calculates the direct sound arrival direction (θ) and the reflected sound arrival direction (γ) (the direction of the reflected sound (γ)) determined using the avatar orientation as a reference, from the direct sound arrival path (pd) and the reflected sound arrival path (pr) calculated by the analysis unit 1301, and the avatar orientation information D1. That is, the determination 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 D1 may be included in the input signal.

[0396] The determination 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 a threshold value to be used in the selection process from a three-dimensional array such as that shown in Figure 15 (S232).

[0397] 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. 22 will be described.

[0398] From the input signal, position information of the avatar, sound source object, and obstacle object, as well as avatar orientation information D1, are acquired. Using this position information and orientation information D1, 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 22, the direction of the direct sound (θ) is approximately 20 degrees, and the direction of the sound image of the reflected sound (γ) is approximately 265 degrees (-95 degrees).

[0399] 15, 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.

[0400] 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).

[0401] 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.

[0402] 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. 16. 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.

[0403] 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).

[0404] Alternatively, threshold data having the values ​​of (θ), (γ) and (T) as a three-dimensional index array as shown in FIG. 15 may be set.

[0405] [Fourth Modification of Operation of Rendering Unit] The processes performed by the analysis unit 1301, determination unit 1302, and reproduction unit 1303 described above may be performed as pipeline processes as described in, for example, Patent Document 3.

[0406] FIG. 24 is a block diagram showing an example of the configuration for the rendering unit 1300 to perform pipeline processing.

[0407] The rendering unit 1300 in Fig. 24 includes a reverberation processing unit 1311, an early reflection processing unit 1312, a distance attenuation processing unit 1313, a determination 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.

[0408] 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.

[0409] 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.

[0410] 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. 24 .

[0411] 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.

[0412] 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.

[0413] 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.

[0414] 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.

[0415] 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.

[0416] 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).

[0417] 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.

[0418] 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 determination unit 1314 selects the reflected sound.

[0419] 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.

[0420] In the selection process, the determination unit 1314 selects a target reflected sound to be generated based on parameters calculated before the selection process. The selection of the target reflected sound may be performed using any of the selection methods disclosed herein.

[0421] 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.

[0422] 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.

[0423] 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.

[0424] 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.

[0425] 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 determination unit 1314 is perceived by the listener as a sound arriving from the reflecting object.

[0426] 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.

[0427] 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.

[0428] 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.

[0429] 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.

[0430] 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.

[0431] 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.

[0432] 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.

[0433] When a sound source object is present on the other side of an obstacle object, the occlusion processing unit generates an audio signal of 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.

[0434] [Example of 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."

[0435] 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 comes are not determined. Therefore, reflected sounds caused by such sound sources may be limited to those selected by the determination 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.

[0436] 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.

[0437] [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.

[0438] 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.

[0439] 25 is a diagram illustrating sound transmission and diffraction. As shown in FIG. 25, 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.

[0440] 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.

[0441] [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.

[0442] 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.

[0443] 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.

[0444] 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.

[0445] The bitstream has a data structure that is made up of, for example, metadata (control information) and an audio signal.

[0446] 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.

[0447] 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.

[0448] 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.

[0449] 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.

[0450] 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.

[0451] 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).

[0452] 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.

[0453] 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.

[0454] 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.

[0455] 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.

[0456] [Examples of Information Included in Metadata] Metadata may be information used to describe a scene represented in a sound space, where a scene is a term that refers to a collection of all elements representing three-dimensional video and sound events in a sound space that is modeled by the stereophonic sound reproduction system 1000 using the metadata.

[0457] 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.

[0458] 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.

[0459] 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.

[0460] 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).

[0461] 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.

[0462] 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.

[0463] 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.

[0464] 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.

[0465] 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.

[0466] 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.

[0467] 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.

[0468] For example, if the sound source area is defined relative to the listener's position and the object's position, it is possible for the listener to perceive sound E coming from the right side of the object and sound F coming from the left side of the object.

[0469] 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.

[0470] 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.

[0471] (Embodiment 2) The following describes embodiment 2. The following mainly describes the differences from embodiment 1, and the description of commonalities will be omitted or simplified.

[0472] [Configuration of the Rendering Unit] First, a description will be given of the configuration of the rendering unit 2300 according to this embodiment. Fig. 26 is a block diagram showing an example of the configuration of the rendering unit 2300 according to this embodiment.

[0473] The rendering unit 2300 includes an analysis unit 2301, a determination unit 2302, and a reproduction unit 2303. As described above, the audio signal processing device according to this embodiment is an example of a decoding device, and the decoding device includes a decoder, which includes the rendering unit 2300. In other words, it can be said that the audio signal processing device according to this embodiment includes the analysis unit 2301, the determination unit 2302, and the reproduction unit 2303. The rendering unit 2300 applies acoustic processing to sound data included in an input signal and outputs the result.

[0474] As in the first embodiment, the input signal is composed of, for example, spatial information, sensor information, and sound data. The spatial information also includes physical information such as the reflection coefficient, transmission coefficient, and diffraction coefficient of non-sound-producing objects (obstacle objects).

[0475] The analysis unit 2301 may perform all or part of the processing performed by the analysis unit 1301 according to embodiment 1. The analysis unit 2301 also creates an audio signal indicating a reflected sound and an audio signal indicating a direct sound, and stores the created audio signals indicating the reflected sound and the audio signals indicating the direct sound.

[0476] In this embodiment, mainly, reflected sound, which is an example of indirect sound, and direct sound, which is an example of predetermined sound, are used. That is, in this embodiment, reflected sound can be used as indirect sound, and direct sound can be used as predetermined sound. Note that, although the description here mainly uses reflected sound and direct sound, the same processing is performed even if indirect sound is used instead of reflected sound, and predetermined sound is used instead of direct sound. Furthermore, examples of indirect sound include reflected sound or diffracted sound, and examples of predetermined sound include direct sound, HOA, or representative sound, which are different from indirect sound.

[0477] The analysis unit 2301 includes a propagation path detection unit 2301a and a memory 2301b.

[0478] The propagation path detector 2301a generates an audio signal indicating a reflected sound and an audio signal indicating a direct sound based on the spatial information and the sound data.

[0479] More specifically, the propagation path detection unit 2301a creates an audio signal indicating reflected sound and an audio signal indicating direct sound based on the position information of the sound source object, the position information of the non-sound-emitting object (obstacle object), the position information and physical information of the listener contained in the spatial information, and the sound data.

[0480] That is, the propagation path detection unit 2301a generates an audio signal generated in a virtual space based on the spatial information and the sound data, and assigns attribute information indicating an attribute that identifies the audio signal to the generated audio signal to generate an audio signal including the attribute information. The attribute is information that indicates whether the audio indicated by the audio signal is direct sound (predetermined sound) or reflected sound (indirect sound).

[0481] A sound that reaches the listener's head directly from a single sound source is called a direct sound, and a sound that is output from the single sound source and then reflected off or diffracted by a non-sound-producing object before reaching the listener's head is called an indirect sound (reflected sound or diffracted sound).

[0482] Furthermore, a direct sound related to an indirect sound refers to a direct sound originating from the same sound source as the indirect sound. An indirect sound related to a direct sound refers to an indirect sound originating from the same sound source as the direct sound.

[0483] An audio signal whose attribute is reflected sound (indirect sound) includes information indicating an audio signal of a direct sound related to the reflected sound (indirect sound).

[0484] The propagation path detection unit 2301a stores the created audio signals in the memory 2301b. Fig. 26 shows audio signals A, B, C, and D stored in the memory 2301b, and also shows the attributes of each of the audio signals A, B, C, and D. Note that the audio signals may be stored in a storage device other than the memory 2301b.

[0485] Furthermore, as in embodiment 1, the propagation path detection unit 2301a may calculate values ​​related to the path taken by each of the direct sound and the reflected sound to reach the listening position, the time it takes for the sound to arrive, the volume at the time of arrival, etc. Similarly, the propagation path detection unit 2301a may calculate information indicating the relationship between the direct sound and the reflected sound, such as a value related to the time difference between the arrival of the direct sound and the reflected sound (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.

[0486] An audio signal whose attribute is reflected sound and an audio signal whose attribute is direct sound may each include information indicating the volume of the sound indicated by the audio signal at the listening position. An audio signal whose attribute is reflected sound may include information indicating the volume (lr) at the time of arrival of the reflected sound as the volume of the reflected sound indicated by the audio signal at the listening position. An audio signal whose attribute is direct sound may include information indicating the volume (ld) at the time of arrival of the direct sound as the volume of the direct sound indicated by the audio signal at the listening position.

[0487] The determination unit 2302 may perform all or part of the processing performed by the determination unit 1302 according to Embodiment 1. The determination unit 2302 also determines whether or not an output signal based on an audio signal created by the analysis unit 2301 (more specifically, the propagation path detection unit 2301 a) is to be output (reproduced) by the reproduction unit 2303.

[0488] The determining unit 2302 includes a classifying unit 2302a, a first determining unit 2302b, and a second determining unit 2302c.

[0489] The classification unit 2302a acquires a voice signal including attribute information created by the propagation path detection unit 2301a and stored in the memory 2301b. The classification unit 2302a outputs the voice signal to the first determination unit 2302b or the second determination unit 2302c in accordance with the attribute specified by the attribute information included in the acquired voice signal.

[0490] When the attribute specified by the attribute information included in the acquired audio signal indicates a direct sound (predetermined sound), the classification unit 2302a outputs the audio signal to the first determination unit 2302b. When the attribute specified by the attribute information included in the acquired audio signal indicates a reflected sound (indirect sound), the classification unit 2302a outputs the audio signal to the second determination unit 2302c.

[0491] The first determination unit 2302b performs a first determination process to determine whether the audio signal output from the classification unit 2302a satisfies a first condition. If the audio signal satisfies the first condition, the first determination unit 2302b outputs the audio signal to the playback unit 2303.

[0492] Furthermore, the first determination unit 2302b does not perform the second determination process. That is, if the attribute specified by the attribute information included in the acquired audio signal is information indicating a predetermined sound, the second determination process is not performed. In this way, when the second determination process is not performed, the amount of calculation and the calculation load can be reduced.

[0493] The second determination unit 2302c performs a first determination process to determine whether the audio signal output from the classification unit 2302a satisfies a first condition, and a second determination process to determine whether the audio signal satisfies a second condition different from the first condition. If the audio signal satisfies both the first and second conditions, the second determination unit 2302c outputs the audio signal to the playback unit 2303.

[0494] The reproduction unit 2303 may perform all or part of the processing performed by the reproduction unit 1303 according to Embodiment 1. The reproduction unit 2303 also acquires the audio signal output from the determination unit 2302, and outputs an output signal based on the acquired audio signal.

[0495] The reproduction unit 2303 includes a first reproduction unit 2303 a and a second reproduction unit 2303 b. The first reproduction unit 2303 a outputs an output signal (first output signal) based on an audio signal whose attribute is a predetermined sound different from indirect sound. The second reproduction unit 2303 b outputs an output signal (second output signal) based on an audio signal whose attribute is indirect sound.

[0496] That is, the first reproduction unit 2303a acquires the audio signal output from the first determination unit 2302b and outputs a first output signal, and the second reproduction unit 2303b acquires the audio signal output from the second determination unit 2302c and outputs a second output signal.

[0497] The first reproduction unit 2303a performs binaural filtering on the acquired audio signal to generate and output a first output signal. The binaural filtering is realized, for example, by processing the acquired audio signal using a head-related transfer function.

[0498] The second reproduction unit 2303b performs binaural filtering and diffusion filtering on the acquired audio signal to generate and output a second output signal. The diffusion filtering is, for example, processing that improves the realism of the indirect sound by diffusing the indirect sound represented by the acquired audio signal. The diffusion filtering is processing that uses a filter that realistically simulates the auditory strength of the diffusion of the sound represented by the acquired audio signal (i.e., simulates the auditory strength of the diffusion of the sound felt by the listener). The diffusion filtering uses a finite impulse filter and / or an infinite impulse filter.

[0499] The first reproduction unit 2303a outputs the first output signal without performing diffusion filtering on the acquired audio signal.

[0500] With the above configuration of the playback unit 2303, the audio signal whose attribute is indirect sound is subjected to diffusion filtering before being output, allowing the listener to hear indirect sound with higher sound quality. Furthermore, since the audio signal whose attribute is predetermined sound is not subjected to diffusion filtering, the amount of calculation and the calculation load are further reduced.

[0501] An example of the operation of the audio signal processing method performed by the audio signal processing device (more specifically, the rendering unit 2300) according to this embodiment will be described below.

[0502] [Example of Operation of Rendering Unit] Fig. 27 is a flowchart showing an example of operation of the audio signal processing device according to this embodiment. Fig. 27 mainly shows processing executed by the rendering unit 2300 included in the audio signal processing device according to this embodiment.

[0503] First, the analyzer 2301 performs an analysis process to analyze an input signal (S501). More specifically, the propagation path detector 2301a of the analyzer 2301 generates an audio signal representing reflected sound and an audio signal representing direct sound based on spatial information and sound data. The propagation path detector 2301a stores the generated audio signals in the memory 2301b.

[0504] The analysis unit 2301 analyzes the input signal and calculates values ​​related 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, as well 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.

[0505] First, the propagation path detector 2301a calculates the characteristics of the direct sound represented by the generated audio signal and the reflected sound represented by the generated audio signal. Specifically, the arrival time and volume of the direct sound and the reflected sound when they reach the listener (listening position) are calculated. Note that the method for calculating these arrival times and volumes can be the method described in the first embodiment.

[0506] Next, the propagation path detection unit 2301a calculates the volume ratio (L), which is the ratio between the volume (ld) when the direct sound arrives and the volume (lr) when the reflected sound arrives, and the time difference (T) between the direct sound and the reflected sound (the time difference (T) when the direct sound and the reflected sound arrive). The volume (ld) when the direct sound arrives means the volume when the direct sound, which is an example of a predetermined sound, arrives at the listening position where the listener is located in the virtual space. In other words, it is the volume of the predetermined sound (direct sound) at the listening position. The volume (lr) when the reflected sound arrives means the volume when the reflected sound, which is an example of an indirect sound, arrives at the listening position. In other words, it is the volume of the indirect sound (reflected sound) at the listening position. In other words, the volume ratio (L) is the volume ratio between the direct sound and the reflected sound (indirect sound) at the listening position. Note that the volume ratio (L) and the time difference (T) can be calculated using the method described in embodiment 1.

[0507] The determination unit 2302 determines the audio signal (S502). More specifically, the determination unit 2302 determines whether or not the audio signal created by the propagation path detection unit 2301a is output by the reproduction unit 2303. That is, the determination unit 2302 performs a determination process (selection process).

[0508] First, the classification unit 2302a acquires the audio signal including the attribute information created by the propagation path detection unit 2301a and stored in the memory 2301b. The classification unit 2302a outputs the audio signal whose attribute is direct sound (predetermined sound) to the first determination unit 2302b, and outputs the audio signal whose attribute is reflected sound (indirect sound) to the second determination unit 2302c.

[0509] The classification unit 2302a also acquires the volume (ld) at the time of arrival of the direct sound, the volume (lr) at the time of arrival of the reflected sound, the volume ratio (L), and the time difference (T) calculated by the propagation path detection unit 2301a.

[0510] Next, the first determination unit 2302b performs a first determination process on the audio signal output from the classification unit 2302a. The second determination unit 2302c performs a first determination process and a second determination process on the audio signal output from the classification unit 2302a. That is, the determination unit 2302 (the second determination unit 2302c) performs the first determination process and the second determination process when the attribute specified by the attribute information included in the acquired audio signal is information indicating reflected sound (indirect sound).

[0511] The first determination process and the second determination process will now be described. The second determination process will be described first. In this embodiment, the second determination process is performed only on audio signals whose attribute is reflected sound (indirect sound).

[0512] The second determination process is a process for determining whether the acquired audio signal satisfies the second condition. In the second determination process, if the volume ratio between the direct sound (predetermined sound) and the reflected sound (indirect sound) when they arrive at the listening position is equal to or greater than a second threshold determined according to the time difference between the direct sound and the reflected sound, the acquired audio signal is determined to satisfy the second condition. In other words, the second determination process corresponds to the process performed by the determination unit 1302 in Embodiment 1. As described above, the time difference is calculated by the propagation path detection unit 2301a, and the second determination unit 2302c acquires the calculated time difference and performs the second determination process.

[0513] As described above, the reflected sound and the direct sound related to the reflected sound are sounds originating from the same sound source.

[0514] The time difference between the direct sound and the reflected sound is, for example, but not limited to, the time difference between the direct sound arrival time (arrival time) and the reflected sound arrival time (arrival time), as described in embodiment 1. The volume ratio between the direct sound and the reflected sound when the reflected sound and the direct sound arrive at the listening position corresponds to the volume ratio (L) in embodiment 1, which is the ratio between the volume (ld) when the direct sound arrives and the volume (lr) when the reflected sound arrives.

[0515] The volume ratio (L) is calculated in the same manner as in the first embodiment.

[0516] The second threshold is a value determined according to the time difference between the direct sound and the reflected sound (indirect sound), in other words, a value that depends on the time difference, and is a value indicated in the threshold data of embodiment 1. The threshold data is expressed as a threshold (second threshold) for whether the reflected sound is perceived or not, for example, in a graph having 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.

[0517] More specifically, the threshold data indicating the second threshold is the data shown in FIGS. 11 to 13, etc.

[0518] In the second judgment process, if the volume ratio between the direct sound and the reflected sound (indirect sound) indicated by the acquired audio signal is greater than or equal to a second threshold, the acquired audio signal is judged to satisfy the second condition.

[0519] Next, the first determination process will be described. In this embodiment, the first determination process is performed on a sound signal whose attribute is indirect sound (reflected sound) and a sound signal whose attribute is information indicating a predetermined sound (direct sound).

[0520] The first determination process is a process for determining whether the acquired audio signal satisfies a first condition. In the first determination process, if the amplitude value of the acquired audio signal is equal to or greater than a first threshold, the acquired audio signal is determined to satisfy the first condition. That is, since the amplitude value of an audio signal corresponds to the volume of the sound represented by the audio signal (reflected sound (indirect sound) or direct sound (predetermined sound)), in the first determination process, if the volume of the sound represented by the acquired audio signal is equal to or greater than a certain level, the acquired audio signal is determined to satisfy the first condition.

[0521] Unlike the second threshold, the first threshold is a constant value that does not depend on the time difference between the direct sound and the reflected sound (indirect sound). The first threshold is a value related to the volume of the acquired audio signal, in other words, a value related to the amplitude value. More specifically, the first 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 with a volume lower than the threshold will not be reproduced. FIG. 28 is a graph showing threshold data indicating the first threshold according to this embodiment. For example, the first threshold is −70 dB. Since the amplitude value of audio signal B shown in FIG. 28 is equal to or greater than the first threshold, audio signal B is determined to satisfy the first condition. Furthermore, since the amplitude value of audio signal A shown in FIG. 28 is less than the first threshold, audio signal A is determined to not satisfy the first condition.

[0522] The first threshold may be set (determined) by an administrator or listener of the virtual space using the threshold setting unit described above, for example.

[0523] The first threshold value used in the first determination unit 2302b and the first threshold value used in the second determination unit 2302c may be the same value or different values.

[0524] The determining unit 2302 performs the first determining process and the second determining process as described above.

[0525] The first determination unit 2302 b then performs a first determination process on the audio signal output from the classification unit 2302 a , and outputs the audio signal to the playback unit 2303 if the audio signal satisfies a first condition.

[0526] The second determination unit 2302c performs a first determination process to determine whether the audio signal output from the classification unit 2302a satisfies a first condition, and a second determination process to determine whether the audio signal satisfies a second condition different from the first condition. In this embodiment, the second determination process is performed after the first determination process is performed.

[0527] Therefore, the second determination unit 2302c first performs a first determination process to determine whether the audio signal output from the classification unit 2302a satisfies a first condition, and if the audio signal satisfies the first condition, performs a second determination process on the audio signal. If the audio signal satisfies the second condition, the second determination unit 2302c outputs the audio signal to the playback unit 2303.

[0528] Then, the reproduction unit 2303 acquires the audio signal output from the determination unit 2302 and outputs an output signal based on the audio signal (S503).

[0529] More specifically, the first reproduction unit 2303a acquires the audio signal output from the first determination unit 2302b and performs binaural filtering on the audio signal to generate and output a first output signal. The second reproduction unit 2303b acquires the audio signal output from the second determination unit 2302c and performs binaural filtering and diffusion filtering on the audio signal to generate and output a second output signal. In other words, the reproduction unit 2303 (more specifically, the second reproduction unit 2303b) outputs an output signal (second output signal) based on the acquired audio signal when the audio signal satisfies the first condition and the second condition.

[0530] If the audio signal does not satisfy the first condition, the first determination unit 2302b does not output the audio signal to the reproduction unit 2303 (first reproduction unit 2303a). In such a case, the first reproduction unit 2303a does not output an output signal based on the audio signal, thereby reducing the amount of calculation and the calculation load.

[0531] Furthermore, the second determination unit 2302c does not output the audio signal to the reproduction unit 2303 (second reproduction unit 2303b) if the audio signal does not satisfy the first condition, and does not output the audio signal to the reproduction unit 2303 (second reproduction unit 2303b) if the audio signal does not satisfy the second condition. In such cases, the second reproduction unit 2303b does not output an output signal based on the audio signal, thereby reducing the amount of calculation and the calculation load.

[0532] As described above, the audio signal processing method according to this embodiment is an audio signal processing method executed by the audio signal processing device (rendering unit 2300), and includes an acquisition step, a determination step, and a reproduction step.

[0533] The acquisition step acquires an audio signal including attribute information that identifies an attribute of the audio signal. The determination step performs a first determination process to determine whether the acquired audio signal satisfies a first condition and a second determination process to determine whether the acquired audio signal satisfies a second condition different from the first condition when the attribute identified by the attribute information included in the acquired audio signal is information indicating indirect sound. The reproduction step outputs an output signal based on the acquired audio signal when the acquired audio signal satisfies the first condition and the second condition.

[0534] In other words, a first determination process and a second determination process are performed on an audio signal whose attribute is indirect sound (reflected sound), and an output signal based on the acquired audio signal is output if the audio signal satisfies the first condition and the second condition. That is, it is appropriately determined whether or not an output signal based on an audio signal whose attribute is indirect sound (reflected sound) is to be output. If an output signal is not to be output, the amount of calculation and the calculation load are reduced. In other words, an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load can be realized.

[0535] In the present embodiment, the first determination process determines that the acquired audio signal satisfies the first condition if the amplitude value of the acquired audio signal is equal to or greater than a first threshold. The second determination process determines that the acquired audio signal satisfies the second condition if the volume ratio of the direct sound and the indirect sound related to the indirect sound when they arrive at a listening position where a listener is located is equal to or greater than a second threshold determined according to the arrival time difference between the direct sound and the indirect sound.

[0536] As a result, in the first determination process, the audio signal is determined to satisfy the first condition if the amplitude value is equal to or greater than a first threshold, and in the second determination process, the audio signal is determined to satisfy the second condition if the volume ratio is equal to or greater than a second threshold. An output signal based on the acquired audio signal is output when such a second condition is satisfied. In other words, it is more appropriately determined whether or not an output signal based on the audio signal is to be output. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0537] While the first determination unit 2302b and the second determination unit 2302c are provided in this embodiment, this is not limiting. For example, the first determination unit 2302b may not be provided, and only the second determination unit 2302c may be provided. In this case, the second determination unit 2302c acquires both an audio signal whose attribute is a predetermined sound (direct sound) and an audio signal whose attribute is an indirect sound (reflected sound), and performs a first determination process on both of them. If the first determination process determines that the audio signal whose attribute is a predetermined sound (direct sound) satisfies the first condition, the audio signal is output to the first reproduction unit 2303a, and the first reproduction unit 2303a outputs a first output signal based on the output audio signal. If the first determination process determines that the audio signal whose attribute is indirect sound (reflected sound) satisfies the first condition, a second determination process is further performed on the audio signal. If the second determination process determines that the audio signal satisfies the second condition, the audio signal is output to the second reproduction unit 2303b, and the second reproduction unit 2303b outputs a second output signal based on the output audio signal. It can be said that a rendering unit that does not include the first determination unit 2302b and that includes the second determination unit 2302c performs substantially the same processing as the rendering unit 2300 according to this embodiment.

[0538] (Embodiment 3) Hereinafter, a description will be given of embodiment 3. The following description will focus on the differences from embodiment 2, and the description of commonalities will be omitted or simplified.

[0539] [Configuration of the Rendering Unit] First, a description will be given of the configuration of the rendering unit 3300 according to this embodiment. Fig. 29 is a block diagram showing an example of the configuration of the rendering unit 3300 according to this embodiment.

[0540] The rendering unit 3300 includes an analysis unit 2301 , a determination unit 3302 , and a reproduction unit 3303 .

[0541] The determining unit 3302 has the same configuration as the determining unit 2302 according to the second embodiment, except that it has a second determining unit 3302c instead of the first determining unit 2302c.

[0542] The reproducing section 3303 has the same configuration as the reproducing section 2303 according to the second embodiment, except that it further includes a gain setting section 3303c.

[0543] The gain setting unit 3303c sets (determines) a gain for the diffusion filter processing in the second reproduction unit 2303b. That is, the gain setting unit 3303c sets (determines) an amplification factor for the amplitude of the audio signal in the diffusion filter processing. In this embodiment, the second reproduction unit 2303b performs the diffusion filter processing using the gain determined by the gain setting unit 3303c.

[0544] For example, the gain setting unit 3303c acquires the gain set (determined) by the administrator or listener of the virtual space, and sets (determines) the acquired gain as the gain in the diffusion filter processing.

[0545] The gain setting unit 3303c outputs the determined gain to the determination unit 3302 (more specifically, the second determination unit 3302c).

[0546] In this embodiment as well, when the attribute specified by the attribute information included in the acquired audio signal is indirect sound, the classification unit 2302a outputs the audio signal to the second determination unit 3302c.

[0547] The second determination unit 3302c performs a first determination process to determine whether or not the audio signal output from the classification unit 2302a satisfies a first condition. In this embodiment, the second determination unit 3302c does not perform the second determination process.

[0548] Before performing the first determination process, the second determination unit 3302c acquires the gain output by the gain setting unit 3303c. The second determination unit 3302c adds the acquired gain to the audio signal output from the classification unit 2302a and performs the first determination process on the audio signal to which the gain has been added. If the audio signal satisfies the first condition, the second determination unit 3302c outputs the audio signal to the playback unit 3303 (more specifically, the second playback unit 2303b).

[0549] (Fourth Embodiment) The following describes a fourth embodiment, focusing on differences from the second embodiment, and omitting or simplifying the description of commonalities.

[0550] [Configuration of the Rendering Unit] First, a description will be given of the configuration of the rendering unit 4300 according to this embodiment. Fig. 30 is a block diagram showing an example of the configuration of the rendering unit 4300 according to this embodiment.

[0551] The rendering unit 4300 includes an analysis unit 2301 , a renderer pipeline unit 4304 , a reproduction unit 4303 , a first gain accumulation unit 4305 , and a second gain accumulation unit 4306 .

[0552] In this embodiment, the explanation will be mainly based on reflected sound, which is an example of indirect sound, and direct sound, which is an example of predetermined sound, but the same processing is performed even if indirect sound is used instead of reflected sound, and predetermined sound is used instead of direct sound.

[0553] In this embodiment, the analysis unit 2301 outputs the audio signal stored in the memory 2301 b to the renderer pipeline unit 4304 .

[0554] The renderer pipeline unit 4304 acquires the audio signal output from the analysis unit 2301. The renderer pipeline unit 4304 acquires, for example, an audio signal whose attribute is indirect sound (reflected sound) (i.e., an audio signal indicating indirect sound (reflected sound)).

[0555] The analyzer 2301 also outputs a reference volume included in spatial information contained in the input signal, more specifically, a reference volume of indirect sound (reflected sound) indicated by the audio signal, to the first gain accumulator 4305, and the first gain accumulator 4305 acquires the reference volume. The renderer pipeline unit 4304 performs one or more first processes (here, multiple first processes), a first determination process, a second determination process, and one or more second processes (here, multiple second processes) on the acquired audio signal.

[0556] The renderer pipeline unit 4304 has one or more first processing units 4304a (here, multiple first processing units 4304a), a determination unit 4302, and one or more second processing units 4304b (here, multiple second processing units 4304b).

[0557] The multiple first processing units 4304a include first processing units 4304a1, first processing units 4304a2, first processing units 4304a3, and first processing units 4304a4. Each of the multiple first processing units 4304a performs a first processing on the acquired audio signal. In this embodiment, the first processing is processing that determines the amount of increase or decrease by which the amplitude of the audio signal is to be amplified when processing based on physical characteristics is performed on the audio signal. Note that the first processing is not limited to this, and does not have to be processing that determines the amount of increase or decrease by which the amplitude of the audio signal is to be amplified.

[0558] The first processes performed by each of the plurality of first processing units 4304a may be different from one another. As an example, each of the first processing units 4304a1 to 4304a4 performs the following first processes.

[0559] When the acquired audio signal has been processed by the initial reflection processing unit 1312, the first processing unit 4304a1 performs, as first processing, processing to determine the amount of increase or decrease by which the amplitude of the audio signal is to be amplified.

[0560] When the acquired audio signal has been processed by the diffraction processing unit, the first processing unit 4304a2 performs, as first processing, processing to determine the amount by which the amplitude of the audio signal is to be increased or decreased.

[0561] When the acquired audio signal has been processed by the distance attenuation processing unit 1313, the first processing unit 4304a3 performs, as first processing, processing to determine the amount of increase or decrease by which the amplitude of the audio signal is to be amplified.

[0562] When the acquired audio signal has been processed by the reverberation processing unit 1311, the first processing unit 4304a4 performs, as first processing, processing to determine the amount by which the amplitude of the audio signal is to be increased or decreased.

[0563] As another example, when processing such as transmission, directionality, sound image localization, or diffusion is performed on the acquired audio signal, the first processing unit 4304a may perform a process of determining the amount of increase or decrease by which to amplify the amplitude of the audio signal as the first processing.

[0564] Each of first processing units 4304 a 1 to 4304 a 4 may perform the first processing described above and output the determined amount of increase or decrease to first gain accumulator 4305 .

[0565] In this embodiment, after the first processing is performed in each of the multiple first processing units 4304a, the acquired audio signal is output to the determination unit 4302, where it is processed.

[0566] The determination unit 4302 performs first determination processing and second determination processing. Note that the first determination processing and second determination processing according to this embodiment will be described again after the first gain accumulator 4305 and the second gain accumulator 4306 have been described.

[0567] The plurality of second processing units 4304b include a second processing unit 4304b1 and a second processing unit 4304b2. Each of the plurality of second processing units 4304b performs a second processing on the acquired audio signal. The second processing is processing that determines the amount of increase or decrease by which the amplitude of the audio signal is to be amplified when processing based on the auditory perceptual characteristics of the listener is performed on the audio signal. Furthermore, the second processing may be a sound quality adjustment function performed based on the listener's selection, such as the listener's preference or convenience, and may involve increasing or decreasing the amplitude of the signal. Note that the second processing is not limited to this and does not have to be processing that determines the amount of increase or decrease by which the amplitude of the audio signal is to be amplified.

[0568] The second processes performed by the second processing units 4304b may be different from each other. As an example, the second processing units 4304b1 and 4304b2 each perform the following second processes.

[0569] When diffusion filtering is performed on the acquired audio signal, the second processing unit 4304b1 performs a process of determining an amount of increase or decrease by which to amplify the amplitude of the acquired audio signal as a second process. The diffusion filtering is a process of improving the reality of the indirect sound represented by the acquired audio signal by diffusing the indirect sound, for example, in the acquired audio signal, and a finite impulse filter and / or an infinite impulse filter is used.

[0570] When the binaural filtering process is performed on the acquired audio signal, the second processing unit 4304b2 performs a process of determining the amount of increase or decrease by which the amplitude of the audio signal is to be amplified as the second processing.

[0571] The second processing by the second processing unit 4304b2 will now be described with reference to Fig. 31 to Fig. 34. Here, the description will be given using an example of a head-related transfer function used in binaural filtering.

[0572] 31 to 34 are diagrams illustrating the energy of head-related transfer functions according to this embodiment. Each of Fig. 31 to 34 shows the listening position P of the listener. Each of Fig. 31 to 34 also shows the front, back, left, right, above, and below the listener.

[0573] Fig. 31 shows a schematic diagram of a sphere centered at the listening position P, and the sphere shows circles along the listener's horizontal plane, median plane, and frontal plane. In Fig. 32, the circle along the listener's horizontal plane is shown in bold, in Fig. 33, the circle along the listener's median plane is shown in bold, and in Fig. 34, the circle along the listener's frontal plane is shown in bold.

[0574] 31 to 34, the direction in which the energy of the head-related transfer function is maximum is indicated by a dotted circle, and the direction in which the energy of the head-related transfer function is minimum is indicated by a hollow circle. Also, each of Fig. 31 to 34 shows the value in which the energy of the head-related transfer function is maximum and the value in which the energy of the head-related transfer function is minimum.

[0575] As shown in Figures 31 to 34, the energy of the head-related transfer function differs for each localization direction. In Figures 31 to 34, the energy of the head-related transfer function is displayed in dB conversion when the energy of a transfer function with a transfer characteristic of 1 is set to 0 dB.

[0576] 31 to 34 show examples of the energy of the head-related transfer function, but the energy varies greatly depending on the head-related transfer function used. In the second processing by the second processing unit 4304b2, processing is performed to determine the amount of increase or decrease depending on the head-related transfer function used.

[0577] Each of the second processing units 4304 b 1 and 4304 b 2 may perform the second processing described above and output the determined increase / decrease amount to the second gain accumulation unit 4306 .

[0578] The first gain accumulator 4305 and the second gain accumulator 4306 will now be described.

[0579] The first gain accumulation unit 4305 calculates a first increase or decrease. The first increase or decrease is an amount of increase or decrease determined by each of the multiple first processes (more specifically, each of the multiple first processing units 4304a), and is a value obtained by accumulating the amount of increase or decrease that amplifies the amplitude of the acquired audio signal. More specifically, the first gain accumulation unit 4305 calculates the first increase or decrease by accumulating the amount of increase or decrease with the reference volume. In other words, the first increase or decrease is a value obtained by accumulating the amount of increase or decrease determined by each of the multiple first processing units 4304a and the reference volume.

[0580] As described above, the multiple first processing units 4304a determine multiple (here, four) increments and decrements for the acquired audio signals. Each of the multiple first processing units 4304a outputs the determined increment or decrement to the first gain accumulator 4305. The first gain accumulator 4305 acquires the output multiple increments and decrements and calculates a first increment or decrement by accumulating the multiple (four) increments and decrements and a reference volume. The first gain accumulator 4305 outputs the calculated first increment or decrement to the second gain accumulator 4306, and the second gain accumulator 4306 acquires the output first increment or decrement.

[0581] The second gain accumulator 4306 calculates a second increase or decrease. The second increase or decrease is an amount of increase or decrease determined by each of the multiple second processes (more specifically, each of the multiple second processing units 4304b) and is an accumulated value of the amount of increase or decrease that amplifies the amplitude of the acquired audio signal. More specifically, the second gain accumulator 4306 accumulates the amount of increase or decrease and further accumulates the first amount of increase or decrease output from the first gain accumulator 4305 to the accumulated amount of increase or decrease. In other words, the second amount of increase or decrease is an accumulated value of the amount of increase or decrease determined by each of the multiple second processing units 4304b and the first amount of increase or decrease.

[0582] As described above, the second processing units 4304b determine multiple (here, two) amounts of increase or decrease for the acquired audio signal. Each of the second processing units 4304b outputs a parameter indicating the determined amount of increase or decrease to the second gain accumulator 4306. The second gain accumulator 4306 acquires the output parameters and calculates the second amount of increase or decrease by accumulating the amount of increase or decrease indicated by each of the multiple parameters and the first amount of increase or decrease.

[0583] Each of the multiple second processing units 4304b may output the increase / decrease amount determined without using the above parameters to the second gain accumulator 4306. In this case, the second gain accumulator 4306 acquires the multiple output increase / decrease amounts and calculates the second increase / decrease amount by accumulating the multiple (two) increase / decrease amounts and the first increase / decrease amount.

[0584] Furthermore, as described above, the second processing may be a sound quality adjustment function selected by the listener, which involves increasing or decreasing the amplitude of the signal. For example, the second increase or decrease amount may include an amplitude increase or decrease amount due to the sound quality adjustment function set by the selection of the listener of the virtual space. In this case, the listener may make a selection, for example, by operating an operation reception unit, the operation reception unit may accept the selection, and the second gain accumulation unit 4306 may calculate the second increase or decrease amount based on the selection accepted by the operation reception unit. For example, the listener may make a selection that satisfies their preferred sound quality, thereby being able to listen to sound with the sound quality of their choice. Furthermore, the second increase or decrease amount may further include an amplitude increase or decrease amount due to the sound quality adjustment function set by the selection of an administrator of the virtual space, instead of the listener of the virtual space.

[0585] The first and second determination processes according to this embodiment will be described again.

[0586] First, the first determination process will be described.

[0587] The determination unit 4302 performs a first determination process on the audio signal acquired by the renderer pipeline unit 4304. In the first determination process, if the first increase / decrease amount (G1) calculated by the first gain accumulation unit 4305 is equal to or greater than a first threshold (T1), it is determined that the acquired audio signal satisfies a first condition.

[0588] The first threshold value according to the present embodiment is a constant value related to the volume of the audio signal, in other words, a value related to the amplitude value. In other words, the first threshold value according to the present embodiment is the same as the first threshold value described in the second embodiment.

[0589] In the first determination process according to the second embodiment, it is determined that the acquired audio signal satisfies the first condition when the amplitude value of the acquired audio signal is equal to or greater than the first threshold. That is, the first determination process according to the second embodiment is the same as the first determination process according to the second embodiment, except that the amplitude value of the acquired audio signal is changed to the first increase / decrease amount (G1).

[0590] The second determination process will now be described.

[0591] The determination unit 4302 performs a second determination process on the audio signal acquired by the renderer pipeline unit 4304. In the second determination process, if the value according to the second increase / decrease amount calculated by the second gain accumulation unit 4306 is equal to or greater than a second threshold, it is determined that the acquired audio signal satisfies the second condition.

[0592] The second threshold value according to this embodiment is a value different from the first threshold value and corresponds to the second threshold value described in embodiment 2. That is, the second threshold value is a value determined according to the time difference between the arrival of a direct sound related to an indirect sound (reflected sound) indicated by an acquired audio signal and the arrival of the indirect sound (reflected sound).

[0593] The second increase / decrease amount calculated by the second gain accumulation unit 4306 corresponds to the volume (Ir) at the time of arrival of the reflected sound shown in embodiments 1 and 2, that is, it indicates the volume when the reflected sound indicated by the acquired audio signal arrives at the listening position.

[0594] The value according to the second amount of increase or decrease is the ratio (volume ratio) between the volume of the direct sound related to the indirect sound (reflected sound) indicated by the acquired audio signal and the second amount of increase or decrease calculated by the second gain accumulation unit 4306. In other words, the value according to the second amount of increase or decrease corresponds to the volume ratio (L) shown in the second embodiment, which is the ratio between the volume (ld) when the direct sound arrives and the volume (lr) when the reflected sound arrives.

[0595] In the second determination process according to the second embodiment, if the volume ratio (L) between the direct sound and the indirect sound when the indirect sound and the direct sound arrive at the listening position is equal to or greater than a second threshold, the acquired audio signal is determined to satisfy the second condition. That is, the second determination process according to the present embodiment is the same as the second determination process according to the second embodiment, except that the volume ratio between the direct sound and the indirect sound is changed to a value according to the second increase / decrease amount.

[0596] Furthermore, the second increase / decrease amount and the value corresponding to the second increase / decrease amount in this embodiment can be calculated in the same manner as the volume (Ir) and volume ratio (L) at the time of arrival of the reflected sound described in embodiment 2.

[0597] In this embodiment, if the first determination process determines that the acquired audio signal does not satisfy the first condition, the second determination process is not performed. On the other hand, if the first determination process determines that the acquired audio signal satisfies the first condition, the second determination process is performed.

[0598] As described above, the second amount of increase or decrease is calculated before the second determination process is performed. The second process performed by the second processing unit 4304b uses processing based on the auditory perceptual characteristics of the listener, and therefore determines the amount of increase or decrease independently of the audio signal (i.e., statically). Therefore, the second amount of increase or decrease can be calculated before the second determination process is performed.

[0599] Then, when it is determined in the second determination process that the acquired audio signal satisfies the second condition, the determination unit 4302 outputs the acquired audio signal to a plurality of second processing units 4304b, and further, the renderer pipeline unit 4304 outputs the acquired audio signal to the playback unit 4303. In this case, the renderer pipeline unit 4304 may output the second increase / decrease amount calculated by the second gain accumulation unit 4306 to the playback unit 4303. Then, processing is performed in the playback unit 4303.

[0600] The playback unit 4303 acquires the audio signal output from the renderer pipeline unit 4304 and the second increase / decrease amount. The playback unit 4303 outputs an output signal based on the acquired audio signal. More specifically, the playback unit 4303 may multiply the acquired audio signal by the acquired second increase / decrease amount to generate and output the output signal. As such, in this embodiment, the playback unit 4303 outputs the output signal when the output audio signal satisfies both the first condition and the second condition.

[0601] An example of the operation of the audio signal processing method performed by the audio signal processing device (more specifically, the rendering unit 4300) according to this embodiment will be described below.

[0602] [Example of Operation of Rendering Unit] Fig. 35 is a flowchart showing an example of operation of the audio signal processing device according to this embodiment. Fig. 35 mainly shows processing executed by the rendering unit 4300 included in the audio signal processing device according to this embodiment.

[0603] First, the analysis unit 2301 performs an analysis process to analyze the input signal (S601).

[0604] Furthermore, the renderer pipeline unit 4304 acquires the audio signal output from the analysis unit 2301 (S602). Note that the renderer pipeline unit 4304 acquires an audio signal that indicates, for example, indirect sound.

[0605] Then, each of the first processing units 4304a determines an amount of increase or decrease by which to amplify the amplitude of the acquired audio signal (S603). Each of the first processing units 4304a outputs the determined amount of increase or decrease to the first gain accumulator 4305. In addition, the analyzer 2301 outputs the reference volume of the indirect sound (reflected sound) to the first gain accumulator 4305.

[0606] The first gain accumulator 4305 acquires the output multiple (four) increments and decrements and the reference volume, and calculates the first increment or decrement by accumulating the acquired multiple increments and decrements and the acquired reference volume (S604). The first gain accumulator 4305 then outputs the calculated first increment or decrement to the determination unit 4302. The first gain accumulator 4305 also outputs the calculated first increment or decrement to the second gain accumulator 4306.

[0607] Next, each of the second processing units 4304b determines an amount of increase or decrease by which to amplify the amplitude of the acquired audio signal (S605). Each of the second processing units 4304b outputs the determined amount of increase or decrease to the second gain accumulator 4306.

[0608] The second gain accumulator 4306 acquires the outputted multiple (two) amounts of increase or decrease and the calculated first amount of increase or decrease, and accumulates the acquired multiple amounts of increase or decrease and the acquired first amount of increase or decrease to calculate the second amount of increase or decrease (S606).The second gain accumulator 4306 then outputs the calculated second amount of increase or decrease to the determiner 4302.

[0609] The determination unit 4302 performs a first determination process (S607). That is, the determination unit 4302 determines whether the calculated first increase / decrease amount is equal to or greater than a first threshold, thereby determining whether the acquired audio signal satisfies a first condition.

[0610] The determination unit 4302 performs a second determination process (S608). That is, the determination unit 4302 determines whether the value corresponding to the calculated second increase / decrease amount is equal to or greater than a second threshold, thereby determining whether the acquired audio signal satisfies the second condition.

[0611] In this embodiment, the second determination process is performed when the acquired audio signal satisfies the first condition.

[0612] Furthermore, if the acquired audio signal satisfies the second condition, the reproduction unit 4303 outputs an output signal based on the acquired audio signal (S609).

[0613] Furthermore, if the audio signal does not satisfy the first condition, the audio signal is not output to the reproduction unit 4303, and if the audio signal does not satisfy the second condition, the audio signal is not output to the reproduction unit 4303. In such cases, the reproduction unit 4303 does not output an output signal based on the audio signal, thereby reducing the amount of calculation and the calculation load.

[0614] In this embodiment, the renderer pipeline unit 4304 may include the classification unit 2302a described in embodiment 2. The classification unit 2302a may be provided, for example, between the plurality of first processing units 4304a and the determination unit 4302, but is not limited to this.

[0615] As described above, the audio signal processing method of this embodiment is an audio signal processing method executed by an audio signal processing device, and includes an acquisition step, a renderer pipeline step, a playback step, a first gain accumulation step, and a second gain accumulation step.

[0616] The acquisition step acquires an audio signal indicative of indirect sound, and the renderer pipeline step performs one or more first processes, a first determination process, a second determination process, and one or more second processes different from the one or more first processes on the acquired audio signal.

[0617] The first gain accumulation step includes a reproduction step of outputting an output signal based on the acquired audio signal, and a step of accumulating the increments and decrements determined by each of the one or more first processes, by which the amplitude of the acquired audio signal is amplified, to calculate a first increment and decrement. The second gain accumulation step includes accumulating the increments and decrements determined by each of the one or more second processes, by which the amplitude of the acquired audio signal is amplified, to calculate a second increment and decrement. In the first determination step, if the calculated first increment and decrement is equal to or greater than a first threshold, it is determined that the acquired audio signal satisfies the first condition. If it is determined that the acquired audio signal does not satisfy the first condition, the second determination step is not performed. If it is determined that the acquired audio signal satisfies the first condition, the second determination step is performed. In the second determination step, if a value corresponding to the calculated second increment and decrement is equal to or greater than a second threshold different from the first threshold, it is determined that the acquired audio signal satisfies the second condition, and if it is determined that the acquired audio signal satisfies the second condition, the reproduction step is performed.

[0618] As a result, the first determination process and the second determination process are performed on the audio signal indicating indirect sound, and an output signal based on the acquired audio signal is output if the audio signal satisfies the first condition and the second condition. That is, it is appropriately determined whether or not an output signal based on the audio signal indicating indirect sound is to be output. If an output signal is not to be output, the amount of calculation and the calculation load are reduced. In other words, it is possible to realize an audio signal processing method that can appropriately reduce the amount of calculation and the calculation load.

[0619] In this embodiment, the second gain calculation step includes a second gain calculation step that amplifies the amplitude of the acquired audio signal and is a diffusion filter process that diffuses the indirect sound to improve the realism of the indirect sound. The second gain calculation step includes a second gain calculation step that amplifies the amplitude of the acquired audio signal and is a ratio between the volume of the direct sound associated with the indirect sound and the second gain calculation step. The second threshold value is determined based on the time difference between the arrival of the direct sound and the arrival of the indirect sound.

[0620] As a result, in the second determination process, if the ratio is equal to or greater than the second threshold, it is determined that the audio signal indicating indirect sound satisfies the second condition, and an output signal based on the acquired audio signal when such second condition is satisfied is output. In other words, it is more appropriately determined whether or not an output signal based on an audio signal indicating indirect sound is to be output. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0621] In this embodiment, the second gain calculation step includes an amplitude increase / decrease amount due to a sound quality adjustment function selected by a listener. The value is a ratio between the volume of the direct sound associated with the indirect sound and the second gain calculation step, and the second threshold is determined according to the time difference between the arrival of the direct sound and the arrival of the indirect sound.

[0622] This allows the listener to select the sound quality that suits their taste, and thus to listen to the sound with the sound quality that they prefer.

[0623] In the present embodiment, the first threshold value is a value related to the volume.

[0624] This makes it possible to realize an audio signal processing method that can use a value related to the volume as the first threshold value.

[0625] In this embodiment, in the second gain accumulation step, parameters are acquired that indicate the amount of increase or decrease determined by each of one or more second processes and that amplify the amplitude of the acquired audio signal, and the second amount of increase or decrease is calculated according to the acquired multiple parameters.

[0626] This allows the second increase or decrease amount to be calculated according to the parameter, thereby more appropriately determining whether or not to output an output signal based on an audio signal whose attribute is indirect sound. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0627] (Embodiment 5) The following describes embodiment 5. The following description will focus on the differences from embodiment 4, and the description of commonalities will be omitted or simplified.

[0628] [Configuration of the Rendering Unit] First, a description will be given of the configuration of the rendering unit 5300 according to this embodiment. Fig. 36 is a block diagram showing an example of the configuration of the rendering unit 5300 according to this embodiment.

[0629] The rendering unit 5300 has the same configuration as the rendering unit 4300 according to the fourth embodiment, except that it further includes a changing unit 5307 and an invalidating unit 5308 .

[0630] The change unit 5307 performs processing to set (determine) a first threshold value used in the first determination processing. For example, the change unit 5307 outputs an instruction to set the first threshold value to the determination unit 4302. The determination unit 4302 acquires the output instruction and sets the first threshold value in accordance with the instruction. Therefore, in the first determination processing, if the calculated first increase / decrease amount is equal to or greater than the set first threshold value, it is determined that the acquired audio signal satisfies the first condition.

[0631] The invalidation unit 5308 performs invalidation processing to invalidate the second determination processing. By performing the invalidation processing, the second determination processing is invalidated, that is, the second determination processing is not performed. For example, the invalidation unit 5308 outputs an instruction indicating the invalidation processing to the determination unit 4302. By acquiring the output instruction, the determination unit 4302 does not perform the second determination processing.

[0632] Here, in this embodiment, a case will be described in which the audio signal acquired in the first determination process is determined to satisfy the first condition and the invalidation process is performed. In this case, the determination unit 4302 outputs the acquired audio signal to multiple second processing units 4304b, and the renderer pipeline unit 4304 outputs the acquired audio signal to the playback unit 4303. That is, in the fourth embodiment, the playback unit 4303 outputs an output signal when the output audio signal satisfies both the first and second conditions, but in this case, the playback unit 4303 outputs an output signal when the output audio signal satisfies the first condition.

[0633] The first threshold value set by the change unit 5307 may have a different value when the invalidation process is performed and when the invalidation process is not performed. More specifically, the first threshold value set when the invalidation process is performed may be larger than the first threshold value set when the invalidation process is not performed. The setting of the first threshold value and the influence of the invalidation process will be described below with reference to FIG. 37 .

[0634] FIG. 37 is a diagram showing a table for explaining the setting of the first threshold value and the influence of the invalidation process according to this embodiment.

[0635] Fig. 37 shows a case where the first threshold is increased ("first threshold: large") and a case where the first threshold is decreased ("first threshold: small") by setting the first threshold. Fig. 37 also shows a case where the disabling process is performed ("second determination process disabled") and a case where the disabling process is not performed ("second determination process enabled"). Fig. 37 also shows "processing complexity," "sound quality maintenance," and "number of culls."

[0636] "Processing complexity" indicates the complexity of processing in the rendering unit 5300. When the "processing complexity" is low, that is, when the processing is easy, "O" is shown, and when the "processing complexity" is high, that is, when the processing is difficult, "X" is shown.

[0637] "Maintain sound quality" indicates whether or not the sound quality of the sound of the output signal output from the playback unit 4303 is maintained. If the sound quality is maintained, "O" is shown, and if the sound quality is reduced, "X" is shown.

[0638] The "thinning number" indicates the number of audio signals (more specifically, multiple audio signals) acquired by the renderer pipeline unit 4304 that are not output as output signals from the playback unit 4303. When the "thinning number" is large, that is, when the number of audio signals that are not output is large and the amount of calculation and the calculation load are reduced, "O" is shown, and when the "thinning number" is small, that is, when the number of audio signals that are not output is small and the amount of calculation and the calculation load are difficult to reduce, "X" is shown.

[0639] 37, when "first threshold: large" and "second determination process disabled", "processing complexity" is O, "sound quality maintenance" is X, and "number of culls" is O. That is, in this case, the processing in the rendering unit 5300 is easy, and the amount of calculation and the calculation load can be sufficiently reduced.

[0640] Similarly, when "first threshold: large" and "second determination process enabled", it is shown that "processing complexity" is X, "sound quality maintenance" is X, and "number of culls" is O. That is, in this case, the amount of calculation and the calculation load can be sufficiently reduced.

[0641] Similarly, when "first threshold: small" and "second determination process disabled", it is shown that "processing complexity" is O, "sound quality maintenance" is O, and "number of culls" is X. In other words, in this case, high sound quality can be maintained.

[0642] Similarly, when "first threshold: small" and "second determination process enabled," it is shown that "processing complexity" is X, "sound quality maintenance" is O, and "number of culls" is O. In other words, in this case, high sound quality is maintained, and the amount of calculation and the calculation load can be sufficiently reduced.

[0643] Whether the change unit 5307 performs the process of setting the first threshold and whether the invalidation unit 5308 performs the invalidation process may be set (determined) by the administrator or listener of the virtual space. Alternatively, the space information may include information indicating whether the change unit 5307 performs the process of setting the first threshold and whether the invalidation unit 5308 performs the invalidation process, and the settings may be determined based on the information. This allows the administrator or listener of the virtual space to select the most appropriate effect through trial and error from among the various effects shown in FIG. 37 . Here, it is desirable that the respective setting areas be adjacent to each other to effectively perform the trial and error process. That is, it is desirable that the storage area storing the value indicating the first threshold set by the change unit 5307 and the storage area storing the signal instructing the invalidation process be adjacent areas. This is because the adjacent storage areas not only make it easier to visually grasp the setting values, but also have the special effect of allowing both values ​​to be set simultaneously (with a single memory access process). In other words, the value indicating the first threshold value and the signal instructing the invalidation process to be performed are linked and placed in the upper and lower bit fields of an area that can be accessed by a single address, and the series of data so placed is written in a single memory access, thereby allowing the two pieces of data to be set simultaneously.

[0644] Furthermore, the process of setting the first threshold value by the change unit 5307 and the invalidation process by the invalidation unit 5308 may be performed when updating the thread.

[0645] As described above, the audio signal processing method according to the present embodiment includes a changing step of setting a first threshold and a disabling step of performing a disabling process of disabling the second determination process. In the first determination process, if the calculated first increase / decrease amount is equal to or greater than the set first threshold, it is determined that the acquired audio signal satisfies the first condition. If the disabling process is performed, the reproduction step is performed when it is determined that the acquired audio signal satisfies the first condition.

[0646] As a result, the second determination process is not performed due to the invalidation process, and the amount of calculation and the calculation load required to perform the second determination process can be reduced. In other words, it is possible to realize an audio signal processing method that can more appropriately reduce the amount of calculation and the calculation load.

[0647] In the audio signal processing method according to the present embodiment, the first threshold value set when the invalidation process is performed is greater than the first threshold value set when the invalidation process is not performed.

[0648] This makes it possible to realize an audio signal processing method that can set the magnitude of the first threshold depending on whether or not the invalidation process is performed.

[0649] (Supplementary Note) The aspects grasped based on the present disclosure are not limited to the embodiments, and may be implemented with various modifications.

[0650] 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.

[0651] 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.

[0652] 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.

[0653] 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.

[0654] 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.

[0655] Furthermore, for example, in the embodiments, the cases where the aspects understood based on the present disclosure are implemented as an audio signal 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 signal processing method, an encoding method, or a decoding method.

[0656] For example, a program for executing the above-described audio signal 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.

[0657] Furthermore, a program for executing the above-described audio signal 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.

[0658] 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.

[0659] 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.

[0660] Furthermore, the FPGA, CPU, etc. may download all or part of the software for realizing the audio signal 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.

[0661] 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. Then, this device and the signal processing device or the signal processing server may perform the audio signal processing method, encoding method, or decoding method described in the present disclosure.

[0662] For example, the audio signal processing device, encoding device, or decoding device in the present disclosure may include an FPGA, a CPU, etc. Furthermore, the audio signal 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.

[0663] 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 audio signal processing device, encoding device, or decoding device described in the present disclosure by installing the software. Note that the terminal or device may be connected to the server via a network and the software may be installed.

[0664] 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 audio signal processing method described using the embodiment.

[0665] 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.

[0666] 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.

[0667] The present disclosure includes aspects that are applicable to, for example, an audio signal processing device, an encoding device, a decoding device, or a terminal or device that includes any of these devices.

[0668] 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, 2301b 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, 2300, 3300, 4300, 5300 Rendering unit 1301, 2301 Analysis unit 1302, 1314, 2302, 3302, 4302 Determination unit 1303, 2303, 3303, 4303 Reproduction 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 2301a Propagation path detection unit 2302a Classification unit 2302b First determination unit 2302c, 3302c Second determination unit 2303a First reproduction unit 2303b Second reproduction unit 3303c Gain setting unit 4304 Renderer pipeline unit 4304a, 4304a1, 4304a2, 4304a3, 4304a4 First processing units 4304b, 4304b1, 4304b2 Second processing units 4305 First gain accumulation unit 4306 Second gain accumulation unit 5307 Change unit 5308 Invalidation unit

Claims

1. A method for processing audio signals performed by an audio signal processing device, Acquisition step of acquiring an audio signal which includes attribute information that identifies the attributes of the audio signal, The method includes a determination step of determining whether the acquired audio signal satisfies a first condition and whether it satisfies a second condition different from the first condition, when the attribute identified by the attribute information contained in the acquired audio signal is information indicating indirect sound, In the determination step, if the acquired audio signal does not satisfy at least one of the first and second conditions, the acquired audio signal is determined to be excluded from output. Audio signal processing method.

2. The aforementioned indirect sound is a reflected sound. The audio signal processing method according to claim 1.

3. If the attribute identified by the attribute information included in the acquired audio signal is information indicating a predetermined sound different from the indirect sound, In the aforementioned determination step, the second determination process is not performed. The audio signal processing method according to claim 1.

4. The aforementioned predetermined sound is a direct sound. The audio signal processing method according to claim 3.

5. In the first determination process, if the amplitude value of the acquired audio signal is greater than or equal to a first threshold, it is determined that the acquired audio signal satisfies the first condition. In the second determination process, if the volume ratio of the direct sound related to the indirect sound and the indirect sound when they arrive at the listening position where the listener is located is greater than or equal to a second threshold determined according to the time difference in the arrival of the direct sound and the indirect sound, the acquired audio signal is determined to satisfy the second condition. The audio signal processing method according to claim 1.

6. A first playback step which outputs a first output signal based on the audio signal whose attribute is a predetermined sound different from the indirect sound, A second playback step which outputs a second output signal based on the audio signal whose attribute is the indirect sound, It further includes, In the second playback step, the acquired audio signal is subjected to a diffusion filter process that improves the realism of the indirect sound by diffusing the indirect sound, thereby outputting the second output signal. In the first playback step, the acquired audio signal is output without applying the diffusion filter processing, The audio signal processing method according to claim 1.

7. After the first determination process is performed, the second determination process is performed. The audio signal processing method according to claim 1.

8. A method for processing audio signals performed by an audio signal processing device, The acquisition step involves acquiring an audio signal, The renderer pipeline step includes one or more first processes, a first determination process, a second determination process, and one or more second processes different from the one or more first processes, with respect to the acquired audio signal. The first determination process is a process that determines whether the audio signal satisfies the first condition, The first condition is a condition based on a first increase / decrease value which is a value corresponding to a first increase / decrease amount that indicates the increase / decrease amount related to one or more first processes, The second determination process is a process that determines whether the audio signal satisfies the second condition, The second condition is a condition based on a second increase / decrease value, which is a value corresponding to a second increase / decrease amount that indicates the increase / decrease amount related to the one or more second processes, The increase or decrease amount for the one or more first processes mentioned above indicates an amount that amplifies the amplitude of the acquired audio signal. The increase or decrease amount for the first or more second processes mentioned above indicates the amount by which the amplitude of the output audio signal is amplified. If it is determined that the first or second condition is not met, the one or more second processes described above will not be performed on the audio signal. Audio signal processing method.

9. The second increase / decrease amount is an increase / decrease amount when diffusion filtering is performed to improve the realism of the indirect sound by diffusing the indirect sound, and includes an increase / decrease amount that amplifies the amplitude of the acquired audio signal. The value corresponding to the second increase or decrease is the ratio of the volume of the direct sound related to the indirect sound to the second increase or decrease. The second threshold is determined according to the time difference between the arrival of the direct sound and the indirect sound. The audio signal processing method according to claim 8.

10. The second increase / decrease amount includes the amount of amplitude increase / decrease by the sound quality adjustment function set by the listener's selection, The value corresponding to the second increase or decrease is the ratio of the volume of the direct sound related to the indirect sound to the second increase or decrease. The second threshold is determined according to the time difference between the arrival of the direct sound and the indirect sound. The audio signal processing method according to claim 8.

11. The first threshold is a value related to volume. The audio signal processing method according to claim 8.

12. A step of obtaining a parameter that indicates the increase or decrease amount for the one or more second processes described above, which amplifies the amplitude of the acquired audio signal, The further step includes calculating the second increase or decrease amount according to a plurality of parameters obtained, The audio signal processing method according to claim 8.

13. A change step to set the first threshold, The process includes a disabling step which involves performing a disabling process to disable the second determination process, In the first determination process, if the first increase / decrease amount is greater than or equal to the set first threshold, it is determined that the acquired audio signal satisfies the first condition. If the invalidation process is performed, and it is determined that the acquired audio signal satisfies the first condition, then one or more of the second processes are performed. The audio signal processing method according to claim 8.

14. The storage area for storing the value indicating the first threshold set in the modification step and the storage area for storing the signal instructing the execution of the invalidation process are adjacent areas. The audio signal processing method according to claim 13.

15. The first threshold set when the disabling process is performed is greater than the first threshold set when the disabling process is not performed. The audio signal processing method according to claim 13.

16. An acquisition unit that acquires an audio signal which includes attribute information that identifies the attributes of the audio signal, The system includes a determination unit that performs a first determination process to determine whether the acquired audio signal satisfies a first condition and a second determination process to determine whether it satisfies a second condition different from the first condition, when the attribute identified by the attribute information contained in the acquired audio signal is information indicating indirect sound, The determination unit determines that the acquired audio signal is not to be output if it does not satisfy at least one of the first and second conditions. Audio signal processing device.

17. The first increase / decrease value is a value based on the cumulative value of the first increase / decrease amount. The audio signal processing method according to claim 8.

18. The first increase / decrease value is further calculated by accumulating the reference volume. The audio signal processing method according to claim 17.

19. The second increase / decrease value is a value based on the cumulative value of the second increase / decrease amount. The audio signal processing method according to claim 8.

20. The second increase / decrease value is further calculated by accumulating the first increase / decrease value. The audio signal processing method according to claim 19.

21. In the first determination process, if the value based on the first increase / decrease value is greater than or equal to the first threshold, it is determined that the acquired audio signal satisfies the first condition. The audio signal processing method according to claim 8.

22. In the second determination process, if the value based on the second increase / decrease value is greater than or equal to the second threshold, it is determined that the acquired audio signal satisfies the second condition. The audio signal processing method according to claim 8.

23. If it is determined that the audio signal satisfies the first condition, the second determination process is performed. The audio signal processing method according to claim 8.

24. A computer program for causing a computer to execute the audio signal processing method described in any one of claims 1 to 15 and 17 to 23.