WO2011070956A1 - オーディオデータ処理装置、オーディオ装置、オーディオデータ処理方法、プログラム及び当該プログラムを記録した記録媒体 - Google Patents
オーディオデータ処理装置、オーディオ装置、オーディオデータ処理方法、プログラム及び当該プログラムを記録した記録媒体 Download PDFInfo
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- 238000012545 processing Methods 0.000 title claims abstract description 65
- 238000003672 processing method Methods 0.000 title claims description 6
- 238000012937 correction Methods 0.000 claims abstract description 13
- 230000005236 sound signal Effects 0.000 claims description 71
- 238000000926 separation method Methods 0.000 claims description 26
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- 238000005070 sampling Methods 0.000 description 7
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- 230000015572 biosynthetic process Effects 0.000 description 3
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
- H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
- H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
- H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
- H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
- H04R2201/403—Linear arrays of transducers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
- H04S2400/11—Positioning of individual sound objects, e.g. moving airplane, within a sound field
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
- H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
Definitions
- the present invention relates to an audio data processing device, an audio device, an audio data processing method, a program, and a recording medium on which the program is recorded.
- WFS wave field synthesis
- a listener who is listening to sound facing the speaker array in the acoustic space provided by the WFS is actually a sound source (hereinafter referred to as a sound source) in which sound radiated from the speaker array is virtually present behind the speaker array. (Referred to as “virtual sound source”) (see FIG. 1 for example).
- Patent Document 1 describes a system that enables a virtual sound source to move.
- the Doppler effect is known as a physical phenomenon in which the sound wave frequency is observed differently depending on the relative speed of the sound source that is the sound wave generation source and the listener.
- the vibrations of the sound waves are increased and the frequency is increased, and when moving away from the observer, the vibrations of the sound waves are increased and decreased. .
- Non-Patent Document 1 is based on the premise that the virtual sound source is fixed without moving, and the Doppler effect generated with the movement of the virtual sound source has not been studied.
- the number of waves of the audio signal that is the basis of the sound emitted by the speaker changes, and the waveform is distorted by the change in the number of waves. Will occur.
- distortion occurs in the waveform, the listener perceives it as noise, so it is necessary to take measures to eliminate the waveform distortion. Details of the waveform distortion will be described later.
- the one described in Patent Document 1 takes into account the Doppler effect generated with the movement of the virtual sound source, and from the appropriate sample data in a certain segment in the audio data that is the basis of the audio signal, The weighting coefficient for the audio data in the range up to the appropriate sample data is changed, and the audio data in the range is corrected.
- segment is a unit of processing audio data.
- the present invention has been made in view of such a problem, and is an audio data processing apparatus or the like that specifies a distortion portion in audio data and corrects the distortion of the specified waveform. It is an object of the present invention to provide an audio data processing apparatus and the like that can output audio data without causing it.
- the audio data processing apparatus of the present invention inputs audio data corresponding to sound emitted by a moving virtual sound source, the position of the virtual sound source, and the position of a speaker that emits sound based on the audio data, and the position of the virtual sound source and
- the first distance and the second distance are calculated from the position of the speaker to the position of the virtual sound source at successive times.
- the specifying means for specifying a distortion portion in the audio data at the previous and subsequent time points, and the audio data of the specified portion as a function Correction means for correcting by the interpolation used.
- the audio data includes sample data
- the specifying unit specifies a repetition portion and a missing portion of the sample data resulting from the separation and approach of the virtual sound source to the speaker, and the correction
- the means corrects the identified repeated portion and missing portion by interpolation using a function.
- the interpolation using the function is linear interpolation.
- the portion for performing the correction is a difference in time width in which sound waves propagate through the first and second distances, or a time width proportional to the difference.
- the audio device of the present invention uses audio data corresponding to sound emitted by a moving virtual sound source, a position of the virtual sound source, and a speaker position that emits sound based on the audio data, and uses the position of the virtual sound source and the speaker.
- the digital content input unit that inputs the digital content including the audio data and the position of the virtual sound source, and the digital content input by the digital content input unit are analyzed, Based on the content information separation unit that separates audio data and virtual sound source position data included in the digital content, the virtual sound source position data and the speaker position data separated by the content information separation unit, content An audio data processing unit that corrects the audio data separated by the information separation unit; and an audio signal generation unit that converts the corrected audio data into an audio signal and outputs the audio signal to the speaker.
- the digital content input unit inputs digital content from a recording medium storing digital content, a server that distributes digital content via a network, or a broadcasting station that broadcasts digital content.
- the audio data corresponding to the sound emitted by the moving virtual sound source, the position of the virtual sound source and the position of the speaker that emits sound based on the audio data are input, and the position of the virtual sound source and the In the audio data processing method in the audio data processing apparatus that corrects the audio data based on the position of the speaker, the first distance and the second distance from the position of the speaker to the position of the virtual sound source at successive times, respectively.
- the program of the present invention provides the audio data corresponding to the sound emitted by the moving sound source based on the position of the virtual sound source formed by the sound emitted from the speaker that inputs the audio signal corresponding to the audio data and the position of the speaker. And calculating a first distance and a second distance from the position of the speaker to the position of the virtual sound source at a point in time, and the first distance and the second distance. If the two distances are different, a step of specifying a distortion portion in the audio data at the previous and subsequent time points and a step of correcting the audio data of the specified portion by interpolation using a function are executed.
- the recording medium of the present invention records the aforementioned program.
- the location of the waveform distortion is specified according to the approach and separation of the virtual sound source from the speaker, and then the specified waveform distortion is corrected by interpolation using a function. Therefore, the audio data can be corrected and output without delay.
- the repeated portion and missing portion of the sample data due to the separation and approach of the virtual sound source to the speaker are specified, and the correcting means is the specified repetition by interpolation using a function. Since the portion and the missing portion are corrected, the audio data can be corrected and output without delay.
- Audio data in order to identify the location of the waveform distortion according to the approach and separation of the virtual sound source to the speaker, and then to correct the identified waveform distortion by linear interpolation, Audio data can be corrected and output without delay.
- the location of the waveform distortion is specified according to the approach and the separation of the virtual sound source from the speaker, and then the specified waveform distortion is corrected by interpolation using a function. Audio data can be corrected and output without delay.
- the location of the waveform distortion is specified according to the approach and separation of the virtual sound source to the speaker, and then the specified waveform distortion is corrected by interpolation using a function. Therefore, the audio data can be corrected and output without delay.
- Audio data can be corrected and output without delay.
- the location of the waveform distortion is specified according to the approach and separation of the virtual sound source from the speaker, and then the specified waveform distortion is performed by interpolation using a function. Therefore, audio data can be corrected and output without delay.
- the audio data processing apparatus and the like it is possible to correct the distortion of the audio data caused by the approach or separation of the virtual sound source from the speaker without delay, and output the corrected audio data.
- FIG. 6 is an explanatory diagram of an example of an audio signal waveform obtained by combining the audio signal waveform formed by the audio data shown in FIG. 4 and the audio signal waveform formed by the audio data shown in FIG. 5.
- FIG. 9 is an explanatory diagram of an example of an audio signal waveform obtained by combining the audio signal waveform formed by the audio data shown in FIG. 7 and the audio signal waveform formed by the audio data shown in FIG. 8.
- FIG. 1 is a block diagram illustrating a configuration example of an audio apparatus including an audio data processing unit according to Embodiment 1.
- FIG. 3 is a block diagram illustrating an internal configuration example of an audio data processing unit according to Embodiment 1.
- FIG. It is explanatory drawing of the example of 1 structure of an input audio data buffer. It is explanatory drawing of the example of 1 structure of the sound wave propagation time data buffer. It is explanatory drawing of an example of the audio signal waveform formed with the audio data after correction
- 3 is a flowchart showing a flow of data processing according to the first embodiment.
- FIG. 6 is a block diagram illustrating an internal configuration example of an audio apparatus according to Embodiment 2.
- Embodiment 1 First, a calculation model on the assumption that the virtual sound source does not move in the acoustic space provided by WFS and a calculation model considering the movement of the virtual sound source will be described, and then the description of the embodiment will proceed.
- FIG. 1 is an explanatory diagram of an example of an acoustic space provided by WFS.
- a speaker array 103 composed of M speakers 103_1 to 103_M and a listener 102 who is listening to the sound while facing the speaker array 103.
- the wavefronts of the sound radiated from the M speakers 103_1 to 103_M are subjected to wavefront synthesis based on Huygens' principle, and are transmitted through the acoustic space as a synthesized wavefront 104.
- the listener 102 feels as if the sound actually radiated from the speaker array 103 is radiated from N virtual sound sources 101_1 to 101_N which are located behind the speaker array 103 and do not actually exist. Receive a good feeling.
- the N virtual sound sources 101_1 to 101_N are collectively referred to as a virtual sound source 101.
- FIG. 2 is an explanatory diagram for generally explaining an audio signal.
- the audio signal is generally expressed as a continuous signal S (t).
- 2A shows a continuous signal S (t)
- FIG. 2B shows an impulse train at a sampling interval ⁇ t
- the continuous signal S (t) is continuous on both the time t axis and the amplitude S axis.
- Sampling aims to obtain a temporally discrete signal from the continuous signal S (t).
- sampling and quantization operation is performed by punching out the continuous signal S (t) with an impulse train (FIG. 2B) of the sampling interval ⁇ t and quantizing them, as shown in FIG. 2C. Is called.
- the quantized data s (b ⁇ t) is referred to as “sample data”.
- sample data at a discrete time t of an audio signal to be given to an m-th speaker (hereinafter referred to as “speaker 103_m”) included in the speaker array 103 is generated.
- the number of virtual sound sources 101 is N and the number of speakers constituting the speaker array 103 is M.
- q n (t) Sample data at discrete times t of sound waves emitted from the nth virtual sound source (hereinafter referred to as “virtual sound source 101_n”) of the N virtual sound sources 101 and reaching the speaker 103_m l m (t): sample data at discrete time t of the audio signal applied to the speaker 103_m
- G n gain coefficient for the virtual sound source 101_n s n (t): sample data of the audio signal applied to the virtual sound source 101_n at the discrete time t ⁇ mn : sound wave resulting from the distance between the position of the virtual sound source 101_n and the position of the speaker 103_m Number of samples for propagation time
- the floor symbol indicates “the largest of the integers not exceeding the given value”.
- the gain coefficient G n for the virtual sound source 101_n is inversely proportional to the square root of the distance from the virtual sound source 101_n to the speaker 103_m. This is because a set of speakers 103_m is modeled as a line sound source.
- the sound wave propagation time ⁇ mn is proportional to the distance from the virtual sound source 101 — n to the speaker 103 — m.
- Equations (1) to (4) are based on the assumption that the virtual sound source 101_n does not move and is still at a certain position.
- the sound source may be stationary or may move. Therefore, in order to deal with such a case, a new calculation model (calculation model according to Embodiment 1) that considers the case where the sound source moves is introduced.
- a new calculation model will be described.
- G n, t Gain coefficient with respect to virtual sound source 101 — n at discrete time t ⁇ mn
- t Number of samples of sound wave propagation time due to distance between virtual sound source 101 — n and speaker 103 — m at discrete time t
- the gain coefficient for the virtual sound source 101_n, the position of the virtual sound source 101_n, and the sound wave propagation time all vary according to the discrete time t. .
- audio data is signal-processed in segment units.
- a “segment” is a unit of processing audio data and is also called a “frame”.
- One segment is composed of, for example, 256 sample data or 512 sample data. Therefore, l m (t) (sample data at the discrete time t of the audio signal applied to the speaker 103 — m ) in Expression (1) is calculated in segment units. Therefore, in this calculation model, a segment of audio data forming an audio signal to be given to the speaker 103 — m calculated at the discrete time t is set as a vector, and L m, t .
- L m, t is vector data composed of a sample data (for example, 256, 512, etc.) included in one segment from the discrete time t ⁇ a + 1 to the discrete time t. And is expressed by Equation (8).
- G n, t and ⁇ mn, t also vary according to the distance the virtual sound source 101_n has moved from the discrete time (t 0 -a) to the discrete time t 0 .
- Equations (9) and (10) shown below represent the amount of gain coefficient variation and the sound wave propagation time that vary according to the distance the virtual sound source 101_n has moved from the discrete time (t 0 -a) to the discrete time t 0. Represents the amount of change in the number of samples per minute.
- .DELTA.G n t0 represents the amount of change of the gain coefficient at discrete time t 0, ⁇ mn, t0 is the number of samples of the wave propagation time duration at discrete time t 0, at the discrete time (t 0 -a) It represents the amount of fluctuation (also called “time width”) from the number of samples for the sound wave propagation time.
- These fluctuation amounts take either a positive value or a negative value depending on the direction in which the virtual sound source 101_n moves when the virtual sound source moves from the discrete time (t 0 -a) to the discrete time t 0. .
- ⁇ G n, t0 and time width ⁇ mn, t0 are generated when the virtual sound source 101_n moves in a direction away from or near the speaker 103_m, a waveform distortion occurs at the discrete time t 0 .
- the state in which the “waveform distortion” has occurred means a state in which the audio signal waveform does not change continuously but changes so discontinuously that the listener perceives that portion as noise.
- the first segment of the segment starting from the discrete time t 0 is used.
- the audio data of the last part in the previous segment appears again by the time width ⁇ mn, t0 .
- the segment immediately preceding segment which starts discrete time t 0 is referred to as a first segment, called a segment which starts discrete time t 0 and the second segment.
- FIG. 3 is an explanatory diagram of a part of an audio signal waveform formed by audio data.
- the audio data shown in FIG. 3 is represented by a total of 28 pieces of sample data from sample data 301 to sample data 328.
- the waveform distortion occurs when the virtual sound source 101_n moves in the direction away from the speaker 103_m and in the approaching direction based on the audio signal illustrated in FIG. 3 will be described.
- FIG. 4 is an explanatory diagram of an example of an audio signal waveform formed by the audio data in the first segment.
- Sample data 301 to 312 are included in the last part of the first segment.
- FIG. 5 is an explanatory diagram of an example of an audio signal waveform formed by the audio data in the second segment.
- the first portion of the second segment includes sample data 308 'to 318.
- the number of samples corresponding to the sound wave propagation time with respect to the distance from the virtual sound source 101_n to the speaker 103_m in the second segment becomes the virtual sound source 101_n in the first segment.
- sample data 308, 309, 310, 311 and 312 in the first segment shown in FIG. 4 is transferred to the first portion in the second segment shown in FIG. Reappear as sample data 308 ′, 309 ′, 310 ′, 311 ′, 312 ′. Therefore, when the audio signal waveform formed by the audio data shown in FIG. 4 and the audio signal waveform formed by the audio data shown in FIG. 5 are combined, waveform distortion occurs in the combined portion.
- FIG. 6 is an explanatory diagram of an example of an audio signal waveform formed by combining the audio signal waveform formed by the audio data shown in FIG. 4 and the audio signal waveform formed by the audio data shown in FIG. From FIG. 6, it can be seen that the audio data is discontinuous in the vicinity of the sample data 308 ′ and waveform distortion occurs. This waveform distortion is perceived by the listener as noise.
- FIG. 7 is an explanatory diagram of an example of an audio signal waveform formed by the audio data in the first segment.
- Sample data 301 to 312 are included in the last part of the first segment.
- the contents are the same as those shown in FIG.
- FIG. 8 is an explanatory diagram of an example of an audio signal waveform formed by audio data in the second segment.
- Sample data 317 to 328 are included in the first portion of the second segment.
- FIG. 9 shows four points between the audio signal waveform formed by the audio data of the first part in the first segment and the audio signal waveform formed by the audio data of the last part in the second segment. It is explanatory drawing which shows the state in which the missing part has generate
- the audio signal waveform formed by the audio data of the last part in the first segment and the audio data of the first part in the second segment are formed.
- Four missing points are generated between the audio signal waveforms. Therefore, when the audio signal waveform formed by the audio data shown in FIG. 7 and the audio signal waveform formed by the audio data shown in FIG. 8 are combined, waveform distortion occurs in the combined portion.
- FIG. 10 is an explanatory diagram of an example of an audio signal waveform obtained by combining the audio signal waveform formed by the audio data shown in FIG. 7 and the audio signal waveform formed by the audio data shown in FIG.
- the audio data is discontinuous in the vicinity of the sample data 317, and waveform distortion occurs. This waveform distortion is also perceived by the listener as noise.
- FIG. 11 is a block diagram illustrating a configuration example of an audio apparatus including the audio data processing unit according to the first embodiment.
- the audio device 1100 includes an audio data processing unit 1101, a content information separation unit 1102, an audio data storage unit 1103, a virtual sound source position data storage unit 1104, a speaker position data input unit 1105, and a speaker position data storage unit 1106 according to Embodiment 1.
- the audio apparatus 1100 includes a CPU (Central Processing Unit) 1111 that controls the above-described units centrally, a ROM (Read-Only Memory) 1112 that stores a computer program executed by the CPU 1111, and data and variables that are processed during the execution of the computer program. Further, a RAM (Random-Access Memory) 1113 is stored. The audio device 1100 outputs an audio signal corresponding to the corrected audio data to the speaker array 103.
- a CPU Central Processing Unit
- ROM Read-Only Memory
- RAM Random-Access Memory
- the playback unit 1109 reads the digital content from the recording medium 1117 that stores the digital content (movie, computer game, music video, etc.), and outputs the digital content to the content information separation unit 1102.
- the recording medium 1117 is, for example, a CD-R (Compact Disc Recordable), a DVD (Digital Versatile Disk), or a Blu-ray Disc (registered trademark).
- a plurality of audio data files corresponding to each of the virtual sound sources 101_1 to 101_N and virtual sound source position data corresponding to the virtual sound sources 101_1 to 101_N are recorded in association with each other.
- the communication interface unit 1110 acquires the digital content from the server 1115 that distributes the digital content via a communication network such as the Internet 1114 and outputs the digital content to the content information separation unit 1102.
- the communication interface unit 1110 includes a device (not shown) such as an antenna or a tuner, receives a program broadcast by the broadcast station 1116, and outputs it as a digital content to the content information separation unit 1102.
- the content information separation unit 1102 acquires digital content from the playback unit 1109 or the communication interface unit 1110, analyzes the digital content, and separates audio data and virtual sound source position data from the digital content. Next, the content information separation unit 1102 outputs the separated audio data and virtual sound source position data to the audio data storage unit 1103 and the virtual sound source position data storage unit 1104, respectively.
- the virtual sound source position data is, for example, position data corresponding to the relative positions of a singer and a plurality of musical instruments displayed on the video screen when the digital content is a music video.
- the virtual sound source position data is stored in the digital content together with the audio data.
- the audio data storage unit 1103 stores audio data acquired from the content information separation unit 1102, and the virtual sound source position data storage unit 1104 stores virtual sound source position data acquired from the content information separation unit 1102.
- the speaker position data storage unit 1106 acquires speaker position data indicating the position in the acoustic space where the speakers 103_1 to 103_M of the speaker array 103 are arranged from the speaker position data input unit 1105, and stores the speaker position data.
- the speaker position data is information set by the user based on the positions of the speakers 103_1 to 103_M constituting the speaker array 103.
- the information is represented by, for example, coordinates in one plane (XY coordinate system) fixed to the audio device 1100 in the acoustic space.
- the user operates the speaker position data input unit 1105 to store the speaker position data in the speaker position data storage unit 1106. If the arrangement of the speaker array 103 is determined in advance due to mounting restrictions, the speaker position data is set as a fixed value. On the other hand, when the user can freely determine the arrangement of the speaker array 103 to some extent, the speaker position data is set as a variable value.
- the audio data processing unit 1101 reads the audio file corresponding to each of the virtual sound sources 101_1 to 101_N from the audio data storage unit 1103. Also, the audio data processing unit 1101 reads virtual sound source position data corresponding to the virtual sound sources 101_1 to 101_N from the virtual sound source position data storage unit 1104. Furthermore, the audio data processing unit 1101 reads speaker position data corresponding to the speakers 103_1 to 103_M of the speaker array 103 from the speaker position data storage unit 1106. The audio data processing unit 1101 performs processing according to the embodiment on the read audio data based on the read virtual sound source position data and speaker position data.
- the audio data processing unit 1101 generates audio data that forms an audio signal to be given to the speakers 103_1 to 103_M by performing arithmetic processing based on the above-described arithmetic model in consideration of movement of the virtual sound sources 101_1 to 101_N.
- the audio data generated by the audio data processing unit 1101 is output as an audio signal by the D / A conversion unit 1107, and is output to the speakers 103_1 to 103_M via the amplification units 1108_1 to 1108_M.
- the speaker array 103 generates sound based on the audio signal and radiates it into the acoustic space.
- FIG. 12 is a block diagram illustrating an internal configuration example of the audio data processing unit 1101 according to the first embodiment.
- the audio data processing unit 1101 includes a distance data calculation unit 1201, a sound wave propagation time data calculation unit 1202, a sound wave propagation time data buffer 1203, a gain coefficient data calculation unit 1204, a gain coefficient data buffer 1205, an input audio data buffer 1206, and output audio data.
- a generation unit 1207, an output audio data superimposing unit 1208, and an output audio data buffer 1209 are provided.
- the distance data calculation unit 1201 is connected to the virtual sound source position data storage unit 1104 and the speaker position data storage unit 1106.
- the input audio data buffer 1206 is connected to the audio data storage unit 1103.
- the output audio data superimposing unit 1208 is connected to the D / A conversion unit 1107.
- the output audio data buffer 1209 is connected to the output audio data generation unit 1207.
- the distance data calculation unit 1201 acquires the virtual sound source position data and the speaker position data from the virtual sound source position data storage unit 1104 and the speaker position data storage unit 1106, and based on them, between the virtual sound source 101_n and each of the speakers 103_1 to 103_M. Distance data (
- the sound wave propagation time data calculation unit 1202 is based on the distance data (
- the sound wave propagation time data buffer 1203 acquires the sound wave propagation time data ⁇ mn, t from the sound wave propagation time data calculation unit 1202 and temporarily stores sound wave propagation time data for a plurality of segments.
- the gain coefficient data calculation unit 1204 calculates gain coefficient data G n, t based on the distance data (
- the input audio data buffer 1206 obtains input audio data corresponding to each virtual sound source 101_n from the audio data storage unit 1103, and temporarily stores input audio data for a plurality of segments therein.
- One segment is composed of, for example, sample data of 256 or 512 audio data.
- the output audio data generation unit 1207 uses the sound wave propagation time data ⁇ mn, t calculated by the sound wave propagation time data calculation unit 1203 and the gain coefficient data G n, t calculated by the gain coefficient data calculation unit 1205 to input audio data.
- Output audio data corresponding to the input audio data temporarily stored in the buffer 1206 is generated.
- the output audio data superimposing unit 1208 synthesizes the output audio data generated by the output audio data generating unit 1207 according to the number of virtual sound sources 101_n.
- FIG. 13 is an explanatory diagram of a configuration example of the input audio data buffer 1206.
- the input audio data buffer 1206 temporarily stores data using a FIFO (First-In, First-Out) method, and discards old data.
- the input audio data buffer 1206 reads the input audio data from the audio data storage unit 1103 according to its own buffer size, and outputs it to the output audio data generation unit 1207 after storage.
- each square block represents a sample data storage area, and one sample data in the segment is temporarily stored in the sample data storage area.
- one sample data of the head part of the latest segment is temporarily stored in the sample data storage area 1300_1, and one of the last part of the latest segment is stored in the sample data storage area 1300_1 + a-1.
- Sample data, that is, the latest one sample data is temporarily stored.
- a is the segment length, which is the number of sample data included in one segment.
- FIG. 14 is an explanatory diagram of a configuration example of the sound wave propagation time data buffer 1203.
- the sound wave propagation time data buffer 1203 is also a temporary storage unit that inputs and outputs data using the FIFO method.
- each square block represents a sound wave propagation time data storage area, and the sound wave propagation time data of each segment is temporarily stored in the sound wave propagation time data storage area.
- FIG. 14 shows that the sound wave propagation time data for two segments is temporarily stored in the sound wave propagation time data buffer 1203. Further, FIG.
- the oldest sound wave propagation time data is temporarily stored in the sound wave propagation time data storage area 1203_1 of the sound wave propagation time data buffer 1203, and the newest sound wave propagation time data is stored in the sound wave propagation time data storage area 1203_2. Is stored temporarily.
- the input audio data buffer 1206 reads one segment of input audio data from the discrete time t 1 to the discrete time (t 1 + a ⁇ 1) from the audio data storage unit 1103 and temporarily stores it.
- sample data from the discrete time t 1 to the discrete time (t 1 + a-1) are stored in order.
- input audio data for a plurality of segments before the discrete time t 1 is already stored in the sample data storage areas other than the sample data storage areas 1300_1 to 1300_1 + a-1.
- the output audio data buffer 1209 already stores sample data of the output audio data corresponding to the previous segment at the discrete time (t 1 ⁇ 1).
- the sound wave propagation time data buffer 1203 already stores sound wave propagation time data of the previous segment.
- the distance data calculation unit 1201 indicates the distance between the first virtual sound source (hereinafter referred to as “virtual sound source 101_1”) and the first speaker (hereinafter referred to as “speaker 103_1”) at the discrete time t 1 .
- Distance data (
- the sound wave propagation time data calculation unit 1202 calculates the sound wave propagation time data ⁇ 11, t 1 based on the distance data (
- the sound wave propagation time data buffer 1203 stores the sound wave propagation time data ⁇ 11, t 1 acquired from the sound wave propagation time data calculation unit 1202. Referring to FIG. 14, after the data already stored in the data storage area 1203_2 is moved to 1203_1, the sound wave propagation time data ⁇ 11, t1 is stored in the data storage area 1203_2. Therefore, at this time, the sound wave propagation time data buffer 1203_1 stores the sound wave propagation time data of the previous segment. Note that as many sound wave propagation time data buffers as the number of speakers ⁇ the number of virtual sound sources existing at time t 1 are prepared. That is, at least M ⁇ N sound wave propagation time data buffers are provided, each storing sound wave propagation time data and current sound wave propagation time data for the past one segment.
- the gain coefficient data calculation unit 1204 calculates gain coefficient data G 1, t1 based on the distance data (
- the output audio data generation unit 1207 generates output audio data using the newer sound wave propagation time data stored in the sound wave propagation time data buffer 1203 and the gain coefficient data calculated by the gain coefficient data calculation unit 1204.
- the waveform distortion shown in FIG. 6 occurs. Street. That is, as shown in the equation (7), the sound wave propagation time data ⁇ mn, t1 is larger than the sound wave propagation time data ⁇ mn, t1- a, and therefore, in the segment starting from the discrete time t 1 .
- the first part is a repetition of the last part in the segment starting from the discrete time (t 1 -a).
- the first part in the segment starting from the discrete time t 1 and the last part in the segment starting from the discrete time (t 1 -a) are the time width ⁇ mn that is the difference in the sound propagation time data.
- the waveform of the audio data becomes discontinuous in the vicinity of the discrete time t 1 .
- the time width ⁇ mn, t1 of the sound wave propagation time data is set to 5.
- FIG. 6 is an explanatory diagram of an example of a waveform before correction.
- the waveform before correction from the discrete time t 1 to the discrete time (t 1 + ⁇ mn, t1 ) is a waveform obtained by connecting the sample data 308 ′, 309 ′, 310 ′, 311 ′, and 312 ′.
- This waveform is the same as the waveform obtained by connecting the sample data 308, 309, 310, 311 and 312 in the previous segment.
- the correction section width is set to 5 similarly to the time width ⁇ mn, t1 .
- the output audio data buffer 1209 already stores the sample data 312 at the last discrete time (t 1 ⁇ 1) of the previous segment.
- the sample data 312 (see FIG. 6) at the discrete time (t 1 ⁇ 1), that is, the sample data stored in the output audio data buffer 1209 is used.
- linear interpolation is used as an example.
- FIG. 15 is an explanatory diagram of an example of an audio signal waveform formed from the corrected audio data. From FIG. 15, in the corrected audio signal waveform, sample data 312 to sample data 313 are linearized by linear interpolation (sample data 1500 to sample data 1504), thereby eliminating the waveform distortion shown in FIG. You can see that
- the sound wave propagation time of the segment starting from the discrete time (t 1 -a) and the sound wave propagation time of the segment starting from the discrete time t 1 are calculated. It only has to be done. That is, in order to correct the distortion in the audio data in the vicinity of the starting point of the current segment, the sound wave propagation time of the audio data of the segment starting from the discrete time (t 1 + a) that is the next segment is calculated. There is no need to keep it. Therefore, when the virtual sound source 101_n is separated from the speaker 103_m, there is no delay for one segment. Therefore, even when the virtual sound source position is changed in real time, the audio data can be corrected without delay.
- FIG. 10 is an explanatory diagram of an example of an audio signal waveform obtained by combining the audio signal waveform formed by the audio data shown in FIG. 7 and the audio signal waveform formed by the audio data shown in FIG.
- the audio data changes abruptly in the vicinity of the sample data 317, resulting in waveform distortion. This waveform distortion is also perceived by the listener as noise.
- the output audio data buffer 1209 stores sample data 312 at the last discrete time (t 1 ⁇ 1) of the previous segment.
- linear interpolation is used as an example. Therefore, in FIG. 10, it is considered that the sample data 312 to the sample data 321 are linear.
- FIG. 16 is an explanatory diagram of an example of an audio signal waveform formed by the corrected audio data. From FIG.
- the sample data 312 to the sample data 321 are linearized by linear interpolation (sample data 1600 to sample data 1603), thereby eliminating the waveform distortion shown in FIG.
- the sound wave propagation time of the segment starting from the discrete time (t 1 -a) The sound wave propagation time of the segment starting from the discrete time t 1 may be calculated.
- the sound wave propagation time of the audio data of the segment starting from the discrete time (t 1 + a) that is the next segment is calculated. There is no need to keep it. Therefore, when the virtual sound source 101_n is separated from the speaker 103_m, there is no delay for one segment. Therefore, even when the virtual sound source position is changed in real time, the audio data can be corrected without delay.
- FIG. 17 is a flowchart showing a data processing flow according to the first embodiment.
- This data processing is executed by the audio data processing unit 1101 under the control of the CPU 1111.
- the audio data processing unit 1101 first substitutes 1 for the number n of the virtual sound source 101_n and 1 for the number m of the speaker 103_m. That is, the audio data processing unit 1101 designates the first virtual sound source 101_1 and the first speaker 103_1 (S10).
- the audio data processing unit 1101 inputs an audio file corresponding to the nth virtual sound source 101_n from the audio data storage unit 1103 (S11).
- the audio data processing unit 1101 inputs the virtual sound source position data and the speaker position data corresponding to the virtual sound source 101_n from the virtual sound source position data storage unit 1104 and the speaker position data storage unit 1106 (S12). Based on the input virtual sound source position data and speaker position data, the audio data processing unit 1101 first and second distance data (
- the audio data processing unit 1101 stores the sound wave propagation time data ⁇ mn, t and the gain coefficient data G n, t in the sound wave propagation time data buffer 1203 and the gain coefficient data buffer 1205, respectively. Next, the audio data processing unit 1101 determines whether or not the first and second distance data are different (S15). Even if it is determined whether or not the sound wave propagation time ⁇ mn, ta corresponding to the previous segment stored in the sound wave propagation time data buffer 1203 is different from the sound wave propagation time data ⁇ mn, t stored this time. Good. That is, in this step, the audio data processing unit 1101 determines whether the virtual sound source 101_n is moving or stationary with respect to the speaker 103_m.
- step S15 If it is determined in step S15 that the first and second distance data are different (S15: YES), that is, if it is determined that the virtual sound source 101_n has moved relative to the speaker 103_m, the audio data processing unit 1101 performs step The process proceeds to S16. On the other hand, if it is determined in step S15 that the first and second distance data are the same (S15: NO), that is, if it is determined that the virtual sound source 101_n is stationary, the audio data processing unit 1101 The process proceeds to step S19. Based on the determination result of step S15, the audio data processing unit 1101 identifies the repeated portion and missing portion of the sample data due to the separation and approach of the virtual sound source to the speaker (S16), and the waveform distortion portion is described above. The waveform is corrected by performing the linear interpolation (S17).
- the audio data processing unit 1101 performs gain control on the virtual sound source 101_n (S18).
- the audio data processing unit 1101 adds 1 to the number n of the virtual sound source 101_n (S19), and determines whether the number n of the virtual sound source 101_n is the maximum value N (S20). If it is determined in step S20 that the number n of the virtual sound source 101_n is the maximum value N (S20: YES), audio data is synthesized (S21).
- step S20 when it is determined that the number of the virtual sound source 101_n is not the maximum value N (S20: NO), the audio data processing unit 1101 returns to the process of step S11, and then continues to the second virtual sound source 101_n.
- the processing from step S11 to step S18 is performed on the sound source 101_2 and the first speaker 103_1.
- the audio data processing unit 1101 After synthesizing the audio data in step S21, the audio data processing unit 1101 substitutes 1 for the number n of the virtual sound source 101_n (S22), and adds 1 to the number m of the speaker 103_m (S23). Next, the audio data processing unit 1101 determines whether or not the number m of the speaker 103_m is the maximum value M (S24), and determines that the number m of the speaker 103_m is the maximum value M (S24: YES). Exit. On the other hand, when it is determined that the number m of the speaker 103_m is not the maximum value M (S24: NO), the process returns to step S11.
- FIG. 18 is a block diagram illustrating an internal configuration example of the audio apparatus 1100 according to the second embodiment.
- the program stored in the ROM 1112 in the audio device 1100 is executed in the first embodiment, whereas the rewritable EEPROM (Electrically Erasable Programmable Read-Only Memory) or the internal storage device 25 is used.
- the stored program is read out and executed.
- the audio device 1100 includes an EEPROM 24, an internal storage device 25, and a recording medium reading unit 23.
- the CPU 17 reads the program 231 from a recording medium 230 such as a CD (Compact Disk) -ROM or a DVD (Digital Versatile Disk) -ROM inserted in the recording medium reading unit 23 and stores it in the EEPROM 24 or the internal storage device 25. It is.
- the CPU 17 is configured to read the program 231 stored in the EEPROM 24 or the internal storage device 25 to the RAM 18 and execute it.
- the program 231 is not limited to being read from the recording medium 230 and stored in the EEPROM 24 or the internal storage device 25, but may be stored in an external memory such as a memory card. In this case, the program 231 is read from an external memory (not shown) connected to the CPU 17 and stored in the EEPROM 24 or the internal storage device 25. Further, communication may be established between a communication unit (not shown) connected to the CPU 17 and an external computer, and the program 231 may be downloaded to the EEPROM 24 or the internal storage device 25.
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Abstract
Description
まず始めに、WFSにより提供される音響空間内で仮想音源が移動しないことを前提とした演算モデル及び仮想音源の移動を考慮した演算モデルについて説明し、次いで、実施の形態の説明に移る。
qn (t):N個の仮想音源101の内のn番目の仮想音源(以下、「仮想音源101_n」と呼ぶ。)から放射されて、スピーカ103_mに到達した音波の離散時刻tにおける標本データ
lm (t):スピーカ103_mに与えるオーディオ信号の離散時刻tにおける標本データ
Gn :仮想音源101_nに対する利得係数
sn (t):仮想音源101_nに与えるオーディオ信号の離散時刻tにおける標本データ
τmn:仮想音源101_nの位置及びスピーカ103_mの位置の間の距離に起因する音波伝播時間の分のサンプル数
w:重み定数
rn :仮想音源101_nの位置ベクトル(固定値)
rm :スピーカ103_mの位置ベクトル(固定値)
Gn,t :離散時刻tにおける仮想音源101_nに対する利得係数
τmn,t:離散時刻tにおける仮想音源101_n及びスピーカ103_mの間の距離に起因する音波伝播時間の分のサンプル数
rn,t :離散時刻tにおける仮想音源101_nの位置ベクトル
図18は、実施の形態2に係るオーディオ装置1100の内部構成例を示すブロック図である。実施の形態2は、実施の形態1がオーディオ装置1100内のROM1112に記憶してあるプログラムを実行するのに対して、書き換え可能なEEPROM( Electrically Erasable Programmable Read-Only Memory )又は内部記憶装置25に記憶されたプログラムを読み出して実行するようにしてある。オーディオ装置1100は、EEPROM24、内部記憶装置25及び記録媒体読込部23を備える。CPU17は、記録媒体読込部23に挿入されたCD( Compact Disk )-ROM又はDVD(Digital Versatile Disk )-ROM等の記録媒体230からプログラム231を読み込んでEEPROM24又は内部記憶装置25に記憶するようにしてある。CPU17は、EEPROM24又は内部記憶装置25に記憶したプログラム231をRAM18に読み出して実行する構成となっている。
1100 オーディオ装置
1101 オーディオデータ処理部
1102 コンテンツ情報分離部
1109 再生部
1110 通信インタフェース部
1115 サーバ
1116 放送局
Claims (9)
- 移動する仮想音源が発する音に対応するオーディオデータ、該仮想音源の位置及び前記オーディオデータに基づき音を放射するスピーカの位置を入力し、前記仮想音源の位置及び前記スピーカの位置に基づいて前記オーディオデータを補正するオーディオデータ処理装置において、
相前後する時点での前記スピーカの位置から前記仮想音源の位置までそれぞれの第1の距離及び第2の距離を算出する算出手段と、
前記第1の距離及び第2の距離が異なる場合、前後の時点における前記オーディオデータにある歪みの部分を特定する特定手段と、
前記特定された部分の前記オーディオデータを、関数を用いた補間によって補正する補正手段と
を備えるオーディオデータ処理装置。 - 前記オーディオデータは標本データを含み、
前記特定手段は、前記仮想音源の前記スピーカに対する離隔及び接近に起因する標本データの繰り返し部分及び欠落部分を特定し、
前記補正手段は、特定された前記繰り返し部分及び欠落部分を、関数を用いた補間によって補正する請求項1記載のオーディオデータ処理装置。 - 前記関数を用いた補間は、線形補間である請求項1又は2に記載のオーディオデータ処理装置。
- 前記補正を行う部分は、前記第1及び第2の距離を音波が伝播する時間幅の差、又は、前記差に比例する時間幅である請求項1から3までのいずれか1項に記載のオーディオデータ処理装置。
- 移動する仮想音源が発する音に対応するオーディオデータ、該仮想音源の位置及び前記オーディオデータに基づき音を放射するスピーカの位置を用い、前記仮想音源の位置及び前記スピーカの位置に基づいて前記オーディオデータを補正するオーディオ装置において、
前記オーディオデータ及び前記仮想音源の位置を含むディジタルコンテンツを入力するディジタルコンテンツ入力部と、
前記ディジタルコンテンツ入力部が入力したディジタルコンテンツを解析し、該ディジタルコンテンツに含まれるオーディオデータ及び仮想音源の位置のデータを分離するコンテンツ情報分離部と、
前記コンテンツ情報分離部が分離した仮想音源の位置のデータ及び前記スピーカの位置のデータに基づいて、前記コンテンツ情報分離部が分離したオーディデータを補正するオーディオデータ処理部と、
補正後のオーディオデータをオーディオ信号に変換してスピーカへ出力するオーディオ信号生成部と
を備え、
前記オーディオデータ処理部は、
相前後する時点での前記スピーカの位置から前記仮想音源の位置までそれぞれの第1の距離及び第2の距離を算出する算出手段と、
前記第1の距離及び第2の距離が異なる場合、前後の時点における前記オーディオデータにある歪みの部分を特定する特定手段と、
前記特定された部分の前記オーディオデータを、関数を用いた補間によって補正する補正手段と
を備えるオーディオ装置。 - 前記ディジタルコンテンツ入力部は、ディジタルコンテンツを格納する記録媒体、ネットワークを介してディジタルコンテンツを配信するサーバ又はディジタルコンテンツを放送する放送局からディジタルコンテンツを入力する請求項5に記載のオーディオ装置。
- 移動する仮想音源が発する音に対応するオーディオデータ、該仮想音源の位置及びオーディオデータに基づき音を放射するスピーカの位置を入力し、前記仮想音源の位置及び前記スピーカの位置に基づいて前記オーディオデータを補正するオーディオデータ処理装置におけるオーディオデータ処理方法において、
相前後する時点での前記スピーカの位置から前記仮想音源の位置までそれぞれの第1の距離及び第2の距離を算出するステップと、
前記第1の距離及び第2の距離が異なる場合、前後の時点における前記オーディオデータにある歪みの部分を特定するステップと、
前記特定された部分の前記オーディオデータを、関数を用いた補間によって補正するステップと
を含むオーディオデータ処理方法。 - オーディオデータに対応するオーディオ信号を入力するスピーカが放射する音によって形成される仮想音源の位置及び該スピーカの位置に基づいて、移動する音源が発する音に対応する前記オーディオデータを補正させるプログラムにおいて、
コンピュータに、
相前後する時点での前記スピーカの位置から前記仮想音源の位置までそれぞれの第1の距離及び第2の距離を算出するステップと、
前記第1の距離及び第2の距離が異なる場合、前後の時点における前記オーディオデータにある歪みの部分を特定するステップと、
前記特定された部分の前記オーディオデータを、関数を用いた補間によって補正するステップと
を実行させるプログラム。 - 請求項8に記載のプログラムを記録したコンピュータ読み取り可能な記録媒体。
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