This disclosure relates to processing of audio data.
Computer-mediated reality systems are being developed to allow computing devices to augment or add to, remove or subtract from, or generally modify existing reality experienced by a user. Computer-mediated reality systems (which may also be referred to as “extended reality systems,” or “XR systems”) may include, as examples, virtual reality (VR) systems, augmented reality (AR) systems, and mixed reality (MR) systems. The perceived success of computer-mediated reality systems are generally related to the ability of such computer-mediated reality systems to provide a realistically immersive experience in terms of both the video and audio experience where the video and audio experience align in ways expected by the user. Although the human visual system is more sensitive than the human auditory systems (e.g., in terms of perceived localization of various objects within the scene), ensuring an adequate auditory experience is an increasingly import factor in ensuring a realistically immersive experience, particularly as the video experience improves to permit better localization of video objects that enable the user to better identify sources of audio content.
This disclosure generally relates to techniques for interpolating an audio stream from one or more existing audio streams. The techniques may improve the listener experience, while also reducing soundfield reproduction localization errors, as the interpolated audio stream may better reflect a location of a listener relative to the existing audio streams, thereby improving the operation of a playback device (that performs the techniques to reproduce the soundfield) itself.
In one example, the techniques are directed to a device configured to process one or more audio streams, the device comprising: a memory configured to store the one or more audio streams; and a processor coupled to the memory, and configured to: obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtain a listener location identifying a location of a listener; perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtain, based on the interpolated audio stream, one or more speaker feeds; and output the one or more speaker feeds.
In another example, the techniques are directed to a method for processing one or more audio streams, the method comprising: obtaining one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtaining a listener location identifying a location of a listener; performing interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtaining, based on the interpolated audio stream, one or more speaker feeds; and outputting the one or more speaker feeds.
In another example, the techniques are directed to a device configured to process one or more audio streams, the device comprising: means for obtaining one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; means for obtaining a listener location identifying a location of a listener; means for performing interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; means for obtaining, based on the interpolated audio stream, one or more speaker feeds; and means for outputting the one or more speaker feeds.
In another example, the techniques are directed to a non-transitory computer-readable storage medium having stored thereon instructions that, when executed, cause one or more processors to: obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtain a listener location identifying a location of a listener; perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtain, based on the interpolated audio stream, one or more speaker feeds; and output the one or more speaker feeds.
The details of one or more examples of this disclosure are set forth in the accompanying drawings and the description below. Other features, objects, and advantages of various aspects of the techniques will be apparent from the description and drawings, and from the claims.
There are a number of different ways to represent a soundfield. Example formats include channel-based audio formats, object-based audio formats, and scene-based audio formats. Channel-based audio formats refer to the 5.1 surround sound format, 7.1 surround sound formats, 22.2 surround sound formats, or any other channel-based format that localizes audio channels to particular locations around the listener in order to recreate a soundfield.
Object-based audio formats may refer to formats in which audio objects, often encoded using pulse-code modulation (PCM) and referred to as PCM audio objects, are specified in order to represent the soundfield. Such audio objects may include metadata identifying a location of the audio object relative to a listener or other point of reference in the soundfield, such that the audio object may be rendered to one or more speaker channels for playback in an effort to recreate the soundfield. The techniques described in this disclosure may apply to any of the foregoing formats, including scene-based audio formats, channel-based audio formats, object-based audio formats, or any combination thereof.
Scene-based audio formats may include a hierarchical set of elements that define the soundfield in three dimensions. One example of a hierarchical set of elements is a set of spherical harmonic coefficients (SHC). The following expression demonstrates a description or representation of a soundfield using SHC:
The expression shows that the pressure pi at any point {rr, θr, φr} of the soundfield, at time t, can be represented uniquely by the SHC, Anm(k). Here,
c is the speed of sound (˜343 m/s), {rr, θr, φr,} is a point of reference (or observation point), jn(·) is the spherical Bessel function of order n, and YNm(θr,φr) are the spherical harmonic basis functions (which may also be referred to as a spherical basis function) of order n and suborder m. It can be recognized that the term in square brackets is a frequency-domain representation of the signal (i.e., S(ω, rr, θr, φr)) which can be approximated by various time-frequency transformations, such as the discrete Fourier transform (DFT), the discrete cosine transform (DCT), or a wavelet transform. Other examples of hierarchical sets include sets of wavelet transform coefficients and other sets of coefficients of multiresolution basis functions.
The SHC Anm(k) can either be physically acquired (e.g., recorded) by various microphone array configurations or, alternatively, they can be derived from channel-based or object-based descriptions of the soundfield. The SHC (which also may be referred to as ambisonic coefficients) represent scene-based audio, where the SHC may be input to an audio encoder to obtain encoded SHC that may promote more efficient transmission or storage. For example, a fourth-order representation involving (1+4)2 (25, and hence fourth order) coefficients may be used.
As noted above, the SHC may be derived from a microphone recording using a microphone array. Various examples of how SHC may be physically acquired from microphone arrays are described in Poletti, M., “Three-Dimensional Surround Sound Systems Based on Spherical Harmonics,” J. Audio Eng. Soc., Vol. 53, No. 11, 2005 November, pp. 1004-1025.
The following equation may illustrate how the SHCs may be derived from an object-based description. The coefficients Amm(k) for the soundfield corresponding to an individual audio object may be expressed as:
A
n
m(k)=g(ω)(−4πik)hn(2)(krs)Ynm*(θs,φs),
where i is √{square root over (−1)}hn(2)(·) is the spherical Hankel function (of the second kind) of order n, and {rs, θs, φs} is the location of the object. Knowing the object source energy g (ω) as a function of frequency (e.g., using time-frequency analysis techniques, such as performing a fast Fourier transform on the pulse code modulated—PCM—stream) may enable conversion of each PCM object and the corresponding location into the SHC Anm(k). Further, it can be shown (since the above is a linear and orthogonal decomposition) that the Anm(k) coefficients for each object are additive. In this manner, a number of PCM objects can be represented by the Anm(k) coefficients (e.g., as a sum of the coefficient vectors for the individual objects). The coefficients may contain information about the soundfield (the pressure as a function of 3D coordinates), and the above represents the transformation from individual objects to a representation of the overall soundfield, in the vicinity of the observation point {rr, θr, φr}.
Computer-mediated reality systems (which may also be referred to as “extended reality systems,” or “XR systems”) are being developed to take advantage of many of the potential benefits provided by ambisonic coefficients. For example, ambisonic coefficients may represent a soundfield in three dimensions in a manner that potentially enables accurate three-dimensional (3D) localization of sound sources within the soundfield. As such, XR devices may render the ambisonic coefficients to speaker feeds that, when played via one or more speakers, accurately reproduce the soundfield.
The use of ambisonic coefficients for XR may enable development of a number of use cases that rely on the more immersive soundfields provided by the ambisonic coefficients, particularly for computer gaming applications and live video streaming applications. In these highly dynamic use cases that rely on low latency reproduction of the soundfield, the XR devices may prefer ambisonic coefficients over other representations that are more difficult to manipulate or involve complex rendering. More information regarding these use cases is provided below with respect to
While described in this disclosure with respect to the VR device, various aspects of the techniques may be performed in the context of other devices, such as a mobile device. In this instance, the mobile device (such as a so-called smartphone) may present the displayed world via a screen, which may be mounted to the head of the user 102 or viewed as would be done when normally using the mobile device. As such, any information on the screen can be part of the mobile device. The mobile device may be able to provide tracking information 41 and thereby allow for both a VR experience (when head mounted) and a normal experience to view the displayed world, where the normal experience may still allow the user to view the displayed world proving a VR-lite-type experience (e.g., holding up the device and rotating or translating the device to view different portions of the displayed world).
The source device 12 may be operated by an entertainment company or other entity that may generate multi-channel audio content for consumption by operators of content consumer devices, such as the content consumer device 14. In many VR scenarios, the source device 12 generates audio content in conjunction with video content. The source device 12 includes a content capture device 300 and a content soundfield representation generator 302.
The content capture device 300 may be configured to interface or otherwise communicate with one or more microphones 5A-5N (“microphones 5”). The microphones 5 may represent an Eigenmike® or other type of 3D audio microphone capable of capturing and representing the soundfield as corresponding scene-based audio data 11A-11N (which may also be referred to as ambisonic coefficients 11A-11N or “ambisonic coefficients 11”). In the context of scene-based audio data 11 (which is another way to refer to the ambisonic coefficients 11″), each of the microphones 5 may represent a cluster of microphones arranged within a single housing according to set geometries that facilitate generation of the ambisonic coefficients 11. As such, the term microphone may refer to a cluster of microphones (which are actually geometrically arranged transducers) or a single microphone (which may be referred to as a spot microphone).
The ambisonic coefficients 11 may represent one example of an audio stream. As such, the ambisonic coefficients 11 may also be referred to as audio streams 11. Although described primarily with respect to the ambisonic coefficients 11, the techniques may be performed with respect to other types of audio streams, including pulse code modulated (PCM) audio streams, channel-based audio streams, object-based audio streams, etc.
The content capture device 300 may, in some examples, include an integrated microphone that is integrated into the housing of the content capture device 300. The content capture device 300 may interface wirelessly or via a wired connection with the microphones 5. Rather than capture, or in conjunction with capturing, audio data via the microphones 5, the content capture device 300 may process the ambisonic coefficients 11 after the ambisonic coefficients 11 are input via some type of removable storage, wirelessly and/or via wired input processes. As such, various combinations of the content capture device 300 and the microphones 5 are possible.
The content capture device 300 may also be configured to interface or otherwise communicate with the soundfield representation generator 302. The soundfield representation generator 302 may include any type of hardware device capable of interfacing with the content capture device 300. The soundfield representation generator 302 may the use ambisonic coefficients 11 provided by the content capture device 300 to generate various representations of the same soundfield represented by the ambisonic coefficients 11.
For instance, to generate the different representations of the soundfield using ambisonic coefficients (which again is one example of the audio data 19), soundfield representation generator 24 may use a coding scheme for ambisonic representations of a soundfield, referred to as Mixed Order Ambisonics (MOA) as discussed in more detail in U.S. application Ser. No. 15/672,058, entitled “MIXED-ORDER AMBISONICS (MOA) AUDIO DATA FO COMPUTER-MEDIATED REALITY SYSTEMS,” filed Aug. 8, 2017, and published as U.S. patent publication no. 20190007781 on Jan. 3, 2019.
To generate a particular MOA representation of the soundfield, the soundfield representation generator 24 may generate a partial subset of the full set of ambisonic coefficients. For instance, each MOA representation generated by the soundfield representation generator 24 may provide precision with respect to some areas of the soundfield, but less precision in other areas. In one example, an MOA representation of the soundfield may include eight (8) uncompressed ambisonic coefficients, while the third order ambisonic representation of the same soundfield may include sixteen (16) uncompressed ambisonic coefficients. As such, each MOA representation of the soundfield that is generated as a partial subset of the ambisonic coefficients may be less storage-intensive and less bandwidth intensive (if and when transmitted as part of the bitstream 27 over the illustrated transmission channel) than the corresponding third order ambisonic representation of the same soundfield generated from the ambisonic coefficients.
Although described with respect to MOA representations, the techniques of this disclosure may also be performed with respect to first-order ambisonic (FOA) representations in which all of the ambisonic coefficients associated with a first order spherical basis function and a zero order spherical basis function are used to represent the soundfield. In other words, rather than represent the soundfield using a partial, non-zero subset of the ambisonic coefficients, the soundfield representation generator 302 may represent the soundfield using all of the ambisonic coefficients for a given order N, resulting in a total of ambisonic coefficients equaling (N+1)2.
In this respect, the ambisonic audio data (which is another way to refer to the ambisonic coefficients in either MOA representations or full order representation, such as the first-order representation noted above) may include ambisonic coefficients associated with spherical basis functions having an order of one or less (which may be referred to as “1st order ambisonic audio data”), ambisonic coefficients associated with spherical basis functions having a mixed order and suborder (which may be referred to as the “MOA representation” discussed above), or ambisonic coefficients associated with spherical basis functions having an order greater than one (which is referred to above as the “full order representation”).
The content capture device 300 may, in some examples, be configured to wirelessly communicate with the soundfield representation generator 302. In some examples, the content capture device 300 may communicate, via one or both of a wireless connection or a wired connection, with the soundfield representation generator 302. Via the connection between the content capture device 300 and the soundfield representation generator 302, the content capture device 300 may provide content in various forms of content, which, for purposes of discussion, are described herein as being portions of the HOA coefficients 11.
In some examples, the content capture device 300 may leverage various aspects of the soundfield representation generator 302 (in terms of hardware or software capabilities of the soundfield representation generator 302). For example, the soundfield representation generator 302 may include dedicated hardware configured to (or specialized software that when executed causes one or more processors to) perform psychoacoustic audio encoding (such as a unified speech and audio coder denoted as “USAC” set forth by the Moving Picture Experts Group (MPEG), the MPEG-H 3D audio coding standard, the MPEG-I Immersive Audio standard, or proprietary standards, such as AptX™ (including various versions of AptX such as enhanced AptX—E-AptX, AptX live, AptX stereo, and AptX high definition—AptX-HD), advanced audio coding (AAC), Audio Codec 3 (AC-3), Apple Lossless Audio Codec (ALAC), MPEG-4 Audio Lossless Streaming (ALS), enhanced AC-3, Free Lossless Audio Codec (FLAC), Monkey's Audio, MPEG-1 Audio Layer II (MP2), MPEG-1 Audio Layer III (MP3), Opus, and Windows Media Audio (WMA).
The content capture device 300 may not include the psychoacoustic audio encoder dedicated hardware or specialized software and instead provide audio aspects of the content 301 in a non-psychoacoustic-audio-coded form. The soundfield representation generator 302 may assist in the capture of content 301 by, at least in part, performing psychoacoustic audio encoding with respect to the audio aspects of the content 301.
The soundfield representation generator 302 may also assist in content capture and transmission by generating one or more bitstreams 21 based, at least in part, on the audio content (e.g., MOA representations and/or third order HOA representations) generated from the HOA coefficients 11. The bitstream 21 may represent a compressed version of the HOA coefficients 11 (and/or the partial subsets thereof used to form MOA representations of the soundfield) and any other different types of the content 301 (such as a compressed version of spherical video data, image data, or text data).
The soundfield representation generator 302 may generate the bitstream 21 for transmission, as one example, across a transmission channel, which may be a wired or wireless channel, a data storage device, or the like. The bitstream 21 may represent an encoded version of the HOA coefficients 11 (and/or the partial subsets thereof used to form MOA representations of the soundfield) and may include a primary bitstream and another side bitstream, which may be referred to as side channel information. In some instances, the bitstream 21 representing the compressed version of the HOA coefficients may conform to bitstreams produced in accordance with the MPEG-H 3D audio coding standard.
The content consumer device 14 may be operated by an individual, and may represent a VR client device. Although described with respect to a VR client device, content consumer device 14 may represent other types of devices, such as an augmented reality (AR) client device, a mixed reality (MR) client device (or any other type of head-mounted display device), a standard computer, a headset, headphones, or any other device capable of tracking head movements and/or general translational movements of the individual operating the client consumer device 14. As shown in the example of
The content consumer device 14 may retrieve the bitstream 21 directly from the source device 12. In some examples, the content consumer device 12 may interface with a network, including a fifth generation (5G) cellular network, to retrieve the bitstream 21 or otherwise cause the source device 12 to transmit the bitstream 21 to the content consumer device 14.
While shown in
Alternatively, the source device 12 may store the bitstream 21 to a storage medium, such as a compact disc, a digital video disc, a high definition video disc or other storage media, most of which are capable of being read by a computer and therefore may be referred to as computer-readable storage media or non-transitory computer-readable storage media. In this context, the transmission channel may refer to the channels by which content stored to the mediums are transmitted (and may include retail stores and other store-based delivery mechanism). In any event, the techniques of this disclosure should not therefore be limited in this respect to the example of
As noted above, the content consumer device 14 includes the audio playback system 16. The audio playback system 16 may represent any system capable of playing back multi-channel audio data. The audio playback system 16A may include a number of different audio renderers 22. The renderers 22 may each provide for a different form of audio rendering, where the different forms of rendering may include one or more of the various ways of performing vector-base amplitude panning (VBAP), and/or one or more of the various ways of performing soundfield synthesis. As used herein, “A and/or B” means “A or B”, or both “A and B”.
The audio playback system 16A may further include an audio decoding device 24. The audio decoding device 24 may represent a device configured to decode bitstream 21 to output reconstructed HOA coefficients 11A′-11N′ (which may form the full first, second, and/or third order ambisonic representation or a subset thereof that forms an MOA representation of the same soundfield or decompositions thereof, such as the predominant audio signal, ambient ambisonic coefficients, and the vector based signal described in the MPEG-H 3D Audio Coding Standard and/or the MPEG-I Immersive Audio standard).
As such, the ambisonic coefficients 11A′-11N′ (“ambisonic coefficients 11′”) may be similar to a full set or a partial subset of the ambisonic coefficients 11, but may differ due to lossy operations (e.g., quantization) and/or transmission via the transmission channel. The audio playback system 16 may, after decoding the bitstream 21 to obtain the ambisonic coefficients 11′, obtain ambisonic audio data 15 from the different streams of ambisonic coefficients 11′, and render the ambisonic audio data 15 to output speaker feeds 25. The speaker feeds 25 may drive one or more speakers (which are not shown in the example of
To select the appropriate renderer or, in some instances, generate an appropriate renderer, the audio playback system 16A may obtain loudspeaker information 13 indicative of a number of loudspeakers and/or a spatial geometry of the loudspeakers. In some instances, the audio playback system 16A may obtain the loudspeaker information 13 using a reference microphone and outputting a signal to activate (or, in other words, drive) the loudspeakers in such a manner as to dynamically determine, via the reference microphone, the loudspeaker information 13. In other instances, or in conjunction with the dynamic determination of the loudspeaker information 13, the audio playback system 16A may prompt a user to interface with the audio playback system 16A and input the loudspeaker information 13.
The audio playback system 16A may select one of the audio renderers 22 based on the loudspeaker information 13. In some instances, the audio playback system 16A may, when none of the audio renderers 22 are within some threshold similarity measure (in terms of the loudspeaker geometry) to the loudspeaker geometry specified in the loudspeaker information 13, generate the one of audio renderers 22 based on the loudspeaker information 13. The audio playback system 16A may, in some instances, generate one of the audio renderers 22 based on the loudspeaker information 13 without first attempting to select an existing one of the audio renderers 22.
When outputting the speaker feeds 25 to headphones, the audio playback system 16A may utilize one of the renderers 22 that provides for binaural rendering using head-related transfer functions (HRTF) or other functions capable of rendering to left and right speaker feeds 25 for headphone speaker playback. The terms “speakers” or “transducer” may generally refer to any speaker, including loudspeakers, headphone speakers, etc. One or more speakers may then playback the rendered speaker feeds 25.
Although described as rendering the speaker feeds 25 from the ambisonic audio data 15, reference to rendering of the speaker feeds 25 may refer to other types of rendering, such as rendering incorporated directly into the decoding of the ambisonic audio data 15 from the bitstream 21. An example of the alternative rendering can be found in Annex G of the MPEG-H 3D audio coding standard, where rendering occurs during the predominant signal formulation and the background signal formation prior to composition of the soundfield. As such, reference to rendering of the ambisonic audio data 15 should be understood to refer to both rendering of the actual ambisonic audio data 15 or decompositions or representations thereof of the ambisonic audio data 15 (such as the above noted predominant audio signal, the ambient ambisonic coefficients, and/or the vector-based signal—which may also be referred to as a V-vector).
As described above, the content consumer device 14 may represent a VR device in which a human wearable display is mounted in front of the eyes of the user operating the VR device.
Video, audio, and other sensory data may play important roles in the VR experience. To participate in a VR experience, a user 402 may wear the VR device 400A (which may also be referred to as a VR headset 400A) or other wearable electronic device. The VR client device (such as the VR headset 400A) may track head movement of the user 402, and adapt the video data shown via the VR headset 400A to account for the head movements, providing an immersive experience in which the user 402 may experience a virtual world shown in the video data in visual three dimensions.
While VR (and other forms of AR and/or MR, which may generally be referred to as a computer mediated reality device) may allow the user 402 to reside in the virtual world visually, often the VR headset 400A may lack the capability to place the user in the virtual world audibly. In other words, the VR system (which may include a computer responsible for rendering the video data and audio data—that is not shown in the example of
The wearable device 400B may represent other types of devices, such as a watch (including so-called “smart watches”), glasses (including so-called “smart glasses”), headphones (including so-called “wireless headphones” and “smart headphones”), smart clothing, smart jewelry, and the like. Whether representative of a VR device, a watch, glasses, and/or headphones, the wearable device 400B may communicate with the computing device supporting the wearable device 400B via a wired connection or a wireless connection.
In some instances, the computing device supporting the wearable device 400B may be integrated within the wearable device 400B and as such, the wearable device 400B may be considered as the same device as the computing device supporting the wearable device 400B. In other instances, the wearable device 400B may communicate with a separate computing device that may support the wearable device 400B. In this respect, the term “supporting” should not be understood to require a separate dedicated device but that one or more processors configured to perform various aspects of the techniques described in this disclosure may be integrated within the wearable device 400B or integrated within a computing device separate from the wearable device 400B.
For example, when the wearable device 400B represents an example of the VR device 400B, a separate dedicated computing device (such as a personal computer including the one or more processors) may render the audio and visual content, while the wearable device 400B may determine the translational head movement upon which the dedicated computing device may render, based on the translational head movement, the audio content (as the speaker feeds) in accordance with various aspects of the techniques described in this disclosure. As another example, when the wearable device 400B represents smart glasses, the wearable device 400B may include the one or more processors that both determine the translational head movement (by interfacing within one or more sensors of the wearable device 400B) and render, based on the determined translational head movement, the speaker feeds.
As shown, the wearable device 400B includes one or more directional speakers, and one or more tracking and/or recording cameras. In addition, the wearable device 400B includes one or more inertial, haptic, and/or health sensors, one or more eye-tracking cameras, one or more high sensitivity audio microphones, and optics/projection hardware. The optics/projection hardware of the wearable device 400B may include durable semi-transparent display technology and hardware.
The wearable device 400B also includes connectivity hardware, which may represent one or more network interfaces that support multimode connectivity, such as 4G communications, 5G communications, Bluetooth, etc. The wearable device 400B also includes one or more ambient light sensors, and bone conduction transducers. In some instances, the wearable device 400B may also include one or more passive and/or active cameras with fisheye lenses and/or telephoto lenses. Although not shown in
Furthermore, the tracking and recording cameras and other sensors may facilitate the determination of translational distance. Although not shown in the example of
Although described with respect to particular examples of wearable devices, such as the VR device 400B discussed above with respect to the examples of
In any event, the audio aspects of VR have been classified into three separate categories of immersion. The first category provides the lowest level of immersion, and is referred to as three degrees of freedom (3DOF). 3DOF refers to audio rendering that accounts for movement of the head in the three degrees of freedom (yaw, pitch, and roll), thereby allowing the user to freely look around in any direction. 3DOF, however, cannot account for translational head movements in which the head is not centered on the optical and acoustical center of the soundfield.
The second category, referred to 3DOF plus (3DOF+), provides for the three degrees of freedom (yaw, pitch, and roll) in addition to limited spatial translational movements due to the head movements away from the optical center and acoustical center within the soundfield. 3DOF+ may provide support for perceptual effects such as motion parallax, which may strengthen the sense of immersion.
The third category, referred to as six degrees of freedom (6DOF), renders audio data in a manner that accounts for the three degrees of freedom in term of head movements (yaw, pitch, and roll) but also accounts for translation of the user in space (x, y, and z translations). The spatial translations may be induced by sensors tracking the location of the user in the physical world or by way of an input controller.
3DOF rendering is the current state of the art for audio aspects of VR. As such, the audio aspects of VR are less immersive than the video aspects, thereby potentially reducing the overall immersion experienced by the user, and introducing localization errors (e.g., such as when the auditory playback does not match or correlate exactly to the visual scene).
In accordance with the techniques described in this disclosure, various ways are described to perform interpolation with respect to the existing audio streams 11 and thereby allow for 6DOF immersion. As described below, the techniques may improve the listener experience, while also reducing soundfield reproduction localization errors, as the interpolated audio stream may better reflect a location of a listener relative to the existing audio streams, thereby improving the operation of a playback device (that performs the techniques to reproduce the soundfield) itself
In operation, the audio playback system 16A may include an interpolation device 30 (“INT DEVICE 30”), e.g., as shown in
The interpolation device may be implemented by one or more processors, including fixed function processing circuitry and/or programmable processing circuitry, such as one or more digital signal processors (DSPs), general purpose microprocessors, application specific integrated circuits (ASICs), field programmable gate arrays (FPGAs), or other equivalent integrated or discrete logic circuitry.
The interpolation device 30 may first obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured the one or more audio streams 11′. More information regarding operation of the interpolation device 30 is described with respect to the examples of
The interpolation device 30 may also receive audio metadata 511A-511N (“audio metadata 511”), which may include a microphone location identifying a location of a corresponding microphone 5A-5N that captured the corresponding one of the audio streams 11′. The microphones 5 may provide the microphone location, an operator of the microphones 5 may enter the microphone locations, a device coupled to the microphone (e.g., the content capture device 300) may specify the microphone location, or some combination of the foregoing. The content capture device 300 may specify the audio metadata 511 as part of the content 301. In any event, the interpolation device 30 may parse the audio metadata 511 from the bitstream 21 representative of the content 301.
The interpolation device 30 may also obtain a listener location 17 that identifies a location of a listener, such as that shown in the example of
The interpolation device 30 may next perform interpolation, based on the one or more microphone locations and the listener location 17, with respect to the audio streams 11′ to obtain interpolated audio stream 15. The audio streams 11′ may be stored in a memory of the interpolation device 30. To perform the interpolation, the interpolation device 30 may read the audio streams 11′ form memory and determine, based on the one or more microphones locations and the listener location 17 (which may also be stored in the memory), a weight for each of the audio streams (which are shown as Weight(1) . . . Weight(n)).
To determine the weights, the interpolation device 30 may calculate each weight as a ratio of inverse distance to the listener location 17 for the corresponding one of the audio streams 11′ by the total inverse distance from all of the other audio streams 11′, except for the edge cases when the listener is at the same location as one of the microphones 5 as represented in the virtual world. That is to say, it may be possible for a listener to navigate a virtual world, or a real world location represented on a display of a device, which has the same location as where one of the microphones 5 captured the audio streams 11′. When the listener is at the same location as one of the microphones 5, the interpolation unit 30 may calculate the weight for the one of the audio streams 11′ captured by the one of the microphones 5 at which the listener is at the same location as one of the microphones 5, and the weights for the remaining audio streams 11′ are set to zero.
Otherwise, the interpolation device 30 may calculate each weight as follows: Weight(n)=(1/(distance of mic n to the listener position))/(1/(distance of mic 1 to the listener position)+ . . . +1/(distance of mic n to the listener position)), In the above, the listener position refers to the listener position 17, Weight(n) refers to the weight for the audio stream 11N′, and the distance of mic <number> to the listener position refers to the absolute value of the difference between the corresponding microphone location and the listener position 17.
The interpolation device 30 may next multiply the weight by the corresponding one of the audio streams 11′ to obtain one or more weighted audio streams, which the interpolation device 30 may add together to obtain the interpolated audio stream 15. The foregoing may be denoted mathematically by the following equation: Weight(1)*audio stream 1+ . . . +Weight(n)*audio stream n=Interpolated audio stream, where Weight(<number>) denotes the weight for the corresponding audio stream <number>, and the interpolated ambisonic audio data refers to the interpolated audio stream 15. The interpolated audio stream may be stored in the memory of the interpolation device 30 and may also be available to be played out by loudspeakers (e.g., a VR or AR device or a headset worn by the listener). The interpolation equation represents the weighted average ambisonic audio shown in the example of
In some examples, the interpolation device 30 may determine the foregoing weights on a frame-by-frame basis. In other examples, the interpolation device 30 may determine the foregoing weights on a more frequent basis (e.g., some sub-frame basis) or on a more infrequent basis (e.g., after some set number of frames). In these and other examples, the interpolation device 30 may only calculate the weights responsive to detection of some change in the listener location and/or orientation or responsive to some other characteristics of the underlying ambisonic audio streams (which may enable and disable various aspects of the interpolation techniques described in this disclosure).
In some examples, the above techniques may only be enabled with respect to the audio streams 11′ having certain characteristics. For example, the interpolation device 30 may only interpolate the audio streams 11′ when audio sources represented by the audio streams 11′ are located at locations different than the microphones 5. More information regarding this aspect of the techniques is provided below with respect to
Returning to the example of
As the listener 52 starts navigating from the starting location, the interpolation device 30 may generate the interpolated audio stream 15 to heavily weight the audio stream 11C′ captured by the microphone 5C, and assign relatively less weight to the audio stream 11B′ captured by the microphone 5B and the audio stream 11D′ captured by the microphone 5D, and still relatively less weight (and possibly no weight) to the audio streams 11A′ and 11E′ captured by the respective microphones 5A and 5E.
As the listener 52 navigates along the line 56 next to the location of the microphone 5B, the interpolation device 30 may assign more weight to the audio stream 11B′, relatively less weight to the audio stream 11C′ and yet less weight (and possibly no weight) to the audio streams 11A′, 11D′, and 11E′. As the listener 52 navigates (where the notch indicates the direction in which the listener 52 is moving) closer to the location of the microphone 5E toward the end of the line 56, the interpolation device 30 may assign more weight to the audio stream 11E′, relatively less weight to the audio stream 11A′, and yet relatively less weight (and possibly no weight) to the audio streams 11B′, 11C′, and 11D′.
In this respect, the interpolation device 30 may perform interpolation based on changes to the listener location 17 based on navigational commands issued by the listener 32 to assign varying weights over time to the audio streams 11A′-11E′. The changing listener location 17 may result in different emphasis within the interpolated audio stream 15, thereby promoting better auditory localization within the area 54.
Although not described in the examples set forth above, the techniques may also adapt to changes in the location of the microphones. In other words, the microphones may be manipulated during recording, changing locations and orientations. Because the above noted equations are only concerned with differences between the microphone locations and the listener location 17, the interpolation device 30 may continue to perform the interpolation even though the microphones have been manipulated to change location and/or orientation.
The audio playback system 16B may output the left and right speaker feeds 103 to headphones 104, which may represent another example of a wearable device and which may be coupled to additional wearable devices to facilitate reproduction of the soundfield, such as a watch, the VR headset noted above, smart glasses, smart clothing, smart rings, smart bracelets or any other types of smart jewelry (including smart necklaces), and the like. The headphones 104 may couple wirelessly or via wired connection to the additional wearable devices.
Additionally, the headphones 104 may couple to the audio playback system 16 via a wired connection (such as a standard 3.5 mm audio jack, a universal system bus (USB) connection, an optical audio jack, or other forms of wired connection) or wirelessly (such as by way of a Bluetooth™ connection, a wireless network connection, and the like). The headphones 104 may recreate, based on the left and right speaker feeds 103, the soundfield represented by the ambisonic coefficients 11. The headphones 104 may include a left headphone speaker and a right headphone speaker which are powered (or, in other words, driven) by the corresponding left and right speaker feeds 103.
Although described with respect to a VR device as shown in the example of
In the example of
The headphones 104 may couple to the audio playback system 16 via a wired connection (such as a standard 3.5 mm audio jack, a universal system bus (USB) connection, an optical audio jack, or other forms of wired connection) or wirelessly (such as by way of a Bluetooth™ connection, a wireless network connection, and the like). The headphones 104 may recreate, based on the left and right speaker feeds 103, the soundfield represented by the ambisonic coefficients 11. The headphones 104 may include a left headphone speaker and a right headphone speaker which are powered (or, in other words, driven) by the corresponding left and right speaker feeds 103.
The interpolation device 30 may, as described above in more detail, perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream (954). The audio playback system 16 may next invoke the audio renderers 22 to obtain, based on the interpolated audio streams (e.g., ambisonic audio data 15), one or more speaker feeds 25 (956). The audio playback system 16 may output the one or more speaker feeds 25 (958) to drive or otherwise power transducers (e.g., speakers).
The audio decoding device 24 may include a low delay decoder 900A, an audio decoder 900B, and a local audio buffer 902. The low delay decoder 900A may process XR audio bitstream 21A to obtain audio stream 901A, where the low delay decoder 900A may perform relatively low complexity decoding (compared to the audio decoder 900B) to facilitate low delay reconstruction of the audio stream 901A. The audio decoder 900B may perform relatively higher complexity decoding (compared to the audio decoder 900A) with respect to the audio bitstream 21B to obtain audio stream 901B. The audio decoder 900B may perform audio decoding that conforms to the MPEG-H 3D Audio coding standard. The local audio buffer 902 may represent a unit configured to buffer local audio content, which the local audio buffer 902 may output as audio stream 903.
The bitstream 21 (comprised of one or more of the XR audio bitstream 21A and/or the audio bitstream 21B) may also include XR metadata 905A (which may include the microphone location information noted above) and 6DOF metadata 905B (which may specify various parameters related to 6DOF audio rendering). The 6DOF audio renderer 22A may obtain the audio streams 901A, 901B, and/or 903 along with the XR metadata 905A and the 6DOF metadata 905B and render the speaker feeds 25 and/or 103 based on the listener positions and the microphone positions. In the example of
Base stations 105 may wirelessly communicate with UEs 115 via one or more base station antennas. Base stations 105 described herein may include or may be referred to by those skilled in the art as a base transceiver station, a radio base station, an access point, a radio transceiver, a NodeB, an eNodeB (eNB), a next-generation NodeB or giga-NodeB (either of which may be referred to as a gNB), a Home NodeB, a Home eNodeB, or some other suitable terminology. Wireless communications system 100 may include base stations 105 of different types (e.g., macro or small cell base stations). The UEs 115 described herein may be able to communicate with various types of base stations 105 and network equipment including macro eNBs, small cell eNBs, gNBs, relay base stations, and the like.
Each base station 105 may be associated with a particular geographic coverage area 110 in which communications with various UEs 115 is supported. Each base station 105 may provide communication coverage for a respective geographic coverage area 110 via communication links 125, and communication links 125 between a base station 105 and a UE 115 may utilize one or more carriers. Communication links 125 shown in wireless communications system 100 may include uplink transmissions from a UE 115 to a base station 105, or downlink transmissions from a base station 105 to a UE 115. Downlink transmissions may also be called forward link transmissions while uplink transmissions may also be called reverse link transmissions.
The geographic coverage area 110 for a base station 105 may be divided into sectors making up a portion of the geographic coverage area 110, and each sector may be associated with a cell. For example, each base station 105 may provide communication coverage for a macro cell, a small cell, a hot spot, or other types of cells, or various combinations thereof. In some examples, a base station 105 may be movable and therefore provide communication coverage for a moving geographic coverage area 110. In some examples, different geographic coverage areas 110 associated with different technologies may overlap, and overlapping geographic coverage areas 110 associated with different technologies may be supported by the same base station 105 or by different base stations 105. The wireless communications system 100 may include, for example, a heterogeneous LTE/LTE-A/LTE-A Pro or NR network in which different types of base stations 105 provide coverage for various geographic coverage areas 110.
UEs 115 may be dispersed throughout the wireless communications system 100, and each UE 115 may be stationary or mobile. A UE 115 may also be referred to as a mobile device, a wireless device, a remote device, a handheld device, or a subscriber device, or some other suitable terminology, where the “device” may also be referred to as a unit, a station, a terminal, or a client. A UE 115 may also be a personal electronic device such as a cellular phone, a personal digital assistant (PDA), a tablet computer, a laptop computer, or a personal computer. In examples of this disclosure, a UE 115 may be any of the audio sources described in this disclosure, including a VR headset, an XR headset, an AR headset, a vehicle, a smartphone, a microphone, an array of microphones, or any other device including a microphone or is able to transmit a captured and/or synthesized audio stream. In some examples, an synthesized audio stream may be an audio stream that was stored in memory or was previously created or synthesized. In some examples, a UE 115 may also refer to a wireless local loop (WLL) station, an Internet of Things (IoT) device, an Internet of Everything (IoE) device, or an MTC device, or the like, which may be implemented in various articles such as appliances, vehicles, meters, or the like.
Some UEs 115, such as MTC or IoT devices, may be low cost or low complexity devices, and may provide for automated communication between machines (e.g., via Machine-to-Machine (M2M) communication). M2M communication or MTC may refer to data communication technologies that allow devices to communicate with one another or a base station 105 without human intervention. In some examples, M2M communication or MTC may include communications from devices that exchange and/or use audio metadata indicating privacy restrictions and/or password-based privacy data to toggle, mask, and/or null various audio streams and/or audio sources as will be described in more detail below.
In some cases, a UE 115 may also be able to communicate directly with other UEs 115 (e.g., using a peer-to-peer (P2P) or device-to-device (D2D) protocol). One or more of a group of UEs 115 utilizing D2D communications may be within the geographic coverage area 110 of a base station 105. Other UEs 115 in such a group may be outside the geographic coverage area 110 of a base station 105, or be otherwise unable to receive transmissions from a base station 105. In some cases, groups of UEs 115 communicating via D2D communications may utilize a one-to-many (1:M) system in which each UE 115 transmits to every other UE 115 in the group. In some cases, a base station 105 facilitates the scheduling of resources for D2D communications. In other cases, D2D communications are carried out between UEs 115 without the involvement of a base station 105.
Base stations 105 may communicate with the core network 130 and with one another. For example, base stations 105 may interface with the core network 130 through backhaul links 132 (e.g., via an S1, N2, N3, or other interface). Base stations 105 may communicate with one another over backhaul links 134 (e.g., via an X2, Xn, or other interface) either directly (e.g., directly between base stations 105) or indirectly (e.g., via core network 130).
In some cases, wireless communications system 100 may utilize both licensed and unlicensed radio frequency spectrum bands. For example, wireless communications system 100 may employ License Assisted Access (LAA), LTE-Unlicensed (LTE-U) radio access technology, or NR technology in an unlicensed band such as the 5 GHz ISM band. When operating in unlicensed radio frequency spectrum bands, wireless devices such as base stations 105 and UEs 115 may employ listen-before-talk (LBT) procedures to ensure a frequency channel is clear before transmitting data. In some cases, operations in unlicensed bands may be based on a carrier aggregation configuration in conjunction with component carriers operating in a licensed band (e.g., LAA). Operations in unlicensed spectrum may include downlink transmissions, uplink transmissions, peer-to-peer transmissions, or a combination of these. Duplexing in unlicensed spectrum may be based on frequency division duplexing (FDD), time division duplexing (TDD), or a combination of both.
In this respect, various aspects of the techniques are described that enable one or more of the following examples:
Example 1. A device configured to process one or more audio streams, the device comprising: a memory configured to store the one or more audio streams; and a processor coupled to the memory, and configured to: obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtain a listener location identifying a location of a listener; perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtain, based on the interpolated audio stream, one or more speaker feeds; and output the one or more speaker feeds.
Example 2. The device of example 1, wherein the one or more processors are configured to: determine, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and obtain, based on the weight, the interpolated audio stream.
Example 3. The device of example 1, wherein the one or more processors are configured to: determine, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and multiply the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and obtain, based on the one or more weighted audio streams, the interpolated audio stream.
Example 4. The device of example 1, wherein the one or more processors are configured to: determine, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and multiply the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and add the one or more weighted audio streams together to obtain the interpolated audio stream.
Example 5. The device of any combination of examples 2-4, wherein the one or more processors are configured to: determine a difference between each of the one or more microphone locations and the listener location; and determine, based on the difference between each of the one or more microphone locations and the listener location, the weight for each of the audio streams.
Example 6. The device of any combination of examples 2-5, wherein the one or more processors are configured to determine the weights for each audio frame of the one or more audio streams.
Example 7. The device of any combination of examples 1-6, wherein audio sources represented by the audio streams reside outside of the one or more microphones.
Example 8. The device of any combination of examples 1-7, wherein the one or more processors are configured to obtain, from a computer mediated reality device, the listener location.
Example 9. The device of example 8, wherein the computer mediated reality device comprises a head mounted display device.
Example 10. The device of any combination of examples 1-9, wherein the one or more processors are configured to obtain, from a bitstream that includes the audio streams, audio metadata that identifies the one or more microphone locations.
Example 11. The device of any combination of examples 1-10, wherein at least one of the one or more microphone locations changes to reflect movement of the corresponding one of the one or more microphones.
Example 12. The device of any combination of examples 1-11, wherein the one or more audio streams include a ambisonic audio stream (including higher order, mixed order, first order, second order), and wherein the interpolated audio stream includes an interpolated ambisonic audio stream (including higher order, mixed order, first order, second order).
Example 13. The device of any combination of claims 1-11, wherein the one or more audio streams include an ambisonic audio stream, and wherein the interpolated audio stream includes an interpolated ambisonic audio stream.
Example 14. The device of any combination of examples 1-13, wherein the listener location changes based on navigational commands issued by the listener.
Example 15. The device of any combination of examples 1-14, wherein the one or more processors are configured to receive audio metadata specifying the microphone locations, each of the microphone locations identifying a location of a cluster of microphones that captured the corresponding one or more audio streams.
Example 16. The device of any combination of examples 15, wherein the cluster of microphones are each positioned at a distance from one another that is greater than five feet.
Example 17. The device of any combination of examples 1-14, wherein the microphones are each positioned at a distance greater than five feet from one another.
Example 18. A method for processing one or more audio streams, the method comprising: obtaining one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtaining a listener location identifying a location of a listener; performing interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtaining, based on the interpolated audio stream, one or more speaker feeds; and outputting the one or more speaker feeds.
Example 19. The method of example 18, wherein performing the interpolation comprises: determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and obtaining, based on the weight, the interpolated audio stream.
Example 20. The method of example 18, wherein performing the interpolation comprises: determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; multiplying the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and obtaining, based on the one or more weighted audio streams, the interpolated audio stream.
Example 21. The method of example 18, wherein performing the interpolation comprises: determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and multiplying the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and adding the one or more weighted audio streams together to obtain the interpolated audio stream.
Example 22. The method of any combination of example 19-21, wherein determining the weights comprises: determining a difference between each of the one or more microphone locations and the listener location; and determining, based on the difference between each of the one or more microphone locations and the listener location, the weight for each of the audio streams.
Example 23. The method of any combination of example 19-22, wherein determining the weights comprises determining the weights for each audio frame of the one or more audio streams.
Example 24. The method of any combination of examples 18-23, wherein audio sources represented by the audio streams reside outside of the one or more microphones.
Example 25. The method of any combination of examples 18-24, wherein obtaining the listener location comprises obtaining, from a computer mediated reality device, the listener location.
Example 26. The method of example 25, wherein the computer mediated reality device comprises a head mounted display device.
Example 27. The method of any combination of examples 18-26, wherein obtaining the one or more microphone locations comprises obtaining, from a bitstream that includes the audio streams, audio metadata that identifies the one or more microphone locations.
Example 28. The method of any combination of examples 18-27, wherein at least one of the one or more microphone locations changes to reflect movement of the corresponding one of the one or more microphones.
Example 29. The method of any combination of examples 18-28, wherein the one or more audio streams include a ambisonic audio stream (including higher order, mixed order, first order, second order), and wherein the interpolated audio stream includes an interpolated ambisonic audio stream (including higher order, mixed order, first order, second order).
Example 30. The method of any combination of examples 18-28, wherein the one or more audio streams include an ambisonic audio stream, and wherein the interpolated audio stream includes an interpolated ambisonic audio stream.
Example 31. The method of any combination of examples 18-30, wherein the listener location changes based on navigational commands issued by the listener.
Example 32. The method of any combination of examples 18-31, wherein obtaining the microphone locations comprises receiving audio metadata specifying the microphone locations, each of the microphone locations identifying a location of a cluster of microphones that captured the corresponding one or more audio streams.
Example 33. The method of example 32, wherein the cluster of microphones are each positioned at a distance from one another that is greater than five feet.
Example 34. The method of any combination of examples 18-31, wherein the microphones are each positioned at a distance greater than five feet from one another.
Example 35. A device configured to process one or more audio streams, the device comprising: means for obtaining one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; means for obtaining a listener location identifying a location of a listener; means for performing interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; means for obtaining, based on the interpolated audio stream, one or more speaker feeds; and means for outputting the one or more speaker feeds.
Example 36. The device of example 35, wherein the means for performing the interpolation comprises: means for determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; and means for obtaining, based on the weight, the interpolated audio stream.
Example 37. The device of example 35, wherein the means for performing the interpolation comprises: means for determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; means for multiplying the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and means for obtaining, based on the one or more weighted audio streams, the interpolated audio stream.
Example 38. The device of example 35, wherein the means for performing the interpolation comprises: means for determining, based on the one or more microphone locations and the listener location, a weight for each of the audio streams; means for multiplying the weight by the corresponding one of the one or more audio streams to obtain one or more weighted audio stream; and means for adding the one or more weighted audio streams together to obtain the interpolated audio stream.
Example 39. The device of any combination of examples 36-38, wherein the means for determining the weights comprises: means for determining a difference between each of the one or more microphone locations and the listener location; and means for determining, based on the difference between each of the one or more microphone locations and the listener location, the weight for each of the audio streams.
Example 40. The device of any combination of examples 36-39, wherein the means for determining the weights comprises means for determining the weights for each audio frame of the one or more audio streams.
Example 41. The device of any combination of examples 35-40, wherein audio sources represented by the audio streams reside outside of the one or more microphones.
Example 42. The device of any combination of examples 35-41, wherein the means for obtaining the listener location comprises means for obtaining, from a computer mediated reality device, the listener location.
Example 43. The device of example 42, wherein the computer mediated reality device comprises a head mounted display device.
Example 44. The device of any combination of examples 35-43, wherein the means for obtaining the one or more microphone locations comprises means for obtaining, from a bitstream that includes the audio streams, audio metadata that identifies the one or more microphone locations.
Example 45. The device of any combination of examples 35-44, wherein at least one of the one or more microphone locations changes to reflect movement of the corresponding one of the one or more microphones.
Example 46. The device of any combination of examples 35-45, wherein the one or more audio streams include a ambisonic audio stream (including higher order, mixed order, first order, second order), and wherein the interpolated audio stream includes an interpolated ambisonic audio stream (including higher order, mixed order, first order, second order).
Example 47. The device of any combination of examples 35-44, wherein the one or more audio streams include an ambisonic audio stream, and wherein the interpolated audio stream includes an interpolated ambisonic audio stream.
Example 48. The device of any combination of examples 35-47, wherein the listener location changes based on navigational commands issued by the listener.
Example 49. The device of any combination of examples 35-48, wherein the means for obtaining the microphone locations comprises means for receiving audio metadata specifying the microphone locations, each of the microphone locations identifying a location of a cluster of microphones that captured the corresponding one or more audio streams.
Example 50. The device of any combination of examples 49, wherein the cluster of microphones are each positioned at a distance from one another that is greater than five feet.
Example 51. The device of any combination of examples 35-48, wherein the microphones are each positioned at a distance greater than five feet from one another.
Example 52. A non-transitory computer-readable storage medium having stored thereon instructions that, when executed, cause one or more processors to: obtain one or more microphone locations, each of the one or more microphone locations identifying a location of a respective one or more microphones that captured each of the corresponding one or more audio streams; obtain a listener location identifying a location of a listener; perform interpolation, based on the one or more microphone locations and the listener location, with respect to the audio streams to obtain an interpolated audio stream; obtain, based on the interpolated audio stream, one or more speaker feeds; and output the one or more speaker feeds.
It is to be recognized that depending on the example, certain acts or events of any of the techniques described herein can be performed in a different sequence, may be added, merged, or left out altogether (e.g., not all described acts or events are necessary for the practice of the techniques). Moreover, in certain examples, acts or events may be performed concurrently, e.g., through multi-threaded processing, interrupt processing, or multiple processors, rather than sequentially.
In some examples, the VR device (or the streaming device) may communicate, using a network interface coupled to a memory of the VR/streaming device, exchange messages to an external device, where the exchange messages are associated with the multiple available representations of the soundfield. In some examples, the VR device may receive, using an antenna coupled to the network interface, wireless signals including data packets, audio packets, video packets, or transport protocol data associated with the multiple available representations of the soundfield. In some examples, one or more microphone arrays may capture the soundfield.
In some examples, the multiple available representations of the soundfield stored to the memory device may include a plurality of object-based representations of the soundfield, higher order ambisonic representations of the soundfield, mixed order ambisonic representations of the soundfield, a combination of object-based representations of the soundfield with higher order ambisonic representations of the soundfield, a combination of object-based representations of the soundfield with mixed order ambisonic representations of the soundfield, or a combination of mixed order representations of the soundfield with higher order ambisonic representations of the soundfield.
In some examples, one or more of the soundfield representations of the multiple available representations of the soundfield may include at least one high-resolution region and at least one lower-resolution region, and wherein the selected presentation based on the steering angle provides a greater spatial precision with respect to the at least one high-resolution region and a lesser spatial precision with respect to the lower-resolution region.
In one or more examples, the functions described may be implemented in hardware, software, firmware, or any combination thereof. If implemented in software, the functions may be stored on or transmitted over as one or more instructions or code on a computer-readable medium and executed by a hardware-based processing unit. Computer-readable media may include computer-readable storage media, which corresponds to a tangible medium such as data storage media, or communication media including any medium that facilitates transfer of a computer program from one place to another, e.g., according to a communication protocol. In this manner, computer-readable media generally may correspond to (1) tangible computer-readable storage media which is non-transitory or (2) a communication medium such as a signal or carrier wave. Data storage media may be any available media that can be accessed by one or more computers or one or more processors to retrieve instructions, code and/or data structures for implementation of the techniques described in this disclosure. A computer program product may include a computer-readable medium.
By way of example, and not limitation, such computer-readable storage media can comprise RAM, ROM, EEPROM, CD-ROM or other optical disk storage, magnetic disk storage, or other magnetic storage devices, flash memory, or any other medium that can be used to store desired program code in the form of instructions or data structures and that can be accessed by a computer. Also, any connection is properly termed a computer-readable medium. For example, if instructions are transmitted from a website, server, or other remote source using a coaxial cable, fiber optic cable, twisted pair, digital subscriber line (DSL), or wireless technologies such as infrared, radio, and microwave, then the coaxial cable, fiber optic cable, twisted pair, DSL, or wireless technologies such as infrared, radio, and microwave are included in the definition of medium. It should be understood, however, that computer-readable storage media and data storage media do not include connections, carrier waves, signals, or other transitory media, but are instead directed to non-transitory, tangible storage media. Disk and disc, as used herein, includes compact disc (CD), laser disc, optical disc, digital versatile disc (DVD), floppy disk and Blu-ray disc, where disks usually reproduce data magnetically, while discs reproduce data optically with lasers. Combinations of the above should also be included within the scope of computer-readable media.
Instructions may be executed by one or more processors, including fixed function processing circuitry and/or programmable processing circuitry, such as one or more digital signal processors (DSPs), general purpose microprocessors, application specific integrated circuits (ASICs), field programmable gate arrays (FPGAs), or other equivalent integrated or discrete logic circuitry. Accordingly, the term “processor,” as used herein may refer to any of the foregoing structure or any other structure suitable for implementation of the techniques described herein. In addition, in some aspects, the functionality described herein may be provided within dedicated hardware and/or software modules configured for encoding and decoding, or incorporated in a combined codec. Also, the techniques could be fully implemented in one or more circuits or logic elements.
The techniques of this disclosure may be implemented in a wide variety of devices or apparatuses, including a wireless handset, an integrated circuit (IC) or a set of ICs (e.g., a chip set). Various components, modules, or units are described in this disclosure to emphasize functional aspects of devices configured to perform the disclosed techniques, but do not necessarily require realization by different hardware units. Rather, as described above, various units may be combined in a codec hardware unit or provided by a collection of interoperative hardware units, including one or more processors as described above, in conjunction with suitable software and/or firmware.
Various examples have been described. These and other examples are within the scope of the following claims.
This application claims the benefit of U.S. Provisional Application No. 62/700,267, entitled “INTERPOLATING AUDIO STREAMS,” and filed Jul. 18, 2018, and U.S. Provisional Application No. 62/870,586, entitled “INTERPOLATING AUDIO STREAMS,” and filed Jul. 3, 2019, the entire contents of both being hereby incorporated by reference as if set forth in its entirety.
Number | Date | Country | |
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62700267 | Jul 2018 | US | |
62870586 | Jul 2019 | US |