WIRELESS SURROUND SOUND SYSTEM AND METHOD

Abstract

A system and method including one or more processors and one or more non-transitory storage devices storing computing instructions, the computing instructions executing on the one or more processors are configured to perform operations of receiving audio source data at a speaker, applying a digital signal processing algorithm to the audio source data at the speaker to create post-processed audio data, encoding the post-processed audio data at the speaker, and outputting the post-processed audio data as encoded through the speaker. Other embodiments are also disclosed herein.

Description

[Technical field]

[0001] Cross-reference to related applications

[0001] This application claims the benefit of U.S. Provisional Patent Application No. 63 / 254,938, filed October 12, 2021, which is incorporated herein by reference in its entirety. Field

[0002] The present disclosure relates generally to wireless speaker systems, and more particularly to wireless surround sound speaker systems. [Background technology]

[0002]

[0003] Traditional surround sound speaker systems, which include multiple speakers, can be difficult to install, equalize, and operate in a home theater environment. Many of today's premium home multi-speaker surround sound systems require cumbersome wiring that must be run around the room and connect to bulky receivers or preamplifiers. Consumer demands for the highest quality audio in a modest décor have led to the emergence of sound bars, but sound bar systems are unable to deliver high quality audio adequately. Furthermore, such systems are not well suited to the advanced surround sound and effects found in high-end formats. Summary of the Invention

[0003]

[0004] In various embodiments, the present disclosure provides systems and methods for implementing surround sound. A system for implementing surround sound may include one or more processors and one or more non-transitory computer-readable storage devices that store computing instructions. The computing instructions, executing on the one or more processors, are configured to cause the one or more processors to perform operations of receiving audio source data at a speaker, applying a digital signal processing algorithm to the audio source data at the speaker to create post-processed audio data, encoding the post-processed audio data at the speaker, and outputting the post-processed audio data as encoded through the speaker.

[0004]

[0005] In various embodiments, the audio source data comprises a packet, the packet including a physical layer communication protocol portion followed by a standardized communication protocol header portion followed by a transport layer protocol portion and a standardized communication protocol message portion. In various embodiments, encoding the post-processed audio data includes splitting the post-processed audio data into at least two different audio data channels and adjusting a balance between frequency components of the at least two different audio data channels. In various embodiments, adjusting the balance includes applying one or more of an equalization effect and a filtering element.

[0005]

[0006] In various embodiments, the speaker includes a plurality of speakers, and the act of transmitting the post-processed audio data as encoded includes an act of transmitting a first audio data channel of the at least two distinct audio data channels to a first speaker of the plurality of speakers, and an act of transmitting a second audio data channel of the at least two distinct audio data channels to a second speaker of the plurality of speakers, the second speaker being different from the first speaker of the plurality of speakers. In various embodiments, the computing instructions are further configured to execute on the one or more processors and cause the processors to perform acts of receiving an AC signal from a power cable and generating a time based signal using the AC signal, and the act of applying a digital signal processing algorithm includes an act of applying a digital signal processing algorithm to the audio source data and the time based signal to create the post-processed audio data.

[0006]

[0007] In various embodiments, the operation of generating the time base signal includes an operation of generating the time base signal using an alternating current signal and a phase-locked loop circuit. In various embodiments, the time base signal includes a jitter-free reference frequency at a predetermined sample rate. In various embodiments, the computing instructions are further configured to execute on the one or more processors and cause the processors to perform an operation of applying a dropout mitigation method to the audio source data after receiving the audio source data at the speakers. In various embodiments, the dropout mitigation method includes one or more of a packet interpolation method, a spectral analysis method, a packet switching method using volumetric data, and a packet switching method using lossy compressed packets.

[0007]

[0008] The above-described features and elements may be combined in various combinations, and these combinations are not excluded, unless expressly stated otherwise in this specification. These features and elements, as well as the operation of the disclosed embodiments, will become more apparent with reference to the following description and the accompanying drawings. [Brief description of the drawings]

[0008]

[0009] The subject matter of the present disclosure is particularly pointed out and distinctly claimed in the concluding portion of this specification. However, a more complete understanding of the present disclosure can be obtained by reference to the detailed description and claims when considered in conjunction with the drawings, in which like numerals refer to like elements. [Figure 1] FIG. 1 is a block diagram illustrating various system components of a surround sound system according to various embodiments. [Diagram 2] FIG. 1 is a block diagram of a control module in a surround sound system according to various embodiments. [Diagram 3] FIG. 1 is a block diagram of a wireless speaker in a surround sound system according to various embodiments. [Figure 4] 1 illustrates a data control scheme in a surround sound system according to various embodiments. [Diagram 5] FIG. 1 is a block diagram of a wireless speaker in a surround sound system according to various embodiments. [Figure 6] 1 illustrates a process flow in a surround sound system according to various embodiments. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0009]

[0016] Some embodiments may include a system that may include one or more processors and one or more non-transitory computer readable storage devices that store computing instructions that execute on the one or more processors and cause the one or more processors to perform operations of receiving audio source data at a speaker, applying a digital signal processing algorithm to the audio source data at the speaker to create post-processed audio data, and The system may be configured to encode the post-processed audio data on a speaker, and output the encoded post-processed audio data through the speaker.

[0010]

[0017] Various embodiments include a method. The method may be implemented by execution of computing instructions configured to execute on one or more processors and stored on a non-transitory computer-readable medium. The method may include receiving audio source data at a speaker, applying a digital signal processing algorithm to the audio source data on the speaker to create post-processed audio data, encoding the post-processed audio data on the speaker, and outputting the encoded post-processed audio data through the speaker.

[0011]

[0018] In the detailed description of the exemplary embodiments herein, reference is made to the accompanying drawings, which illustrate various exemplary embodiments as representative examples and their best modes. Although these various exemplary embodiments are described in sufficient detail to enable one skilled in the art to practice the present disclosure, it should be understood that other embodiments may be realized, and that logical, chemical, and mechanical changes may be made without departing from the spirit and scope of the present disclosure. That is, the detailed description herein is presented for purposes of illustration only, and not limitation. For example, in any method or process description, the recited steps may be performed in any order and are not necessarily limited to the order presented. Furthermore, whenever a singular reference is made, a plurality of embodiments is also included, and whenever more than one component or step is referred to, a single embodiment or step is also included. Also, whenever a reference is made to attached, fixed, connected, etc., it can include permanent, removably, temporarily, partially, completely, and / or any other possible attachment options. Additionally, whenever reference is made to no contact (or similar phrases), little or minimal contact can also be included.

[0012]

[0019] An audio system, such as a home theater system, can have multiple speakers (e.g., 2, 4, 6, 8, 10, 12, 14, 34, or any number desired by a user). Traditional central amplifier based systems often require multiple pairs of wires, most typically one pair of wires to drive each speaker. In this regard, it is fair to say that traditional systems are cumbersome and time consuming to install.

[0013]

[0020] As described herein, the present system helps alleviate the problems of previous systems by providing each speaker with an independent power supply, amplifier, and data transport interface for streaming audio. In this regard, by locating the amplifier in the same enclosure as the speaker, the power and spectral characteristics of the amplifier can be tailored to the characteristics of the speaker and its enclosure, thereby improving efficiency and sound quality. The transmitting unit (i.e., control module) of the speaker system can include an input section, a processing system, a Bluetooth transceiver, a data transport device, and a power supply.

[0014]

[0021] The input section can accept data sources such as audio signals in the form of HDMI, TOSLink, Digital coax, analog inputs, stored data such as .mp3 or .wav files, or audio from a streaming network, computer, phone, or tablet. Audio can be input or converted as one or more digital streams of uncompressed samples with 16, 24, 32, and / or other bits per sample, at data rates of 44.1ksps, 48ksps, 96ksps, and / or other sample rates. Audio can also accommodate a number of channels, such as stereo, quad, 5.1, 7.1, and / or other formats. It can also be formatted for processing by a spatializer, for example, DOLBY ATMOS™.

[0015]

[0022] The processing system may perform a variety of functions. It may resample the incoming audio signal and convert the stream to a desired output sample rate. It may process the audio and provide digital signal processing (DSP) functions such as equalization, room compensation, speech enhancement, and / or add special effects such as echo or spatial separation enhancement. Effects may be applied to all audio channels or separately to each speaker channel. The processing system may communicate with a smartphone or tablet via a BLUETOOTH interface to allow user control of parameters such as volume, equalization levels, and / or effect selection. The processed digital audio channels may be converted into a stream of packets and sent to the speakers via a data transport device.

[0016]

[0023] The transceiver provides a link between the processing system in the control module and a device such as a smartphone or tablet for user control of the system. It will be appreciated that a BLUETOOTH interface is one exemplary type of interface, and other possibilities may include WiFi, a proprietary wireless link, and / or a wired connection. The smartphone or tablet may also be replaced or augmented by a dedicated interface device.

[0017]

[0024] The data transport device can send packetized digital audio data to the speaker module. The transmission method can be WiFi, HaLow, White Space Radio, 60 GHz radio, a proprietary radio design, and / or power line Ethernet such as G.Hn. In various embodiments, most of the bandwidth for this device (e.g., between 60% and 99%, or between 80% and 99%, or between 90% and 99%) is directed from the control module to the speakers, but a small amount of data can be sent in the other direction to discover active speakers in the system and / or constitute system control data. In addition to streaming audio data, some control information can also be included in the packets to control the operating conditions of the speakers. Such control information can include, for example, volume and mute functions, as well as control of any DSP functions. The DSP functions can also be implemented in the speaker module. Other control functions can include wake-up messages to wake up any speakers that are dormant (low power mode) when the system is transitioning from an idle state to an active state.

[0018]

[0025] Digital audio data can be received by the speaker's data transport device. This data passes through the speaker's processor. The processor can modify the signal using DSP algorithms, such as signal shaping, to match the speaker characteristics. The drive signal required for a particular speaker is sent to the amplifier that drives that speaker. The power supply circuit provides power to all the devices in the speaker unit.

[0019]

[0026] Many data transport systems have limited bandwidth. To make the most of this bandwidth, it is often advantageous to use data compression. Lossless compression, such as FLAC, can reduce the data rate while maintaining the desired audio quality. Lossy compression can further reduce the required data rate, but at the expense of sound quality. The choice of compression can vary depending on the number of speakers and the available bandwidth of the data transport system.

[0020]

[0027] In various embodiments, the system may employ User Datagram Protocol (UDP) for communication. UDP comprises low-level packetized data communication, but there is no guarantee of receipt. It reduces overhead by not assuming handshaking. In contrast, Transmission Control Protocol (TCP) communication is packetized data that guarantees delivery. However, TCP communication includes the risk of delays due to retransmission, which tends to increase system latency beyond practical thresholds. For example, when using a wireless audio system in conjunction with a video source, latency must be low to avoid problems with synchronization between video and audio. In various embodiments, the system latency is less than 25 ms, or less than 20 ms, or less than 15 ms, or less than 5 ms.

[0021]

[0028] In various embodiments, the system may employ one or more loss mitigation methods depending on the character of losses typically encountered in data transport systems. For example, if packet losses are infrequent and the number of consecutive packets lost when there is a loss is small (e.g., less than four), the system may employ a first lost packet handling method. The first method may include filling in the lost data by interpolating the last sample received before the loss and the first sample received after the loss. The second method may include performing a spectral analysis of the last correct packet received and the first correct packet after the gap, and then interpolating in the frequency and phase domains. A third method employed by the system to mitigate data losses is to determine where audio from one channel resembles audio from another channel. In this regard, if a packet from one speaker is lost, the system may replace it with a packet intended for a different speaker without any noticeable effect to the listener. This exchange can produce a packet-switched equalizer configured to enhance and make one channel sound more like another by tracking and comparing the overall volume differences between the various speakers and / or the differences in multiple frequency bands. A fourth missing packet handle method that can be employed by the system is to have each packet contain a lossy compressed version of the data of the subsequent packet. In normal operation, this lossy compressed data can be ignored. In response to a lost packet, data for this lost packet can be constructed from compressed data already received in a previous lossy compressed version of the associated packet.

[0022]

[0029] In various embodiments, the system can perform time base correction. The control module can send data at a nominal rate (e.g., 48,000 samples per second). This rate can depend on a crystal oscillator or other time base in the control module, or can be encoded on the data coming in to the unit. Thus, the frequency can be higher or lower than the nominal frequency by a small but measurable amount. Each speaker must receive these packets of samples and play them back at exactly the same rate as they were generated in the control module. However, if a speaker does not do this, but instead uses its own time base, which may be faster or slower than the time base in the control module, over time the speaker will either be ahead of the control module or behind the control module. This can result in an easily noticeable and unpleasant time difference between the speakers, which can easily lead to poor sound quality for the listener. Another problem that can arise from using a local time base at each speaker is that if the speaker runs slower than the control module, packets can accumulate (e.g., in a first-in, first-out (FIFO) memory) until the memory is exhausted. If the speaker runs faster than the control module, the speaker can run out of packets in the queue when it is ready to output a new sample.

[0023]

[0030] The system can perform a time base correction process. Initially, packets received at the speaker are stored locally in a FIFO buffer at the speaker. The FIFO buffer may be empty on power up, but after receiving a few packets, the FIFO buffer will accommodate a nominal number of packets (e.g., 4 packets). This nominal number can be used to set the latency of the system according to the number of samples in the packet. The FIFO buffer also allows for a loss mitigation method to fill in any lost packets, as described above. As new packets are inserted into the FIFO buffer, the FIFO buffer grows in size, and as packets are removed and sent to the speaker, the FIFO buffer shrinks. An oscillator that sets the sample output frequency can be controlled by a phase-locked loop. The frequency control can be adjusted in system software by a processing unit located in the speaker. If the FIFO buffer has fewer than the nominal number of packets in it, it responds by reducing the output sample rate. If the FIFO buffer has more than the nominal number of packets in it, it responds by increasing the output sample rate, thus allowing the system to maintain roughly the correct output rate.

[0024]

[0031] When the number of packets in the FIFO buffer is exactly the nominal number, the system can match the frequency of the incoming packets using a phase comparator and a loop filter. Each time a packet is received by the processor, the processor time stamps the reception event. This time stamp can be a counter driven by an oscillator. In this regard, the oscillator can also set the output frequency. By measuring with this clock, the system can match the sample rate in the control module when it measures exactly the same frequency at the speaker. The time stamps are generated with sufficient resolution to provide many bits of accuracy in measuring the phase of the incoming packets relative to the output sample rate. This phase measurement can then be low pass filtered and used as an input by the system to adjust the oscillator frequency of a phase locked loop. In this regard, the system can provide a stable output frequency that matches and tracks the average frequency of the samples in the control module.

[0025]

[0032] In various embodiments, each speaker can send a "heartbeat" packet to the transmitting unit at a low frequency, such as 5 Hz, to allow the control module to identify which speakers are active. The heartbeat packet can include information about the speaker, such as the speaker's placement (e.g., front right, center, rear left, subwoofer, etc.), its specific channel number, and / or IP address. The control module can monitor the various heartbeat packets via a timeout process to determine which of the multiple speakers is currently active and available to play audio. The control module can provide this information to the user via an app native to the user's device.

[0026]

[0033] When all audio sources are idle and / or the control module is powered down, the control module can send a command to each of the multiple speakers to enter a sleep mode. In sleep mode, the speakers reduce their power consumption from operating power to a low power consumption. While in sleep mode, the speakers only need to periodically monitor the transport channel to determine if the transmitter is commanding the speaker to wake up again for use.

[0027]

[0034] In various embodiments, and with reference to FIGURE 1, an exemplary system 100 for wireless surround sound is shown. The system 100 can include an audio / visual source (A / V source) 102, a control module 104, one or more speakers (e.g., multiple wireless speakers 108), and a user device 112. The speakers 108 include at least one main speaker 116 (e.g., a front speaker) and a secondary speaker 118, such as a subwoofer or rear speaker. The speakers 108 are described in more detail below with reference to FIGURE 3.

[0028]

[0035] In various embodiments, the control module 104 can be configured as a central network element or hub for accessing the various systems, engines, and components of the system 100. The control module 104 can be a computer-based system and / or a software component configured to provide an access point to the various systems, engines, and components of the system 100. The control module 104 can communicate with the A / V source 102 via a first interface 106. The control module can communicate with the speaker 108 via a second interface 110. In various embodiments, the control module 104 can communicate with the speaker 108 via a fourth interface 120. The control module 104 can communicate with the speaker 108 via the second interface 110 and the fourth interface 120 simultaneously. The control module 104 can communicate with the user device 112 via a third interface 114. In this regard, the control module 104 may enable communication from the user device 112 to various systems, engines, and components of the system 100 (such as, for example, the speakers 108 and / or the A / V sources 102). In this regard, the system may transmit high quality audio signals along with data (such as, for example, command and control signals) to any type or number of speakers configured to communicate with the control module 104.

[0029]

[0036] In various embodiments, the first interface 106 may be an audio and / or visual interface, such as, for example, a High-Definition Multimedia Interface (HDMI), DisplayPort, USB-C, AES3, AES47, S / PDIF, BLUETOOTH, etc. In various embodiments, any of the first interface 106, the second interface 110, and / or the third interface 114 may be a wireless data interface, such as, for example, an interface operating on a physical layer protocol, such as, for example, IEEE 802.11, IEEE 802.15, BLUETOOTH, etc. In various embodiments, the fourth interface 120 may be a Powerline Communication (PLC) type interface configured to carry audio data. As described in more detail below, each of the various systems, engines, and components of the system 100 may also be configured to communicate via the GRAVITY™ Standardized Communication Protocol for Wireless Devices (SCP). As described in more detail below, it operates over physical layer protocols and is offered by Fasetto, Inc. of Scottsdale, Arizona.

[0030]

[0037] In various embodiments, the user device 112 can comprise software and / or hardware for communicating with the system 100 via the third interface 114, which comprises hardware and / or software configured to allow a user or the like to access the control module 104. The user device can comprise any suitable device configured to allow a user to communicate via the third interface 114 and the system 100. The user device can include, for example, a personal computer, a personal digital assistant, a mobile phone, a remote control device, or the like, that allows a user to send instructions to the system 100. In various embodiments, the user device 112 described herein can run web applications or native applications to communicate with the control module 104. The native applications can be installed on the user device 112, for example, via download, physical media, or an app store. A native application utilizes a development code base provided for use with the operating system of the user device 112 and may execute system calls to manipulate data stored or displayed on the user device 112 and to communicate with the control module 104. A web application is compatible with and can be written expressly to run on a web browser, i.e., a web application may be a browser-based application that operates with the system 100.

[0031]

[0038] In various embodiments, and with further reference to Figure 2, there is shown a control module 104. The control module 104 may include a controller 200, an A / V receiver 202, a transcoding module 204, an effects processing module (FX module) 206, a user device interface 208, a speaker interface 210 (e.g., a transmitter or transceiver), a power supply 212, and a power line communication modulator / demodulator (PLC modem) 214.

[0032]

[0039] In various embodiments, the controller 200 may comprise a processor and may be configured as a central network element or hub for accessing various systems, engines, and components of the system 100. In various embodiments, the controller 200 may be implemented in a single processor. In various embodiments, the controller 200 may be implemented as or may include one or more processors and / or one or more tangible, non-transitory memories and may implement logic. Each processor may be a general-purpose processor, a digital signal processor (DSP), an application specific integrated circuit (ASIC), a field programmable gate array (FPGA), or other programmable logic device, discrete gate or transistor logic, discrete hardware components, or any combination thereof. The controller 200 may comprise a processor configured to perform various logical operations in response to execution of instructions, e.g., in response to instructions stored on a non-transitory, tangible, computer-readable medium configured to communicate with the controller 200.

[0033]

[0040] The system program instructions and / or the controller instructions can be loaded onto a non-transitory, tangible computer-readable medium, which stores instructions that, in response to execution by the controller, cause the controller to perform various operations. It is commonly understood that the term "non-transitory" excludes only transitory propagating transitory signals, per se, from the claims, and does not waive the right to all standard computer-readable media that are not merely transitory propagating signals, per se. In other words, the meaning of the terms "non-transitory computer-readable medium" and "non-transitory computer-readable storage medium" should be interpreted to exclude only the types of transitory computer-readable media that Re Nuijten found not to fall within the scope of patentable subject matter under 35 U.S.C. § 101.

[0034]

[0041] In various embodiments, the A / V receiver 202 is configured to receive source audio data from the A / V source 102 via the first interface 106. The controller 200 can pass the source audio data to the transcoding module 204 for further processing. In various embodiments, the transcoding module 204 is configured to perform a conversion operation between a first encoding and a second encoding. For example, the transcoding module 204 can convert the source audio from a first encoding to a second encoding and generate transcoded audio data for further processing by the FX module 206. In various embodiments, the transcoding module 204 can be configured to decode and / or transcode one or more channels of audio information contained within the source audio data, such as information encoded as Dolby Digital, DTS, ATMOS, Sony Dynamic Digital Sound (SDDS), etc. In this regard, the transcoding module 204 can generate transcoded audio data including multiple channels of audio information, which can then be further processed by the system.

[0035]

[0042] In various embodiments, the FX module 206 may comprise one or more digital signal processing (DSP) elements or may be configured to adjust the balance between frequency components of the transcoded audio data. In this regard, the FX module 206 may act as an equalization module to boost or attenuate the energy of one or more frequency bands present in the transcoded audio data. In various embodiments, the FX module 206 may also include one or more filtering elements, such as, for example, a bandpass filter configured to eliminate or reduce undesired and / or unwanted elements of the source audio data. Similarly, the FX module may include one or more effect elements and / or effect functions configured to modify the transcoded audio data. For example, the effect functions can enhance the data quality of the transcoded audio data, compensate for room modes, apply distortion effects, dynamic effects, modulation, pitch / frequency shifting, time-based, feedback, sustain, equalization, and / or other effects. In various embodiments, the FX module can be software defined and / or configured to receive over-the-air updates. In this regard, the system can allow for the loading of new and / or user-defined effect functions. In various embodiments, the FX module 206 can be configured to apply any number of effect functions to the transcoded audio data to generate audio data including an audio information channel and a desired effect.In various embodiments, the FX module 206 can also resample the audio stream to change the data rate. The controller 200 can pass the effected audio data to a speaker interface 210.

[0036]

[0043] In various embodiments, the DSP functionality of the FX module resides entirely within the control module 104, with no additional processing taking place in the speakers. In various embodiments, and as discussed briefly below with reference also to FIGURE 3, the functionality of the FX module 206 may be incorporated by the DSP 306 of each of the plurality of speakers 300. In this regard, by implementing software-defined FX module functionality through an integrated DSP in one or more of the plurality of speakers 300, the control module 104 may be smaller and less complex.

[0037]

[0044] In various embodiments, the speaker interface 210 can be configured to communicate with the multiple speakers 108 via the second interface 110. In various embodiments, the speaker interface 210 can include multiple communication channels, each communication channel being associated with one of the multiple speakers 108. The controller 200 can assign each of the channels of audio information to the multiple speakers 108. For example, the speaker interface 210 can assign a first channel of effected audio data to a communication channel for the primary speaker 116 and a second channel of effected audio data to a communication channel for the secondary speaker 118. In this regard, the system can assign the multiple channels of audio information to the multiple speakers on a one-to-one basis. The speaker interface 210 can thereby facilitate streaming of the audio information by the processor to the speakers of the various channels. In various embodiments, the speaker interface 210 may also be configured to distribute instructions (e.g., control commands) to the speakers.

[0038]

[0045] In various embodiments, the speaker interface 210 can include a PLC modem 214. In this regard, the speaker interface 210 can be configured to communicate with multiple speakers 108 via the fourth interface 120. The speaker interface 210 can be configured to distribute only control commands via the second interface 110 and only audio information via the fourth interface 120. In various embodiments, the speaker interface can also be configured to distribute all control commands and audio data only via the second interface 110 or only via the fourth interface 120.

[0039]

[0046] In various embodiments, the user device interface 208 is configured to enable communication between the controller 200 and the user device 112 via the third interface 114. The user device interface 208 can be configured to receive control commands from the user device 112. The user device interface 208 can be configured to return command confirmations or other data to the user device 112. For example, the user device interface 208 can be configured to return performance information about the control module 104, effected audio data, the status of the speaker interface 210, the performance or status of the speaker 108, etc. In various embodiments, the user device interface 208 can be further configured to receive source audio data from the user device 112.

[0040]

[0047] In various embodiments, the power source 212 is configured to receive electrical power. Further, the power source 212 can be configured to distribute the received electrical power to the various components of the system 100.

[0041]

[0048] In various embodiments, and with further reference to FIG. 3, an exemplary speaker 300 is shown from among the plurality of speakers 108. The speaker 300 includes a power source 302 configured to receive power and distribute the power to various components of the speaker 300. The speaker 300 may further include a transceiver 304, a DSP 306, an amplifier 308, and a speaker driver 310. In various embodiments, the transceiver 304 is configured to receive assigned channels of audio information and control commands from the control module 104 via the second interface 110. In various embodiments, the transceiver may further be configured to pass status information and other data about the speaker 300 to the control module 104. In various embodiments, the transceiver 304 may also be configured to communicate directly with the user device 112.

[0042]

[0049] In various embodiments, the DSP 306 can be configured to receive the assigned audio channels and apply one or more digital signal processing functions, such as, for example, sound effect algorithms, to the audio data. In this regard, the DSP 306 can also perform further effect functions on the audio data already processed by the FX module 206. In various embodiments, the DSP 306 can perform further processing in response to commands from the control module 104. For example, the control module can instruct the DSP to apply a processing function that equalizes the output of the speaker 300 based on its particular location in the room, a processing function that emulates a desired room profile, a processing function that adds one or more effects (e.g., reverb, echo, gate, flange, chorus, etc.), etc. As discussed above, in various embodiments, the DSP 306 can include and implement all of the functionality of the FX module 206. All of the functionality can be defined in software. In this regard, the DSP 306 can generate DSP audio channels and pass the DSP audio channels to an amplifier 308 for further processing. The amplifier 308 can receive the DSP audio channels and amplify the signal strength of the DSP audio channels to generate a drive signal that can be passed to a speaker driver 310. In various embodiments, the speaker driver 310 can receive the drive signal from the amplifier 308 and, in response, convert the drive signal 310 into sound.

[0043]

[0050] As discussed above, and with further reference to FIG. 4, a schematic diagram of a data control scheme for wireless surround sound is shown. In various embodiments, each of the user device 112, the A / V source 102, the control module 104, and the speaker 108 can be further configured to communicate via an SCP. In various embodiments, the SCP can comprise a network layer protocol. In various embodiments, the system can prepend an SCP header 404 to the packet or datagram 400. In this regard, the SCP header can be inserted between the physical layer communication protocol 402 (e.g., 802.11, 802.15, etc.) data and the transport layer protocol 406 (e.g., TCP / IP, UDP, DCCP, etc.) data. Elements of the system 100 can be configured to recognize the SCP header 404 and identify an associated SCP message 408. The system can then perform various actions or instructions based on the SCP message 408.

[0044]

[0051] For example, the SCP may define the ability of devices (such as the speaker 108, the control module 104, and the user device 112) to discover each other, request transfer of raw data, send confirmations upon receipt of data, and perform steps involved in data transmission. The SCP may define various control commands for the speaker 300 to switch or apply various DSP functions, to turn the power 302 on or off, to affect the signal strength output by the amplifier 308, etc. In various embodiments, the SCP may define the ability of the control module 104 to change the effect functions of the FX module 206 and / or the DSP 306, to select a code for the transcode module 204, to select audio source data, to turn the power 212 on or off, to assign or change the interface of the speaker interface 210, etc. In this regard, when implemented in the system 100, the SCP allows for real-time individual control of each of the multiple speakers 300 to deploy audio signal processing functions to selected individual speakers (e.g., the main speaker 116) or to a collection of speakers (e.g., the main speaker 116 and the secondary speakers 118), such as frequency shaping, dialogue-enhancement, room mode correction, effects functions, equalization functions, tone control, balance, level and volume control, etc. Thus, the system 100 allows for individual control of the sound output characteristics of the speakers 300.

[0045]

[0052] 5, an exemplary speaker 500 of the plurality of speakers 108 is shown. The speaker 500 has similar features, configuration, structure, materials, manufacturing techniques, and / or internal components as the speaker 300, but includes a PLC modem 512. The power supply 502 of the speaker 500 is configured to receive power and distribute the power to the various components of the speaker 500. In various embodiments, the PLC modem 512 may also comprise a module of the power supply 502. The speaker 500 may also include a transceiver 504, a DSP 506, an amplifier 508, and a speaker driver 510. In various embodiments, the transceiver 504 is configured to receive control commands from the control module 104 via the second interface 110.

[0046]

[0053] The PLC modem 512 can be configured to receive audio information via the fourth interface 120, such as, for example, multiple channels of audio information that can be broadcast from the control module 104 to the speaker 500. In various embodiments, the control command can include instructions directed to the PLC modem, such as a channel selection, regarding an assigned audio information channel. The PLC modem 512 can be configured to filter out the assigned audio information channel from the multiple audio information channels based on the channel selection. In various embodiments, the transceiver 504 can further be configured to pass status information and other data about the speaker 500 to the control module 104. In various embodiments, the transceiver 504 can also communicate directly with the user device 112.

[0047]

[0054] 6, a process 600 for streaming audio data in the system 100 is shown in accordance with various embodiments. The system may receive audio source data 602, such as an HDMI source, via the first interface 106. The audio source data 602 may be encrypted. The system may decrypt the audio source data (step 604), such as with a decryption module or algorithm (e.g., an HDMI decoder using an HDCP key). In response, the system may generate one or more decrypted data streams (608). For example, the audio source data may be 8-channel audio source data and the output of the HDMI decoder may be an 8-channel parallel I2S data stream.

[0048]

[0055] The system applies a first digital signal processing (DSP) algorithm to the decrypted data stream (step 610). For example, the system may apply DOLBY ATMOS™ processing. This processing may generate up to 34 channels of audio from the 8 channels of data decoded in step 604. In response to applying the first DSP algorithm, the system may generate a plurality of audio data channels 612. In various embodiments, the system may apply additional DSP algorithms (e.g., a second DSP algorithm, a third DSP algorithm, ..., an nth DSP algorithm) to the plurality of audio data channels (step 614). Further processing in step 614 may include volume, equalization, or other effects as desired. Effects may be applied to all channels or may be applied to each channel separately, either individually or by group. In various embodiments, each channel of the plurality of audio data channels may be processed by a channel-specific DSP algorithm (i.e., an algorithm assigned to each of the plurality of audio data channels). In this regard, the system can generate post-processed audio data (e.g., effects applied data) 616. The post-processed audio data can comprise multiple audio streams associated one-to-one with each speaker in the multiple speakers. In various embodiments, additional processing can be applied to convert the sample rate to match the sample rate provided by the time base signal generated by time base generation process 628, as described below.

[0049]

[0056] In various embodiments, the post-processed audio data 616 may be encoded to generate encoded post-processed audio data 620 for transmission to multiple speakers (step 618). For example, the post-processed audio data 616 may be encoded as Ethernet data. The audio stream may be loaded into packets and sent at a rate determined by a sample rate set by a time base signal. The encoded post-processed audio data 620 may be passed to a fourth interface for transmission (step 622). For example, the Ethernet encoded data may be passed to an Ethernet-type PLC modem implementing a protocol such as H.Gn and coupled to a power cable 624.

[0050]

[0057] In various embodiments, the system can receive a power signal 626 from a power cable 624. The power signal can be a 50 Hz or an AC signal between 50 Hz and 60 Hz, or other signal modulated onto the power cable 624. A process 628 can generate a time base signal based on the power signal. In various embodiments, the process 628 can include a phase-locked loop circuit to generate a relatively jitter-free reference frequency at a desired sample rate and multiples thereof. For example, the process 628 can generate a 48 kHz sample rate and a 256×48 kHz, or 12.288 MHz, reference clock or time base signal 630. The reference clock or time base signal 630 can be used to drive digital signal processing and to drive various systems and modules of the system 100 and the process 600.

[0051]

[0058] At each speaker, a PLC modem 632 can receive the encoded data and extract the Ethernet packets 634. The PLC modem 632 can pass the Ethernet packets 634 to the speaker's processor 636 for further processing. The processor 636 can only accept packets that are addressed to a corresponding speaker 642. The processor 636 can reconstruct the packets to generate audio data 638 and pass the audio data 638 to a digital input audio amplifier 640 configured to drive the speaker 642. In various embodiments, the audio data can be passed over I2S.

[0052]

[0059] Benefits, other advantages, and solutions to problems have been described herein with reference to specific embodiments. Additionally, the connecting lines shown in the various figures contained herein are intended to represent representative functional relationships and / or physical connections between the various elements. It should be noted that many alternative or additional functional relationships or physical connections may exist in an actual system. However, benefits, advantages, solutions to problems, and elements that may cause or enhance any benefit, advantage, or solution should not be construed as essential, required, or essential features or elements of the disclosure.

[0053]

[0060] The scope of the present disclosure is therefore not to be limited except by the appended claims. In the claims, reference to an element in the singular is intended to mean "one or more" and not to mean "one and only one" unless specifically stated. Furthermore, when phrases similar to "at least one of A, B, or C" are used in the claims, this is intended to be interpreted to mean that in some embodiments only A may be present, in some embodiments only B may be present, in some embodiments only C may be present, or in an embodiment any combination of elements A, B, and C may be present, e.g., A and B, A and C, B and C, or A and B and C. Different shading is used throughout the figures to indicate different parts, but does not necessarily indicate that the materials are the same or different.

[0054]

[0061] In the present specification, systems, methods, and devices are provided. When referring to "one embodiment," "an embodiment," "example embodiment," and the like in the detailed description of the present specification, it is intended to indicate that the described embodiment includes a particular feature, structure, or characteristic, but that not all embodiments necessarily include the particular feature, structure, or characteristic. Moreover, such phrases do not necessarily refer to the same embodiment. Furthermore, when a particular feature, structure, or characteristic is described in connection with an embodiment, it is submitted that it is within the knowledge of a person skilled in the art to implement such feature, structure, or characteristic in connection with other embodiments, whether or not explicitly described. After reading the description, it will be apparent to a person skilled in the art how to implement the present disclosure in alternative embodiments.

[0055]

[0062] Furthermore, no element, component, or method step in this disclosure is intended to be dedicated to the public, regardless of whether that element, component, or method step is expressly recited in a claim. No claim element is intended to invoke 35 U.S.C. 112(f) unless the element is expressly recited using the phrase "means for." As used herein, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion such that a process, method, article, or apparatus making up a list of elements may include not only those elements, but also other elements not expressly recited or inherent to such process, method, article, or apparatus.

Claims

1. 1. A system comprising: one or more processors; one or more non-transitory computer-readable storage devices for storing computing instructions; Equipped with The computing instructions execute on the one or more processors and cause the one or more processors to: receiving audio source data at a speaker; applying a digital signal processing algorithm to the audio source data on the speaker to produce post-processed audio data; encoding the post-processed audio data on the speaker; outputting the post-processed audio data as encoded through the speaker; A system configured to cause

2. 2. The system of claim 1, wherein the audio source data is organized into packets, the packets comprising: (1) a physical layer communication protocol unit; (2) a standardized communication protocol header section that follows this; (3) a transport layer protocol section following this; (4) a standardized communication protocol message portion; Including, the system.

3. 2. The system of claim 1, wherein the act of encoding the post-processed audio data comprises: splitting the post-processed audio data into at least two different audio data channels; adjusting a balance between frequency components of the at least two different audio data channels; Including, the system.

4. 4. The system of claim 3, wherein the balancing operation comprises: The system includes operations for applying one or more of an equalization effect and a filtering element.

5. 4. The system of claim 3, the speaker includes a plurality of speakers; transmitting the post-processed audio data as encoded, transmitting a first audio data channel of the at least two different audio data channels to a first speaker of the plurality of speakers; transmitting a second audio data channel of the at least two different audio data channels to a second speaker of the plurality of speakers, the second speaker being different from the first speaker of the plurality of speakers; Including, the system.

6. 10. The system of claim 1, The computing instructions further execute on the one or more processors and cause the processors to: receiving an AC signal from a power cable; generating a time base signal using the AC signal; Execute the operation of applying the digital signal processing algorithm comprising: applying the digital signal processing algorithm to the audio source data and the time-based signal to create the post-processed audio data.

7. 7. The system of claim 6, wherein the act of generating the time base signal comprises: The system includes the operation of generating the time base signal using the AC signal and a phase-locked loop circuit.

8. 7. The system of claim 6, wherein the time base signal comprises a jitter-free reference frequency at a predetermined sample rate.

9. 10. The system of claim 1, wherein the computing instructions further execute on the one or more processors and cause the processors to: The system performs the operation of applying a loss mitigation method to the audio source data after receiving the audio source data at the speaker.

10. 10. The system of claim 9, wherein the loss mitigation method comprises one or more of: (1) a packet interpolation method; (2) a spectral analysis method; (3) a packet switching method using volumetric data; and (4) a packet switching method using lossy compressed packets.

11. 1. A method implemented by execution of computing instructions, the computing instructions configured to execute on one or more processors and further configured to be stored on a non-transitory computer-readable medium, the method comprising: receiving audio source data at a speaker; applying a digital signal processing algorithm to the audio source data on the speaker to produce post-processed audio data; encoding the post-processed audio data on the speaker; outputting the encoded post-processed audio data through the speaker; A method comprising:

12. 12. The method of claim 11, wherein the audio source data is organized into packets, the packets comprising: (1) a physical layer communication protocol unit; (2) a standardized communication protocol header section that follows this; (3) a transport layer protocol section following this; (4) a standardized communication protocol message portion; A method comprising:

13. 12. The method of claim 11, wherein the step of encoding the post-processed audio data comprises: splitting the post-processed audio data into at least two different audio data channels; adjusting a balance between frequency components of the at least two different audio data channels; A method comprising:

14. 14. The method of claim 13, wherein the step of adjusting the balance comprises: A method comprising applying one or more of an equalization effect and a filtering element.

15. 14. The method of claim 13, the speaker includes a plurality of speakers; transmitting the post-processed audio data as encoded, transmitting a first audio data channel of the at least two different audio data channels to a first speaker of the plurality of speakers; transmitting a second audio data channel of the at least two different audio data channels to a second speaker of the plurality of speakers, the second speaker being different from the first speaker of the plurality of speakers; A method comprising:

16. 12. The method of claim 11, further comprising: receiving an AC signal from a power cable; generating a time base signal using the AC signal; Including, applying the digital signal processing algorithm, applying said digital signal processing algorithm to said audio source data and said time-based signal to produce said post-processed audio data.

17. 17. The method of claim 16, wherein generating the time base signal comprises: The method includes generating the time base signal using the AC signal and a phase-locked loop circuit.

18. 17. The method of claim 16, wherein the time base signal comprises a jitter-free reference frequency at a predetermined sample rate.

19. 12. The method of claim 11, further comprising: A method comprising applying a loss mitigation method to the audio source data after receiving the audio source data at the speaker.

20. 20. The method of claim 19, wherein the defect mitigation method comprises one or more of: (1) a packet interpolation method; (2) a spectral analysis method; (3) a packet switching method using volume data; and (4) a packet switching method using lossy compressed packets.

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