WO2019136076A1 - Synchronisation et séquencement audio basés sur un logiciel - Google Patents

Synchronisation et séquencement audio basés sur un logiciel Download PDF

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Publication number
WO2019136076A1
WO2019136076A1 PCT/US2019/012063 US2019012063W WO2019136076A1 WO 2019136076 A1 WO2019136076 A1 WO 2019136076A1 US 2019012063 W US2019012063 W US 2019012063W WO 2019136076 A1 WO2019136076 A1 WO 2019136076A1
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WIPO (PCT)
Prior art keywords
audio
clock
frequency
wall time
timing
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PCT/US2019/012063
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English (en)
Inventor
Kenneth A. Boehlke
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Summit Wireless Technologies, Inc.
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Priority claimed from US15/863,637 external-priority patent/US10582461B2/en
Application filed by Summit Wireless Technologies, Inc. filed Critical Summit Wireless Technologies, Inc.
Publication of WO2019136076A1 publication Critical patent/WO2019136076A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/61—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio
    • H04L65/612—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio for unicast
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/20—Servers specifically adapted for the distribution of content, e.g. VOD servers; Operations thereof
    • H04N21/23—Processing of content or additional data; Elementary server operations; Server middleware
    • H04N21/242—Synchronization processes, e.g. processing of PCR [Program Clock References]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/4302—Content synchronisation processes, e.g. decoder synchronisation
    • H04N21/4305—Synchronising client clock from received content stream, e.g. locking decoder clock with encoder clock, extraction of the PCR packets
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W56/00—Synchronisation arrangements
    • H04W56/001—Synchronization between nodes
    • H04W56/0015—Synchronization between nodes one node acting as a reference for the others
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W56/00—Synchronisation arrangements
    • H04W56/004—Synchronisation arrangements compensating for timing error of reception due to propagation delay
    • H04W56/005—Synchronisation arrangements compensating for timing error of reception due to propagation delay compensating for timing error by adjustment in the receiver
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04J—MULTIPLEX COMMUNICATION
    • H04J3/00—Time-division multiplex systems
    • H04J3/02—Details
    • H04J3/06—Synchronising arrangements
    • H04J3/062—Synchronisation of signals having the same nominal but fluctuating bit rates, e.g. using buffers
    • H04J3/0632—Synchronisation of packets and cells, e.g. transmission of voice via a packet network, circuit emulation service [CES]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04J—MULTIPLEX COMMUNICATION
    • H04J3/00—Time-division multiplex systems
    • H04J3/02—Details
    • H04J3/06—Synchronising arrangements
    • H04J3/0635—Clock or time synchronisation in a network
    • H04J3/0638—Clock or time synchronisation among nodes; Internode synchronisation
    • H04J3/0658—Clock or time synchronisation among packet nodes
    • H04J3/0661—Clock or time synchronisation among packet nodes using timestamps
    • H04J3/0667—Bidirectional timestamps, e.g. NTP or PTP for compensation of clock drift and for compensation of propagation delays
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00—Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/005—Audio distribution systems for home, i.e. multi-room use

Definitions

  • the present invention again relates to wireless data networks and, more particularly, to a system and method for synchronizing outputs at multiple endpoints in a network which includes a wireless communication link.
  • Video and uncompressed audio may be streamed from an audio/video source in a media room or closet to a display and multiple speakers of a surround sound system in a remote room or rooms in a residence. Since it is difficult to retrofit finished structures with cabling, in many cases data, including video and audio data, is transmitted from a source to a display, speakers or other output devices over a network that includes a wireless communication link(s) utilizing low cost radio technologies such as frequency modulation and spread spectrum modulation to transport packetized digital data.
  • Synchronization of outputs and minimization of system latency are critical requirements for high quality audio whether or not combined with video.
  • the human ear is sensitive to phase delay or channel-to-channel latency and multi-channel audio output with channel-to-channel latency greater than 50 microsecond (ps) is commonly described as disjointed or blurry sound.
  • source-to-output delay or latency (“lip-sync") greater than 50 milliseconds (ms) is commonly considered to be noticeable in audio-video systems.
  • a source of digital data transmits a stream of data packets to the network's end points where the data is presented.
  • a pair of clocks at each node of the network controls the time at which a particular datum is presented and the rate at which data is processed, for examples, an analog signal is digitized or digital data is converted to an analog signal for presentation.
  • the actual or real time that an activity, such as presentation of a video datum, is to occur is determined by "wall time," the output of a "wall clock” at the node.
  • a sample or media clock controls the rate at which data is processed, for example, the rate at which blocks of digital audio data introduced to a digital to analog converter.
  • Audio video bridging is the common name of a set of technical standards developed by the Institute of Electrical and Electronics Engineers (IEEE) and providing specifications directed to time-synchronized, low latency, streaming services over networks.
  • IEEE Institute of Electrical and Electronics Engineers
  • the Precision Time Protocol (PTP) specified by "IEEE Standard for a Precision Clock Synchronization Protocol for Networked Measurement and Control Systems," IEEE Std. 1588-2008 and adopted in IEEE 802.1AS-201 1 - "IEEE Standard for Local and Metropolitan Area Networks - Timing and Synchronization for Time-Sensitive Applications in Bridged Local Area Networks" describes a system enabling distributed wall clocks to be synchronized within 1 ps over seven network hops.
  • a master clock to which the remaining distributed clocks, the slave clocks, are to be synchronized is selected either by a "best master clock” algorithm or manually.
  • the device comprising the master clock (the "master device") and the device(s) comprising the slave clock(s) (the “slave device(s)") exchange messages which include timestamps indicating the master clock's "wall time” when the respective message was either transmitted or received by the master device.
  • the slave device notes the local wall times when the respective messages were received or transmitted by it and calculates the offset of the slave clock relative to the master clock and the network delay, the time required for the messages to traverse the network from the master device to the slave device.
  • the frequency drift of the slave clock relative to the master clock can also be determined enabling the slave clock to be synchronized with the master clock by adjusting the slave clock's wall time for the offset and the network delay and adjusting the slave clock's frequency for any frequency drift relative to the master clock.
  • PTP can synchronize wall clocks of an extensive network or even plural networks, but the accuracy of PTP can be strongly influenced by the loading and exposure to interference of the wireless communication link(s).
  • An alternative to PTP for synchronizing the wall time at plural devices of a wireless network is the Time Synchronization Function (TSF) specified in IEEE 802.1 1 , "IEEE
  • Every 802.1 1 compliant device in a network known as a basic service set includes a TSF counter.
  • devices of the BSS Periodically, during a beacon interval, devices of the BSS transmit a beacon frame containing a timestamp indicating the local wall time at the transmitting device and other control information.
  • a receiving node or slave device receiving the beacon frame synchronizes its local time by accepting the timing information in the beacon frame and setting its TSF counter to the value of the received timestamp if the timestamp indicates a wall time later than the node's TSF counter.
  • AVBTP Audio Video Bridging Transport Protocol
  • IEEE 1722-201 1 Layer 2 Transport Protocol for Time Sensitive Applications in a Bridged Local Area Network
  • Each data packet comprises plural application data samples, for example, audio data samples, and a time stamp indicating the wall time at which presentation of the application data in the packet is to be initiated.
  • a sample clock is generated which oscillates at a frequency that enables the plural application data samples in a data packet to be presented for processing within the time interval represented by successive timestamps.
  • PTP, TSF and AVBTP provide means for synchronizing distributed clocks
  • each network end point for example, the plural speaker units of a surround sound audio system, receives a respective aliased subsample of the timestamps and over time the clocks of the respective network endpoints will not track. What is desired, therefore, are accurate consistently synchronized sample clocks at a plurality of related network endpoints.
  • FIG. 1 is a pictorial representation an exemplary audio/video/data distribution network.
  • FIG. 2 is a graphic representation of a method of assembling and presenting application data included in data packets having AVBTP timestamps.
  • FIG. 3 is a graphical representation of the Precise Time Protocol (PTP) message exchanges between a master clock and a slave clock.
  • PTP Precise Time Protocol
  • FIG. 4 is a graphical representation of a method of sample clock recovery.
  • FIG. 5A is a block diagram of a first portion of a wireless network.
  • FIG. 5A is a block diagram of a first portion of a wireless network.
  • FIG. 5B is a block diagram of a second portion of the wireless network of FIG. 5A including a first embodiment of a sample clock recoverer.
  • FIG. 5C is a block diagram of a second portion of the wireless network of FIG. 5A including a second embodiment of a sample clock recoverer.
  • FIG. 6 is a graphical representation of a sample timing for the resampler.
  • data including video and audio data
  • a storage medium such as a digital versatile disc (DVD) by a DVD player or from a data portal 24 connected to, for example, a wide area fiber optic network or a satellite receiver, and distributed throughout the residence.
  • digital video and/or multi- channel audio may be distributed from a source 22, for example, a DVD player, located in a media room 26 or closet for presentation by video displays 28 and/or surround sound or stereo speaker units 30 in the media room, a bedroom 32 or a den 34.
  • At least part of the distribution network may comprise a radio transmitter 36 which may be part of an audio/video/data source 22 and one or more radio receiver(s) 38 which may be incorporated in the networked devices such as a computer 40, a video display 28, or the speaker units 30 of one or more a stereo or surround sound systems.
  • Synchronization of the various outputs and minimization of system latency are critical requirements of high quality audio/video systems.
  • Source-to- output delay or latency (“lip- sync") is important in audio/video systems, such as home theater systems, where a slight difference, on the order of 50 milliseconds (ms), between display of a video sequence and the output of the corresponding audio is noticeable.
  • the human ear is even more sensitive to phase delay or channel-to-channel latency between the corresponding outputs of the different channels of multi-channel audio.
  • Channel-to channel latency greater than 1 microsecond (ps) can result in the perception of disjointed or blurry audio.
  • Audio video bridging is the common name of a set of technical standards developed by the Institute of Electrical and Electronics Engineers (IEEE) and providing specifications for time-synchronized, low latency, streaming services over networks.
  • each network endpoint a network node capable of transmitting and/or receiving a data stream, includes two clocks; a "wall" clock 56,
  • sample clock 58, 80 is typically an alternating signal which controls the rate at which data is passed to a media processing device for processing.
  • sample clocks govern the rate at which an analog signal is sampled and the rate at which digital samples are to be passed to a digital-to- analog converter (DAC) controlling the emission of sound by a speaker.
  • DAC digital-to- analog converter
  • IEEE Std. 1588-2008 provides a method 100 of synchronizing the wall time at "slave" clocks 102 distributed among the nodes of a network to the wall time of the network's "master" clock 104.
  • a master clock is selected either manually or by a "best master clock” algorithm. Afterward, messages are periodically exchanged between the device comprising the master clock (the “master device") and the network devices comprising the slave clocks (the “slave devices”) enabling determination of the offset, the time by which a slave clock leads or lags the master clock, and the network delay, the time required for data packets to traverse the network. At defined intervals, by default two second intervals, the master device multicasts a Sync message 106 to the other network devices. The precise master clock wall time of the Sync message's transmission, t1 , 108 is determined and included as a timestamp in either the Sync message or in a Follow-up message 1 10.
  • the slave device determines the local wall time, t2, 1 12 at which the device received the Sync message.
  • a Delay_Req message 1 14 is then sent by the slave device to the master device at time, t3, 1 16.
  • the master clock's time of receipt, t4, 1 18 of the Delay_Req message 1 14 is determined and the master device responds with a Delay_Resp message 120 which includes a timestamp indicating t4, 1 18.
  • the slave device determines the network delay and the slave clock's offset from the four times, t1 , t2, t3 and t4:
  • Offset ((t2 - 11 ) - (t4 - 13))/2 (4)
  • each slave clock is adjusted to match the wall time of the master clock by adding or subtracting the offset to or from the local wall time and adjusting the slave clock's frequency.
  • IEEE 802.1 1 "IEEE Standard for Information Technology- Telecommunications and Information Exchange Between Systems Local and Metropolitan Area Networks" provides media access control (MAC) and physical layer (PHY) specifications for implementing wireless local area networks (WLAN) referred to basic service sets (BSS).
  • the devices which are parts of a BSS are identified by a service set identification (SSID) which may be assigned or established by the device which starts the network.
  • SSID service set identification
  • Each network device or station includes a local timing synchronization function (TSF) timer, the device's wall clock 56, 218, which is based on a 1 mega-Hertz (MHz) clock which ticks in microseconds.
  • TSF timing synchronization function
  • I BSS independent basic service set
  • Each station calculates a random delay interval and sets a delay timer scheduling transmission of a beacon when the timer expires. If a beacon arrives before the delay timer expires, the receiving station cancels its pending beacon transmission.
  • the beacon comprises a beacon frame including a timestamp indicating the TSF timer value, the wall time, of the station that transmitted the beacon.
  • the receiving station Upon receiving a beacon, if the timestamp is later than the receiving station's TSF timer the receiving station sets its TSF timer, for example the wall clock 218, to the value of the timestamp thus synchronizing the TSF timers, the wall clocks, of the transmitting station and the receiving station.
  • TSF timer for example the wall clock 218,
  • PTP and TSF are responsible for synchronizing the wall clocks of all nodes in the respective network to the same wall time but not for synchronizing the sample clocks controlling the processing of the various media transported by the network.
  • the sample clocks are recovered from the data stream at each of the network's listeners, endpoints receiving the data stream, enabling different sample clocks for different media to be transported on the same network.
  • media sources or talkers 52 embed presentation timestamps 60 in certain data packets 62 transmitted by the respective source.
  • a timestamp generator 64 controlled by the talker's sample clock 58 for the particular medium being processed, inserts a timestamp 60 into a header of a data packet 62 with the wall time from the talker's wall clock 56 as adjusted by a latency
  • the timestamp indicates the wall time at which presentation of application data included in the data packet is to be initiated.
  • Plural blocks 66 of application data 76 such as Inter-IC Sound (128) digital audio data, which are obtained from a digital data source 70 or converted from -an analog signal 72 by an analog-to-digital converter (ADC) 74 at a rate determined by the sample clock 58 are also included in each data packet 62.
  • a data block count included in the header of the data packet, specifies the number of application data blocks 66 to be presented in the interval represented by the difference between successive presentation timestamps.
  • the sample clock 56 is recovered by a sample clock recoverer 78.
  • each I2S data block contains a respective data sample 124, 126 for each of the two channels.
  • the sample or media clock recovery provides, in essence, a distributed phase locked loop (PLL) for the network with identical sample clocks generated at each listener 54 processing a respective medium.
  • PLL phase locked loop
  • this application accomplishes these functions with an Estimator 308 and a Sample Rate Converter (SRC) code 310.
  • AVB synchronizes the outputs of the network's listeners by delivering data to each end point's media interface, for examples, a controller for a video display or the digital-to-analog converters (DAC) of plural wireless speakers, at the synchronized wall times specified in the timestamps and at a rate determined by synchronic sample clocks.
  • DAC digital-to-analog converters
  • a wireless network 150 for example a network transporting video and surround sound audio, comprises plural network endpoints (stations) 52, 149, 156, 158, 160, 162A-162C.
  • One of the end points 149 includes the master clock or master TSF timer and periodically transmits a PTP message or a TSF beacon frame to facilitate synchronization of the wall clocks, for examples wall clocks 178, 218, of the other network endpoints.
  • One of the network's endpoints is a talker 52 that receives application data, for example, audio and/or video data, from a source such as a digital video disk (DVD) player or a television set-top box, and transmits the data in a packetized serial data stream 154 to a plurality of listeners 54, for examples, a wireless video display 156 and a plurality wireless speakers 158, 160, 162A-162C for multichannel surround sound.
  • Six channel surround sound audio systems known as 5.1 (“five point one") surround systems, utilize five full bandwidth channels; a front left channel, a front center channel, a front right channel and left and right surround channels, each reproduced by a corresponding speaker.
  • the 5.1 surround sound system includes one low- frequency effects channel, the point one (0.1 ) channel, which is reproduced by a subwoofer.
  • the talker 52 comprises a multiplexer/buffer (MUX/buffer) 162 which serially, packetizes digitized analog audio/video data 174 output by a coder/decoder (codec) 164 or digital audio/video data 166 obtained from a digital data source.
  • MUX/buffer multiplexer/buffer
  • codec coder/decoder
  • a clock divider 168 driven by phase locked loop (Type-ll PLL) 170 and a crystal oscillator 172, outputs a sample clock 58, an alternating signal, which times the sampling of the analog audio/video data 174 by the codec 164.
  • the sample clock 58 is also input to a timestamp generator 64 which based on wall time 178 output by the talker's wall clock 56 produces a presentation timestamp 60 indicating the wall time for initiating presentation of application data in a data packet 62 and signals the MUX to insert the presentation timestamp into the header of the data packet.
  • the sample clock 58 is also input to the MUX/buffer 162 to control the rate at which the MUX captures the data at its inputs and multiplexes the data to serial data packets containing plural application data samples, for example audio data samples.
  • the serialized data packet is buffered and transmitted from the buffer to a radio transceiver/media access controller (MAC) 180.
  • MAC radio transceiver/media access controller
  • a bus interface clock signal 182 times the transmission of the packetized data from the MUX/buffer 162 to the radio transceiver and MAC.
  • the media access controller (MAC) adds a media access address identifying the device that is to receive the data packet and the transceiver modulates the data packet with a carrier and transmits the radio frequency data stream 154 to appropriate network listeners, for examples, a receiver of the video display 156 and the respective receivers of the surround sound audio speaker units 158, 160, 162A-162C.
  • AVB also provides for transmissions to a "bridge" 184 which may relay data transmitted by the talker 50 to a second network, including listener 186, and which acts as a slave clock to the talker network's grandmaster clock and as a master clock to the network, comprising listener 186, to which it retransmits the data.
  • a "bridge" 184 which may relay data transmitted by the talker 50 to a second network, including listener 186, and which acts as a slave clock to the talker network's grandmaster clock and as a master clock to the network, comprising listener 186, to which it retransmits the data.
  • Each of the network's listeners for example, the speaker units 158,
  • one of the plural speaker units 158 is designated as the sample clock recoverer for the other speaker units 160, 162A-162C of the surround sound system.
  • the radio transceiver and MAC unit 202 of the sample clock recoverer receives the steam of data packets addressed to speaker unit 158 and transmits them to a demultiplexer/buffer 204 where the data in the data packets are
  • the presentation timestamp 60 for each data packet is transmitted to a timestamp comparer 206.
  • the time interval 63 represented by successive time stamps is signaled to a Type-ll PLL 208 by the timestamp comparer 206.
  • the number of data blocks in a data packet is input to a counter 210 in a feedback loop of the PLL 208.
  • the counter 210 in the feedback loop causes the PLL 208 to output an alternating signal, a raw recovered sample clock 79, with a respective clock edge 120 for each of the data blocks 66 included in the data packet.
  • the raw recovered sample clock 79 is input to a low bandwidth Type-ll PLL 21 1 which frequency filters the raw sample clock signal to eliminate jitter and produce a cleaner band limited recovered sample clock signal 80.
  • Band limiting the recovered sample clock signal produces a signal with a frequency centered on the mean frequency of the raw signal, substantially reducing jitter in the sample clock signal so that the sample clocks of other listeners utilizing the recovered sample clock, for example, the other surround sound speaker units 160, 162A-162C, are more nearly identical to the recovered sample clock 80.
  • the bandlimited recovered sample clock 80 is also input to a clock divider 212 which outputs a bus clock signal 214 to the buffer/DEMUX 204.
  • the recovered sample clock signal 80 is transmitted to the buffer/DEMUX 204 and to the digital-to-analog converter (DAC) 216 to control the processing rate for the audio data samples contained in the data packets.
  • the timestamp comparer 206 compares the timestamp in the data packet to the wall time of the slave clock 218 of the speaker unit and appropriately outputs a signal to the DAC when the DAC is to initiate converting the respective digital audio data, at the rate established by the recovered sample clock 80, to an analog signal 86 which controls the operation of the speaker 220.
  • the bandlimited recovered sample clock 80 is transmitted to the transceiver and MAC 202 of the sample clock recoverer 158 where it is modulated with a carrier and transmitted to other surround sound speaker units 160, 162A- 162C.
  • the rate at which the sample clock information is updated at the other speakers is set at least twice the limiting bandwidth of the recovered sample clock. For example, if the timing at a speaker is updated every 100 ms and the peak packet error rate (PER) is 75%, the low bandwidth PLL 21 1 of the sample clock recoverer 158 is set to no more than 1.25 Hz.
  • the modulated recovered sample clock signal is received by the transceiver and MAC 240 of the other speaker unit(s) where it is input to the respective buffer/DEMUX units 242 and transmitted to the DAC 244 to control the rate at which audio data samples in data packets addressed to the respective MAC by the talker 52 are processed.
  • the timestamps 60 in the data stream are separated from the application data and compared to the synchronized local wall time 246 by the timestamp comparer 248 which signals the buffer/DEMUX to input the application data from the respective data packet to the DAC 244 for presentation by the speaker.
  • the bandlimited sample clock may be recovered without frequency filtering the raw sample clock output.
  • the raw PTP and TSF wall time data 217 contained respectively in the PTP messages and
  • TSF beacon frames is separated in the buffer/DEMUX 204 of the listener 159 and
  • a frequency filtering device preferably, a low band width Type-ll PLL
  • the bandlimited wall time 223 is
  • sample clock 80 to be output by the respective PLL 209.
  • the application data in the data stream 154 from the talker is processed at a rate determined by the bandwidth limited sample clock 80 and presented at wall times determined by the synchronized local wall time 223.
  • the low bandwidth PLL 219 band limits wall time 223 which can be transmitted to
  • low bandwidth PLL 21 1 and/or the low bandwidth PLL 219 jitter is removed from the sample clock and/or the wall clock substantially reducing aliasing and improving the synchronization of the outputs of the network.
  • Estimator which measures and filters TSF value against the internal Counter/Timer of the audio subsystem processor.
  • the Estimator is presented with TSF and
  • the Frequency and Delay Coefficients are generated which then can be applied to the Counter value at any time to generate the filtered TSF value.
  • Adaptive filters are required for this application, so that the Coefficients are available immediately to be used to process audio and over time will improve in accuracy by lowering the jitter.
  • This filtered TSF value is used at the audio talker to generate the Presentation Time of the block of audio data that is transmitted.
  • the block size is set to be some multiple of the audio data interleaver length.
  • the Presentation Time and the beacon TSF values are estimated to the local audio clock by the same method used at the audio transmitter source.
  • the audio is then resampled to the local audio clock rather than use a PLL to generate the clock.
  • the frequency and delay coefficients are combined (Frequency Coefficients are multiplied and Delay Coefficients are added) to make the Play Time of each block.
  • the Play Time is used to generating the sample timing for the resampler. An example of this sample timing is shown in Fig. 6 where S x denotes sampling intervals.
  • Synchronization function (TSF) value and to generate the output audio clock (see 208 and 210).
  • TSF Synchronization function
  • SRC Sample Rate Converter
  • the TSF PLL is replaced with an Estimator (308) which measures and filters TSF value against the internal Counter/Timer of the audio subsystem processor.
  • the Estimator is presented with TSF and Counter pairs sampled at the same instant so that the frequency and delay relationship between the two can be determined.
  • This filtered TSF value is used at the audio talker to generate the Presentation Time of the block of audio data that is transmitted.
  • the block size is set to be some multiple of the audio data interleaver length.
  • the Presentation Time and the beacon TSF values are estimated to the local audio clock by the same method used at the audio transmitter source.
  • the audio is then resampled to the local audio clock, rather than to a PLL to generate the clock.
  • the frequency and delay coefficients are combined (Frequency Coefficients are multiplied, and Delay Coefficients are added) to make the Play Time of each block.
  • the Play Time is used to generating the sample timing for the resampler, an example of which is as illustrated in Fig. 6.
  • the sample spacing is assumed to be ideal, and at the beginning of each new block, the Play Time is compared to elapsed time to that point.
  • the residual error from the previous block is distributed over the next block or alternatively future blocks recursively. This allows the audio playback to start immediately without having to wait for the next PlayTime to determine the exact sample spacing.
  • One embodiment of this method of recovering audio timing in a network includes bandlimiting a raw wall time signal about a mean frequency of the raw wall time signal by estimating the frequency and delay of the raw wall time signal to a local transmitter audio clock; including timing datum in a data packet about the mean frequency of the timing signal from the estimate of the frequency and delay of the timing signal and the audio playtime at the transmitter; bandlimiting the raw wall time signal about the mean frequency of the raw wall time signal by estimating the frequency and delay of said raw wall time signal to a local receiver audio clock;
  • an audio system comprising a first listener having a wall clock maintaining a wall time for occurrence of an event, said wall time updated by data transmitted by a master clock and received by said first listener; a sample clock recoverer retrieving a sample clock from data received by said first listenerfrom a talker, said sample clock regulating a processing rate for a datum included in said data received from said talker; and a frequency filter bandlimiting said sample clock output by said sample clock recoverer, wherein said frequency filter comprises an estimator connected to receive said sample clock output by said sample clock recoverer; anda transmitter transmitting said bandlimited sample clock; and the audio system further comprises a second listener arranged to receive said transmitted bandlimited sample clock and use said bandlimited sample clock to regulate processing data received from
  • Such a system may include that first listener is a frequency filter attenuating a frequency of said updating data received from said master clock.
  • the updating data received from said master clock is a timing synchronization function datum or a precision time protocol datum.

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Synchronisation In Digital Transmission Systems (AREA)

Abstract

L'invention concerne la synchronisation de plusieurs sorties de données transportées par un réseau sans fil facilitée par la limitation de bande d'un signal d'horloge d'échantillonnage qui commande la vitesse à laquelle les données sont traitées par les dispositifs du réseau et/ou par la limitation de bande de données de temps au mur qui commande le temps réel pour présenter une donnée.
PCT/US2019/012063 2018-01-05 2019-01-02 Synchronisation et séquencement audio basés sur un logiciel WO2019136076A1 (fr)

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US15/863,637 2018-01-05
US15/863,637 US10582461B2 (en) 2014-02-21 2018-01-05 Software based audio timing and synchronization

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WO2022011397A1 (fr) * 2020-07-09 2022-01-13 Microchip Technology Incorporated Contrôleurs matériels synchronisés dans le temps, systèmes audio et ensemble de circuits associés

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US20110216660A1 (en) * 2010-03-02 2011-09-08 Jung Gun Lee Synchronization in a wireless node
US8965942B1 (en) * 2013-03-14 2015-02-24 Audience, Inc. Systems and methods for sample rate tracking
US20150245306A1 (en) * 2014-02-21 2015-08-27 Summit Semiconductor Llc Synchronization of audio channel timing

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US7103072B1 (en) * 2002-12-19 2006-09-05 Occam Networks System and method for synchronization of devices across a packet network
US20110216660A1 (en) * 2010-03-02 2011-09-08 Jung Gun Lee Synchronization in a wireless node
US8965942B1 (en) * 2013-03-14 2015-02-24 Audience, Inc. Systems and methods for sample rate tracking
US20150245306A1 (en) * 2014-02-21 2015-08-27 Summit Semiconductor Llc Synchronization of audio channel timing
US20170325185A1 (en) * 2014-02-21 2017-11-09 Summit Semiconductor Llc Synchronization of audio channel timing

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2022011397A1 (fr) * 2020-07-09 2022-01-13 Microchip Technology Incorporated Contrôleurs matériels synchronisés dans le temps, systèmes audio et ensemble de circuits associés

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