WO2013018139A1 - Speaker system, audio transmission device, speaker, audio transmission method and program - Google Patents

Speaker system, audio transmission device, speaker, audio transmission method and program Download PDF

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Publication number
WO2013018139A1
WO2013018139A1 PCT/JP2011/004383 JP2011004383W WO2013018139A1 WO 2013018139 A1 WO2013018139 A1 WO 2013018139A1 JP 2011004383 W JP2011004383 W JP 2011004383W WO 2013018139 A1 WO2013018139 A1 WO 2013018139A1
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WO
WIPO (PCT)
Prior art keywords
speaker
data
audio
transmission
unit
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PCT/JP2011/004383
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French (fr)
Japanese (ja)
Inventor
浩一郎 角谷
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パイオニア株式会社
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Priority to PCT/JP2011/004383 priority Critical patent/WO2013018139A1/en
Publication of WO2013018139A1 publication Critical patent/WO2013018139A1/en

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/02—Spatial or constructional arrangements of loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels

Definitions

  • the present invention provides a speaker system, a sound transmission device, a speaker, and a plurality of speakers connected in series via a single transmission line, and a sound transmission device capable of transmitting sound data to each speaker via the transmission line.
  • the present invention relates to an audio transmission method and program.
  • a daisy chain speaker system that transmits audio data to a plurality of speakers connected in a daisy chain (series connection by a single transmission line, daisy chain connection) (for example, Patent Document 1 and Patent Document 2).
  • This daisy chain speaker system can reduce the number of transmission paths compared to a conventional configuration in which a plurality of speakers are individually connected to one audio (AV) amplifier as in the prior art. There are merits such that the connection work can be reduced.
  • the number of channels tends to increase year by year, but in the case of a daisy chain speaker system, speakers can be added flexibly, and there is also an advantage that it is easy to cope with the increase in number of channels.
  • the present invention provides a speaker system, an audio transmission device, a speaker, an audio transmission method, and a program capable of simultaneously reproducing a plurality of types of audio data having different sampling rates in one daisy chain speaker system. Objective.
  • the speaker system of the present invention includes a plurality of speakers connected in series via a single transmission line, an audio transmission device that transmits audio data input from a plurality of audio input sources to each speaker via the transmission line,
  • the audio transmission device is provided corresponding to each audio input source, and converts a reproduction clock of each audio input source into a constant frequency clock, and each frequency.
  • Device-side memory for buffering each audio data after frequency conversion by the converter, and each audio data read from the device-side memory are packed into one frame and transmitted with a single clock corresponding to a fixed frequency clock And a control unit.
  • the audio transmission method of the present invention is an audio transmission method for transmitting audio data input from a plurality of audio input sources to a plurality of speakers connected in series on a single transmission line via the transmission line.
  • a frequency conversion step of converting a reproduction clock of each audio input source into a fixed frequency clock by a plurality of frequency conversion units provided corresponding to each audio input source, and each audio data after frequency conversion on the device side A buffering step of buffering in a memory, and a transmission control step of packing each audio data read from the device-side memory into one frame and transmitting it with a single clock corresponding to a constant frequency clock.
  • a frequency conversion unit is provided for each audio input source, and the reproduction clock of each audio input source is converted into a constant frequency clock. Can be played simultaneously on the speaker system.
  • the audio data after frequency conversion is buffered in the device-side memory, the output timing of the audio data can be synchronized (synchronized). Further, since a plurality of audio data is packed and transmitted in one frame, transmission control can be easily performed. “Audio data” refers to sampling data input from each audio input source.
  • the frequency clock used for the A / D converter is a fixed frequency clock.
  • the “constant frequency clock” indicates that the frequency clock used for sampling the analog signal is the same as the clock after the sampling rate conversion.
  • each speaker has a data selection / acquisition unit that separates the sync data and the audio data for the speaker from the frame transmitted by the audio transmission device, and an internal clock in synchronization with the selected / acquired sync data. And a phase synchronizer for generating.
  • the internal clock is generated in synchronization with the selected and acquired sync data, so that synchronization with other speakers can be achieved.
  • each speaker further includes a speaker-side memory for buffering the selected and acquired audio data, and a counter unit for measuring the output timing of the audio data from the speaker-side memory, To do.
  • an enable flag indicating validity / invalidity of the corresponding data is defined for each audio data transmitted as a frame, and each speaker detects the enable flag “invalid” for a certain period of time.
  • the apparatus further includes a standby control unit that puts the speaker in a standby state.
  • the standby state is set (the audio amplifier, the D / A converter, the counter, the memory, etc. are suspended. Power consumption can be suppressed. However, it is assumed that only daisy chain transmission can be performed even in the standby state.
  • each speaker has a sound amplifier function.
  • each can be controlled independently, so that only the speaker of the channel that requires sound reproduction needs to be driven and the power used in the amplifier portion can be suppressed. it can.
  • the audio transmission device of the present invention is used in the above speaker system.
  • the speaker of the present invention is used in the above speaker system.
  • the program of the present invention causes a computer to execute each step in the above-described audio transmission method.
  • FIG. 1 is a system configuration diagram of a speaker system according to an embodiment of the present invention. It is a figure which shows the example of arrangement
  • FIG. 1 is a system configuration diagram of the speaker system SY.
  • the speaker system SY includes an audio transmission device 1 that functions as an AV center unit (AV receiver), a plurality of speakers 2, and a daisy chain cable 3 (transmission) for connecting the speakers 2 in series. Path, hereinafter simply referred to as “cable 3”) and an audio input source 4 for inputting audio data.
  • AV center unit AV receiver
  • daisy chain cable 3 transmission
  • path hereinafter simply referred to as “cable 3”
  • an audio input source 4 for inputting audio data.
  • the audio input source 4 is wired / wirelessly connected to the audio transmission device 1 and mainly uses content via a network (including streaming playback), a CD player, a Blu-ray player, a smartphone, a network player, and the like.
  • the audio transmission device 1 incorporates a decoder 41 (see FIG. 16 and the like) corresponding to each audio input source 4, and multi-channel audio data input from each audio input source 4 is transmitted via the cable 3 to each Transmit to the speaker 2.
  • a cable 3 in which audio data of each channel is packed is serially transmitted.
  • the audio transmission device 1 also functions as a power supply device, and supplies power to each speaker 2 via the cable 3. The supply of power will be described in the fourth embodiment.
  • As the cable 3, a general-purpose cable or the like (see FIG. 32) is used.
  • the speaker 2 has a built-in amplifier 56 (sound amplifier function, see FIG. 18 and the like), and volume control and channel selection designation are performed using the control line of the cable 3.
  • a plurality of speakers 2 are connected in series via one cable 3 (physically, the number of speakers 3 connecting between the speakers 2), and are arranged at home or in a hall.
  • FIG. 2 is a diagram illustrating an arrangement example of the speakers 2 when the speaker system SY is constructed in two rooms at home. In the example of the figure, a total of 12 speakers are arranged in the order of a part of speakers 2 arranged in the living room from the audio transmission device 1, a speaker 2 arranged in the kitchen, and the remaining speakers 2 arranged in the living room. Are connected in a daisy chain.
  • the speaker system SY of this embodiment can introduce the speaker system SY across a plurality of rooms. Therefore, by installing only one audio transmission apparatus 1 at home, different contents can be reproduced simultaneously in a plurality of zones (reproduction areas).
  • the audio transmission device 1 is connected to a plurality of audio input sources 4 as described above, and can simultaneously reproduce audio data input from the audio input sources 4. Accordingly, it is possible to enjoy the music of the CD player in the living room and the music of the Blu-ray player in the kitchen.
  • the speaker system SY in the home it is not necessary to arrange the audio transmission device 1 for each room, and the number of cables can be reduced, so that an audio viewing environment can be simply constructed.
  • the number of cables can be reduced significantly and the problem that the cable 3 becomes obstructive at the time of cleaning etc. can also be eliminated. .
  • FIG. 3A is a rear view of the audio transmission device 1.
  • a power supply port 11 that receives power supply from an AC power supply and one output terminal 12 for performing daisy chain output are provided.
  • the output terminal 12 has a simple configuration, thereby reducing the size of the housing. Can do.
  • FIG. 3B is a front view and a rear view of the speaker 2.
  • a vibration member 21 is provided on the front surface of the speaker 2.
  • one input terminal 22 and one output terminal 23 for performing daisy chain input / output are provided.
  • the rear configuration of all speakers 2 connected in a daisy chain is the same.
  • the audio transmission device 1 includes an ID initialization unit 101, a speaker assignment setting unit 102, an audio input unit 103, a channel information acquisition unit 104, a channel information setting unit 105, and a transmission control unit 106 as main functional configurations.
  • the ID initialization unit 101 performs initial setting for recognizing the configuration of the speakers 2 connected in a daisy chain.
  • the ID initialization unit 101 issues a unique speaker ID to each speaker 2 and stores the issued speaker ID in an internal memory (not shown) according to the issue order. Details will be described later with reference to FIG.
  • the speaker assignment setting unit 102 performs assignment processing that prompts the user to assign a channel to the speaker 2.
  • the audio transmission device 1 is provided with a display and a user interface such as an operation panel or a remote controller.
  • the corresponding speaker 2 can be assigned to each zone (multi-listening area). Details will be described later with reference to FIGS.
  • the voice input unit 103 inputs voice data (audio sampling data) from each voice input source 4. It is also possible to input a plurality of channels of audio data from each audio input source 4.
  • the channel information acquisition unit 104 acquires it.
  • the channel information is information indicating the number of content reproduction channels and channels to be thinned / not thinned.
  • the audio data to be transmitted to each speaker 2 is thinned out in the time axis direction, thereby realizing audio reproduction with an increased number of allowable transmission channels than before without increasing the transmission frequency.
  • the “channel to be thinned out” is set to, for example, “fourth to thirteenth channels” when 13 channels are realized. In this case, the fourth to thirteenth total 10 channels are divided into two groups of fourth to eighth and ninth to thirteenth, and audio data is transmitted alternately. As a result, 13-channel content reproduction is realized with a data transmission amount for a maximum of 8 channels.
  • the channel information setting unit 105 sets channel information (thinning target / non-thinning target channel) using a user interface provided in the audio transmission device 1.
  • the channel information acquired by the channel information acquisition unit 104 may be prioritized.
  • the user may be able to set which is prioritized, or may be configured to prompt the user to set channel information only when channel information is not embedded in the playback content. In the latter case, the following setting method can be considered.
  • the audio transmission device 1 determines whether division transmission is necessary by comparing with the number of reproduction channels included in the content.
  • the user is prompted to select the speaker 2 that is not desired to be divided or the speaker 2 that may be divided.
  • the audio transmission device 1 determines whether or not the transmission rate is within the allowable transmission rate (the number of speakers 2 to be divided is the necessary number that can be transmitted).
  • the user is notified of the completion of the divided transmission setting.
  • the user can set the viewing environment suitable for the viewing environment by setting the channel information.
  • the transmission control unit 106 packs a plurality of audio data input from the plurality of audio input sources 4 into one frame for transmission.
  • FIG. 12 shows a case where the audio data of channels 1 to 8 is 1 frame, and channels 1 to 3 and 9 to 13 are 1 frame, and each frame configuration is transmitted alternately. Specifically, at least a part of the plurality of speakers 2 connected in a daisy chain is divided into two speaker groups, and the audio data is divided into two pieces of sampling for each divided speaker group. Are transmitted sequentially (intermittent transmission control is performed). For example, in FIG. 12, channels 4 to 8 are grouped as a first group, and channels 9 to 13 are grouped as a second group.
  • Channels 1 to 3 belong to non-divided groups that are not grouped. Audio data is always transmitted to the speakers 2 of these non-divided groups. In this non-divided group, relatively important speakers such as front L, R, and center channel, which are easy for the user to feel a difference in quality, are set.
  • the speaker 2 includes an ID storage unit 201, a data selection acquisition unit 202, a daisy chain output unit 203, a data buffer 204, an interpolation data generation unit 205, a selection circuit 206, and an audio output unit 207 as main control configurations. Yes.
  • the ID storage unit 201 stores the speaker ID issued by the ID initialization unit 101, and is realized by a nonvolatile memory.
  • the data selection / acquisition unit 202 selects and acquires a start code packet and audio data for the own speaker 2 from the frame transmitted from the audio transmission device 1. Whether the speaker 2 is used is determined by using the speaker ID stored in the ID storage unit 201.
  • the daisy chain output unit 203 performs daisy output to the next-stage speaker 2 connected downstream. That is, the frame transmitted from the upstream cable 3 is output to the downstream cable 3.
  • the data buffer 204 stores audio data for three consecutive samples in the partial buffers 204a, 204b, and 204c.
  • the interpolation data generation unit 205 adds the real data (actually transmitted audio data) stored in the preceding and subsequent partial buffers 204a and 204c, for example. Interpolation data is generated by halving.
  • the interpolation data generation unit 205 determines the lack of data based on the channel setting flag included in the start code packet of the frame. That is, when the channel setting flag indicates “invalid”, it is determined that the audio data at that timing is missing, and interpolation data is generated.
  • the selection circuit 206 selects whether to select interpolation data or real data according to the channel setting flag. When the channel setting flag indicates “invalid”, the interpolation data is selected. When the channel setting flag indicates “valid”, the real data stored in the data buffer 204 is selected.
  • the audio output unit 207 outputs real data or interpolation data based on the selection result of the selection circuit 206. In the case of the present embodiment (when transmission is performed by dividing into two groups), the speaker 2 to be divided alternately outputs real data and interpolation data.
  • the audio transmission device 1 accesses the head speaker 2 (the most upstream speaker 2) (S12).
  • the ID storage unit 201 of the speaker 2 is cleared (S14), and a speaker ID is assigned (S15).
  • daisy-out active setting is performed for the speaker 2 (S16), and the next (downstream) speaker 2 is accessed (S17).
  • S18: Yes the steps after S14 are repeated.
  • the audio transmission device 1 accesses the head speaker 2 (S31) and waits for reception of an ACK signal.
  • the ACK signal is received (S32: Yes)
  • the speaker ID of the speaker 2 is acquired (S33), and it is determined whether or not the value is an expected value (S34). That is, it is determined whether or not the speaker IDs are acquired in the stored order. If the speaker ID is not the expected value (S34: No), there is a possibility that the speaker 2 has been replaced or the order has been changed, so the daisy chain ID initialization process shown in FIG. 5 is performed ( S35).
  • FIG. 7A is a flowchart showing the zone number setting process.
  • the voice transmission device 1 first prompts the user to input the number of zones (S41). Specifically, the user is notified that input is required by display on the display or audio output. If the number of zones is input (S42: Yes), it is determined whether or not the value is valid (S43). If it is valid (S43: Yes), the set value is stored in the internal memory. Store (S44). If S42: No, input standby is performed, and if a time-out occurs (S45: Yes), error processing is performed (S46). If the time has not expired (S45: No), the process returns to S42. Furthermore, when the input zone number is an invalid value (S43: No), error processing is performed (S46).
  • FIG. 7B is a flowchart showing the setting process of the number of channels (number of speakers 2).
  • the user is prompted to input the number of channels (S51). In this case as well, the user is notified that input is required by display on the display or audio output.
  • the number of channels is input (S52: Yes)
  • FIG. 8 is a flowchart showing the speaker assignment setting process. This processing is assumed to be performed after the setting processing of FIGS. 7 (a) and 7 (b).
  • the audio transmission device 1 determines whether or not there is a speaker 2 with an ID set and an unassigned setting (S61). If it does not exist (S61: No), error processing is performed (S62). If there is a speaker 2 with an ID set and no assignment (S61: Yes), the speaker ID to be set (initially “1”) is set (S63), and the assignment loop process is started sequentially (S63). S64).
  • an identification signal (sound or indicator (LED)) is output from the speaker 2 to be set (S65), and the user is made aware of where to set the speaker 2 currently arranged.
  • the user is prompted to input an assignment (S66).
  • a zone number is input using a GUI on a display device (such as a TV monitor) connected to the audio transmission device 1.
  • the generation of the identification signal is performed by the identification signal Gen67 (see FIG. 28).
  • the speaker ID to be set is incremented (S74). If the ID is valid (S75: Yes if assignment of all speakers 2 has not been completed), S64 and thereafter. repeat. If the ID is invalid (when assignment of all speakers 2 is completed, S75: No), the speaker assignment setting process is terminated.
  • data transmission is performed using three signals as shown in FIG.
  • data of a plurality of channels is collected and data transmission is performed as one frame.
  • data of up to 32 bits for each channel is embedded between Sync_data (start code packet) and Sync_data.
  • Each channel is separated by a ch_div signal.
  • the speaker 2 corresponding to each channel captures the data of each channel every frame, measures the synchronization timing with Sync_data, and takes the synchronization timing with clk.
  • Sync_data is detected by determining a specific pattern in advance (see FIG. 15A, etc.).
  • the data protocol of the figure is an example, and other data protocols may be used.
  • the bit length is also arbitrary, and data such as 48, 64 bits may be embedded.
  • voice (audio) data serial transmission will be described with reference to FIG.
  • “S” indicates Sync_data, and each number indicates a channel number.
  • the audio data (Nomal_data) of each channel is transmitted in time series.
  • audio data for all channels (8 channels in the example in the figure) has been transmitted for all channels until the next Sync_data is sent (within 1 / fs). There is a need to.
  • each speaker 2 is driven in synchronization, so that sound field reproduction can be performed for each moment (in sampling frequency units).
  • FIG. 11 is an explanatory diagram of a channel setting flag.
  • the Sync_data includes a payload and a channel setting flag.
  • the payload includes various types of data information that follows.
  • the channel setting flag indicates validity / invalidity for each channel (speaker 2).
  • the setting (incorporation) of the channel setting flag is performed by the transmission control unit 106.
  • Speaker 2 (interpolation data generation unit 205) recognizes transmission data according to the setting of the channel setting flag and determines that interpolation is necessary (when the channel setting flag for the speaker 2 indicates “invalid”). Generate interpolation data.
  • FIG. 12 is a conceptual diagram of audio data serial transmission according to the present embodiment.
  • data transmission is intermittently performed for some channels.
  • channels 4 to 8 are divided into a first group and channels 9 to 13 are divided into a second group, and data transmission is performed alternately for each group.
  • Channels 1 to 3 belong to a non-divided group that is not a grouping target (intermittent transmission target).
  • the non-divided group + first group data is transmitted as an odd-numbered frame
  • the non-divided group + second group data is transmitted as an even-numbered frame.
  • each channel of the non-divided group and the second group reproduces real data that is audio data transmitted as a frame.
  • each channel (channels 4 to 8) of the first group is obtained by multiplying, for example, the audio data transmitted in the first frame and the audio data transmitted in the third frame by a factor of 1/2. Interpolated data is generated and output in synchronization with the audio output timing of the second frame.
  • each channel of the non-divided group and the first group reproduces real data.
  • each channel (channels 9 to 13) of the second group is obtained by multiplying, for example, the audio data transmitted in the second frame and the audio data transmitted in the fourth frame by a factor of 1/2.
  • Interpolated data is generated and output in synchronization with the audio output timing of the third frame. By repeating this, only 8 channels are always transmitted when viewed in units of frames. However, in the first group and the second group, real data and interpolated data are alternately output to constantly reproduce all 13 channels. Is possible.
  • FIG. 13 is a flowchart showing the reproduction processing according to the present embodiment.
  • the audio transmission apparatus 1 side data is alternately transmitted to each speaker group to be divided, and on the speaker 2 side, on the time axis acquired by the own speaker 2. Since interpolated data for interpolating untransmitted audio data is generated from the audio data before and after the above, the number of reproducible speakers (number of channels) can be increased without increasing the carrier frequency. Further, since it is not a method of generating interpolated data of the speaker 2 between the sampling data of the adjacent speakers 2, the sound field can be obtained even when a plurality of music reproductions are performed or when the correlation between adjacent channels is low. Thus, it is possible to transmit audio data for a number of channels while maintaining a sense of localization. In addition, it is possible to create multi-channel content that exceeds the upper limit on the number of channels determined by the standards of disk media and the like, and it is possible to meet the demand of content creators to express with more channels.
  • the channel setting flag indicating validity / invalidity is defined for each speaker 2 in Sync_data, it is possible to accurately determine whether interpolation data should be generated on the speaker 2 side.
  • a non-divided group that constantly transmits sampling data can be set, it is possible to prevent deterioration in sound quality.
  • speakers that are important for direct sound such as front speakers, as non-divided speakers, it is possible to suppress the influence of sound quality on hearing.
  • a part of all the channels is divided into two groups and the audio data is transmitted alternately.
  • it may be divided into three or more groups (N groups).
  • audio data is transmitted to each speaker 2 in the group to be divided at a rate of 1 / N times as compared with the case of constant transmission.
  • (N + 1) partial buffers are provided as buffers, and when the intermediate partial buffer is empty, interpolation data is generated using data in the preceding and subsequent partial buffers.
  • the voice transmission device 1 may be configured to determine the number of divisions (the number of N when dividing into N groups) so as to be within the allowable transmission rate.
  • the interpolation data generation method is exemplified by a method of adding 1/2 of the preceding and succeeding data, but other filtering methods may be employed. Further, the dividing method of each channel does not necessarily need to be grouped according to the connection order of the speakers. For example, a method of grouping with odd speakers and even speakers may be considered.
  • FIG. 14 is a functional block diagram of the speaker system SY according to the second embodiment.
  • the audio transmission device 1 includes an audio input unit 103, a frequency conversion unit 111, a device-side memory 112, and a transmission control unit 113 as main functional configurations.
  • the voice input unit 103 inputs voice data from each voice input source 4.
  • the frequency conversion unit 111 converts the reproduction clock of each audio input source 4 into a constant frequency clock by the SRC 43 (see FIG. 16) provided corresponding to the signal input unit of each audio input source 4. Note that the frequency converter 111 may not be provided for the analog signal input unit.
  • the device-side memory 112 buffers each audio data after frequency conversion by each frequency conversion unit 111, and corresponds to the memory 46 in FIG.
  • the transmission control unit 113 packs each piece of audio data read from the device-side memory 112 into one frame and transmits it with a single clock corresponding to a certain frequency clock. “Corresponding to a fixed frequency clock” means that the single clock is the fixed frequency clock itself or a clock obtained by multiplying the fixed frequency clock (the clock on the reading side of the SRC 43). As a result, it is possible to prevent an error in data transfer that occurs when the sampling rate is the same and the reproduction clock of each audio input source 4 is slightly different or when the sampling rate is different.
  • the transmission control unit 113 includes the daisy format Gen45 and the CPU 48 of FIG. 16 as main parts.
  • the speaker 2 includes a data selection acquisition unit 202, a daisy chain output unit 203, a speaker side memory 208, a phase synchronization unit 211, a counter unit 212, an audio output unit 207, and a standby control unit 213 as main functional configurations.
  • a data selection acquisition unit 202 and the daisy chain output unit 203 are the same as those in the first embodiment, description thereof is omitted.
  • the speaker side memory 208 buffers the audio data selected and acquired by the data selection acquisition unit 202, and corresponds to the memory 53 of FIG.
  • the phase synchronization unit 211 generates an internal clock in synchronization with Sync_data separated by the data selection / acquisition unit 202, and corresponds to the PLL 52 (Phase-locked loop) in FIG.
  • the counter unit 212 measures the output timing of audio data from the speaker-side memory 208, and corresponds to the counter 54 in FIG.
  • the audio output unit 207 outputs audio based on audio data read from the speaker-side memory 208, and corresponds to the audio output mechanism 57 in FIG.
  • the standby control unit 213 sets the speaker 2 to the standby state when detection of “invalid” of the enable flag (corresponding to the channel setting flag of the first embodiment) continues for a certain time.
  • the enable flag is a flag set corresponding to the audio data of each channel, and indicates the validity / invalidity of the corresponding data. Therefore, when the detection of the enable flag “invalid” continues for a certain period of time, it can be determined that the speaker 2 is not used. Therefore, the power consumption can be suppressed by setting the standby state (pausing the amplifier 56 and the like). . However, even when the speaker 2 is in a standby state, only daisy chain transmission can be performed.
  • the Sync_data of this embodiment sets the header (0th to 7th bits) to “10111111”, and determines whether it is Sync_data by detecting the header.
  • the ninth bit defines reset. When the reset is “1: ON”, each speaker 2 performs a reset process.
  • FIG. (B) in the figure shows an assignment example of the Normal_data pack.
  • an enable (enable flag) is defined at the 0th bit. “Enable” indicates whether or not the corresponding data is valid, and each speaker 2 performs standby processing when the detection of the enable “0: invalid” continues for a predetermined time. Further, even if audio data is transmitted to all the speakers 2 on the daisy chain, if there is time remaining until the next sync timing, the remaining Normal_data pack enable flags are set to “0: invalid”.
  • an example of a maximum of 32 bits is shown, but the number of bits per channel is not limited.
  • FIG. 16 is a control block diagram of the audio transmission device 1.
  • the audio transmission apparatus 1 includes a decoder 41a to 41c corresponding to each audio input source 4 such as a CD player, a Blu-ray player, a network player, a PLL 42a to 42c corresponding to each audio input source 4, and an SRC 43a corresponding to each audio input source 4.
  • a clock generation unit 44 for generating a system clock
  • a daisy format Gen45 for transmitting each sound data after frequency conversion by the SRCs 43a to 43c in synchronism with the sound output timing
  • a memory 46 for storing each sound data
  • various controls A nonvolatile memory 47 for storing programs and control data, and a CPU 48 (Central Processing Unit) for performing various arithmetic processes according to the control program stored in the nonvolatile memory 47 are provided.
  • a CPU 48 Central Processing Unit
  • Daisy format Gen 45 performs conversion to a data protocol as shown in FIG. Specifically, the audio data output from the SRCs 43a to 43c is rearranged in the order of transfer, or is replaced with a frequency clock for daisy chain transmission (a transmission clock synchronized with the clock on the reading side of the SRC 43). Or embed Sync_data. With this configuration, the audio transmission device 1 can send out audio data having different sampling rates on a single daisy chain.
  • FIG. 17 is a control block diagram showing a modification of FIG. FIG. 17 shows an example of inputting a microphone signal.
  • an analog signal is input in this way, it is digitized by the A / D converter 49.
  • the clock used by the A / D converter 49 is the frequency clock on the reading side of the SRCs 43a and 43b, or is multiplied or This is a divided frequency clock.
  • the SRC 43 corresponding to the analog signal input unit can be omitted.
  • FIG. 18 is a control block diagram of the speaker 2.
  • the speaker 2 includes a daisy data separation unit 51 that separates Sync_data and Normal_data from a frame, a PLL 52 that performs phase synchronization, a memory 53 that temporarily stores audio data, a counter 54 that measures sound output timing, DA converter 55, amplifier 56, audio output mechanism 57, non-volatile memory 58 for storing various control programs and various control data, control unit 59 for controlling the entire speaker 2, daisy active for daisy output active setting / inactive setting A control unit 60 is provided.
  • the speaker 2 operates by generating an internal system clock by separating the Sync_data of the data input from Daisy In and assembling the PLL 52 in synchronization with the data. That is, a pulse is generated at the timing at which Sync_data is detected, and the pulse is input to the phase comparator of the PLL 52 as a primary source. Thereby, the operation clock of the DA converter 55 can be generated by the PLL 52. Since Sync_data is transmitted at a sampling frequency cycle, the PLL 52 generates a frequency clock obtained by multiplying the sampling frequency. In addition, the speaker 2 synchronizes the timing of sound output by temporarily storing the input data in the memory 53 (the sound output between the speakers 2 is adjusted with respect to the shift due to daisy chain transmission within one sampling frequency).
  • FIG. 19 is a diagram illustrating the relationship between input data with different sampling rates and SRC output data.
  • the figure for simplification, only one channel is input from each audio input source 4 (actually, a plurality of channels can be input for each audio input source 4).
  • the upper part shows the signal input on the input side
  • the lower part shows the output signal of the SRC 43. In this way, by passing through the SRC 43, the reproduction clock of each audio input source 4 is converted into a constant frequency clock.
  • FIG. 20 is a diagram showing the relationship between the SRC output data and the transfer image on the daisy chain.
  • S indicates Sync_data
  • d * indicates the period (frame number).
  • the SRC 43 is provided for each audio input source 4 and the reproduction clock of each audio input source 4 is converted into a constant frequency clock. Can be played simultaneously in one daisy chain. Also, since the audio data after frequency conversion is buffered in the memory 46, the output timing of the audio data can be synchronized (synchronized). Further, since a plurality of audio data is packed and transmitted in one frame, transmission control can be easily performed. In other words, compared to the case where packing is not performed, only one Sync_data (information for identifying data of the corresponding channel) is required, so the amount of data can be reduced, and one frame of data is aligned with 32 bits. Therefore, Sync_data detection and data separation of each channel can be easily performed. In addition, although the example of 32 bits was demonstrated, as long as it does not deviate from the meaning of this application, other numbers of bits are applicable.
  • the speaker 2 is provided with a PLL 52 and generates an internal clock in synchronization with the separated Sync_data, so that synchronization with other speakers 2 can be achieved. Further, since the audio data is temporarily stored in the memory 53 and the output timing is counted by the counter 54, it is possible to eliminate the deviation of the sound output between the speakers due to the daisy chain transmission within one sampling frequency.
  • the speaker 2 pauses the amplifier 56 and the like to enter a standby state, thereby reducing power consumption.
  • the amplifier 56 is provided on the speaker 2 side, control can be performed independently for each channel, so that power consumption can be suppressed as compared with a configuration in which the amplifier is provided on the audio transmission device 1 side.
  • FIGS. 27 a third embodiment of the present invention will be described with reference to FIGS.
  • the present embodiment is characterized in that synchronized playback of a plurality of types of audio data having different sampling rates is realized using FiFo61 (First-In First-Out) (see FIG. 27). Only differences from the first and second embodiments will be described below.
  • the same components as those in the first and second embodiments are denoted by the same reference numerals, and detailed description thereof is omitted. Also, modifications applied to the same configurations and parts as those in the first and second embodiments are similarly applied to this embodiment.
  • FIG. 21 is a functional block diagram of the speaker system SY according to the third embodiment.
  • the audio transmission device 1 includes an option ID assignment setting unit 121, an audio input unit 103, a device side buffer 122, a buffer monitoring unit 123, a transmission control unit 124, and a resync setting command unit 125 as main functional configurations.
  • the option ID assignment setting unit 121 performs an option ID assignment process.
  • the speaker ID of each speaker 2 is associated with the option ID. Details will be described later with reference to FIG.
  • the voice input unit 103 inputs voice data from each voice input source 4.
  • the apparatus-side buffer 122 buffers the audio data input from each audio input source 4, and corresponds to the FiFo 61 in FIG.
  • the buffer monitoring unit 123 monitors the data amount of each device-side buffer 122 and corresponds to the FiFo data monitoring unit 62 in FIG.
  • the transmission control unit 124 determines the number of pieces of audio data to be read from each device-side buffer 122 based on the monitoring result of each buffer monitoring unit 123, and packs each piece of audio data having the determined number of data into one frame. And transmit with a single clock.
  • the transmission control unit 124 has the daisy format Gen 45 and the CPU 48 of FIG. 27 as main parts.
  • the frame of this embodiment includes a channel data area group (“Ch1... Ch Last”) composed of a plurality of Normal_data (channel data areas) corresponding to each speaker 2 and each speaker.
  • 2 includes an option data area group (“Op1... Op Last”) composed of a plurality of Option_data (option data areas) corresponding to 2.
  • the transmission control unit 124 embeds a total of two samplings (two data) of audio data in the channel data area and the option data area.
  • the transmission control unit 124 does not incorporate data into each data area.
  • the transmission control unit 124 indicates, in an option flag, whether each channel data area corresponding to each speaker 2 has data in each option data area corresponding to each speaker 2. Similarly, the validity / invalidity of the corresponding data is represented by an enable flag in each channel data area corresponding to each speaker 2. With these flags, the speaker 2 is notified of the location of the transmitted data.
  • the transmission control unit 124 represents a resync flag in the Sync_data pack of the frame.
  • “when a predetermined condition is satisfied” refers to switching of audio input, changing the sampling rate of input audio data, turning on the power of the audio transmission device, starting output of audio data, and the like.
  • the resync setting command unit 125 classifies the plurality of speakers 2 for each zone (for example, for each room) and commands the resync setting to each speaker 2 in the zone to be subjected to the synchronization process. On the speaker 2 side, synchronization processing is performed based on the command of the resync setting command unit 125 and the detection of the resync flag. Details will be described later with reference to FIGS.
  • the speaker 2 has, as main control configurations, a data selection acquisition unit 202, a daisy chain output unit 203, a speaker side buffer 221, a buffer control unit 222, an audio output unit 207, a standby control unit 213, a resync setting unit 223, and a synchronization.
  • a processing unit 224 is provided.
  • the speaker-side buffer 221 buffers audio data transmitted from the audio transmission device 1, and corresponds to FiFo64 in FIG.
  • the buffer control unit 222 writes audio data to the speaker side buffer 221 and monitors the amount of written data, and performs variable control of the read clock based on the monitoring result.
  • the audio output unit 207 outputs audio based on audio data read from the speaker-side buffer 221 and corresponds to the audio output mechanism 57 in FIG.
  • the standby control unit 213 sets the speaker 2 to the standby state when the detection of the enable flag “invalid” continues for a certain period of time.
  • the resync setting unit 223 performs the resync setting of the speaker 2 according to the command of the resync setting command unit 125. Specifically, resync is set in the resync register.
  • the synchronization processing unit 224 performs synchronization processing when the resync is set by the resync setting unit 223 and the resync flag “ON” is detected from the Sync_data pack.
  • Sync_data of this embodiment defines a header at 0 to 7 bits and a reset at 9 bits. These are the same as in the second embodiment, but differ in that resync is defined at the eighth bit.
  • the resynchronization is for causing the speaker 2 to execute resynchronization processing when it is necessary to regain synchronization as described above. Since there is no need for resynchronization processing when there is no change in the audio input, the normal Sync_data resync flag is set to “0: OFF”.
  • (B) in the figure shows an assignment example of the Normal_data pack.
  • the Normal_data of this embodiment is similar to the second embodiment in that enable is defined in the 0th bit, but differs in that an option (option flag) is defined in the 1st bit.
  • the option is set to “1: Yes” if audio data is incorporated in the option data area, and “1: None” if audio data is not incorporated in the option data area.
  • the audio transmission device 1 determines whether or not there is a speaker 2 for which the speaker ID is set and the option ID is not set (S91). If it does not exist (S91: No), there is no need for the option ID assignment setting process, so an error process is performed (S92). If it exists (S91: Yes), the speaker ID to be set (initially “1”) is set (S93), and assignment loop processing is started sequentially (S94).
  • the option ID to be set is set (the number of speakers plus the speaker ID, S95), and the set option ID is stored (S96).
  • the speaker ID to be set is incremented (S98), and it is determined whether the speaker ID is valid (S99).
  • S99 Yes when assignment of all the speakers 2 is not completed
  • S94 and subsequent steps are repeated. If the speaker ID is invalid (if all the speakers 2 have been assigned, S99: No), the option ID assignment setting process is terminated.
  • FIG. 25 is a flowchart showing a synchronization reset process on the voice transmission device 1 side. If the audio transmission device 1 determines that a synchronous reset process is necessary, such as when the sampling rate of input data is changed or when input switching is performed (S111: Yes), the audio transmission device 1 reads the speaker ID in the corresponding zone. (S112), the resync preprocessing loop of the speaker 2 in the corresponding zone is started (S113). In the resync preprocessing loop, the corresponding speaker 2 is set to mute (S114), and the resync setting is performed for each speaker 2 (S115).
  • the mute command is issued to the corresponding speaker 2 and the setting to the resync register is commanded.
  • the mute setting is for preventing noise due to the resync operation.
  • the speaker ID of the first speaker 2 in the corresponding zone is read (S120), and a resync post-processing loop is started (S121).
  • the resync post-processing loop the resync setting of the corresponding speaker 2 is canceled and the mute setting is canceled (S122, S123).
  • S124 the post-resync processing loop is completed (S124)
  • the next speaker ID in the corresponding zone is read (S125), and it is determined whether or not the remaining speaker 2 exists in the corresponding zone (S126). When it exists (S126: Yes), S120 and subsequent steps are repeated. If not (S126: No), the synchronization reset process is terminated.
  • the speaker 2 refers to the resync register to determine whether or not the resync setting is performed (S131). If the resync setting is performed (S131: Yes), the mute setting is performed (S132). When Sync_data of the resync flag “1: ON” is input (S133: Yes), the resync operation is performed (S134). Thereafter, when a mute setting cancel command is acquired from the audio transmission device 1 (S135: Yes), the mute setting is canceled (S136), and the synchronization reset process is terminated. As described above, by performing the synchronization reset processing according to the flow of FIG. 25 and FIG. 26, even in the present embodiment in which each audio data is transmitted as one frame (even when only one Sync_data exists in the frame) ), You can resync for each zone.
  • FIG. 27 is a control block diagram of the audio transmission device 1.
  • the audio transmission device 1 includes decoders 41a and 41b corresponding to each audio input source 4 such as a CD player and a Blu-ray player, PLLs 42a and 42b corresponding to each audio input source 4, FiFo 61a and 61b corresponding to each audio input source 4, FiFo data monitoring units 62a and 62b that monitor the amount of data in each FiFo 61a and 61b, a clock generation unit 44 that generates a system clock, a daisy format Gen45 that transmits each audio data read from each FiFo 61a and 61b, and each audio data
  • a memory 46 for storing, a resync timing Gen 63 for performing resync setting, a non-volatile memory 47 for storing various control programs and control data, and a CPU 48 for performing various arithmetic processes
  • the audio transmission apparatus 1 detects that the amount of data in the FiFo 61 corresponding to the channel 1 exceeds the upper threshold by the FiFo data monitoring unit 62, the input of the channel 1 is faster than the daisy chain transmission ( The input rate exceeds the transfer rate), data for one sampling is stored in the channel data area of “Ch1”, and the next sampling data is stored in the option data area of “Ch1” (FIG. 22). reference).
  • the input rate exceeds the transfer rate
  • data for one sampling is stored in the channel data area of “Ch1”
  • the next sampling data is stored in the option data area of “Ch1” (FIG. 22). reference).
  • the daisy format Gen45 incorporates the enable “1: valid” in the 0th bit of the Normal_data pack and the option “1: present” in the 1st bit. .
  • daisy format Gen 45 determines that the input of channel 1 is slower than daisy chain transmission, and the channel of “Ch1” Do not include in the data area. In this case, the daisy format Gen 45 enables the 0th bit of the Normal_data pack “0: invalid”.
  • the resync timing Gen63 creates the timing for setting the sync_data resync flag. Specifically, when the CPU 48 detects that the sampling rate of input data has been changed or input switching has been performed, resync setting of the corresponding speaker 2 is performed. When the resync settings for all the speakers 2 have been completed, the resync timing Gen63 outputs a timing signal at the timing of Sync_data embedding, and the daisy format Gen45 sets the resync flag to “1: ON” in accordance with this signal. Is generated.
  • FIG. 28 is a control block diagram of the speaker 2. As shown in the figure, the speaker 2 measures a sound output timing, a daisy data separation unit 51 that separates Sync_data and Normal_data from a frame, a PLL 52 for synchronizing with Sync_data, a memory 53 that temporarily stores audio data.
  • a daisy data separation unit 51 that separates Sync_data and Normal_data from a frame
  • a PLL 52 for synchronizing with Sync_data
  • a memory 53 that temporarily stores audio data.
  • FiFo64 for buffering audio data
  • FiFo data monitoring unit 65 for monitoring the amount of data in FiFo64
  • PLL 66 for changing the readout clock of FiFo64
  • a transmission source for outputting channel assignment sound (identification signal) Identification signal Gen67
  • selector 68 for switching between identification signal and normal sound
  • DA converter 55 mute circuit 69 for muting the audio signal during synchronization reset processing
  • amplifier 56 audio output mechanism 57
  • Non-volatile memory 58 for storing data
  • Manufacturers 2 controller 59 that controls the whole
  • a daisy active control unit 60 for active set / inactive settings daisy output.
  • the speaker 2 separates and captures “Ch1” data from the data (frame) input from Daisy In by the daisy data separation unit 51. If the enable flag of the fetched “Ch1” data is set to “1: Valid”, the data is written into the FiFo64. In addition, when the option flag is set to “1: present”, the data of “Op1” is also written. On the other hand, when the enable flag is set to “0: invalid”, writing to the FiFo 64 is not performed. Further, even when the option flag is set to “0: None”, the data of “Op1” is not written. Further, the data amount of FiFo64 is monitored by the FiFo data monitoring unit 65.
  • the PLL 66 When the upper limit threshold is exceeded, the PLL 66 is controlled to increase the frequency clock on the reading side of the FiFo64. On the other hand, when the value falls below the lower limit threshold, the PLL 66 is controlled to lower the frequency clock on the reading side of the FiFo64. As a result, it is possible to perform audio reproduction in synchronization with the input of the audio transmission device 1 indirectly without emptying or overflowing the FiFo64.
  • FIG. 29 is a diagram illustrating a relationship between input data with different sampling rates and data of FiFo output.
  • the upper part shows the signal input on the input side
  • the lower part shows the output signal on the reading side of the FiFo 64 provided in the speaker 2.
  • the speaker 2 performs reproduction with a frequency clock equivalent to the frequency clock of the input audio input source 4.
  • FIG. 30 is a diagram showing a relationship between input data with different sampling rates and a transfer image on the daisy chain.
  • S indicates Sync_data
  • the number represented by “d *” indicates the period (frame number).
  • Dx indicates an enable flag “0: Invalid”, and “ox” indicates an option flag “0: None”.
  • the first sampling data is the channel data area of the first frame
  • the second sampling data is the channel data area of the second frame
  • the third sampling data is the option of the second frame.
  • the data data area, the fourth sampling data are incorporated into the channel data area of the third frame, and data transmission is performed.
  • the audio transmission apparatus 1 side monitors the data amount of the FiFo 61 corresponding to each audio input source 4, and each audio data included in one frame based on the monitoring result.
  • a plurality of types of audio data having different sampling rates can be transmitted.
  • the transmitted audio data is once written in the FiFo 64, and the read clock is variably controlled according to the amount of data, so a plurality of types of audio data are reproduced simultaneously with one daisy chain speaker system.
  • the voice transmission device 1 side when the data amount of FiFo 61 exceeds the upper threshold, when it is lower than the upper threshold and equal to or higher than the lower threshold, it is divided into three patterns when it falls below the lower threshold, and is transferred to the frame. Since the audio data is recorded and each pattern is indicated by an option flag and an enable flag, each pattern can be accurately discriminated on the speaker 2 side, and buffer writing processing and reproduction clock generation can be performed.
  • the resync flag can be set in the Sync_data pack, when the audio input is switched, when the sampling rate of the input audio data is changed, when the audio transmission device 1 is turned on, when the output of the audio data is started, etc. Resynchronization processing can be performed at a necessary timing.
  • the resync setting command unit 125 can command the resync setting to each speaker 2 in the zone to be synchronized, an efficient resynchronization process only for the speakers 2 in the zone requiring the resynchronization process. It can be performed.
  • FIGS. 1 a fourth embodiment of the present invention will be described with reference to FIGS.
  • the present embodiment is characterized in that power is supplied from the audio transmission device 1 to each speaker 2. Only the feature points different from the above embodiments will be described below.
  • the same components as those in the above-described embodiments are denoted by the same reference numerals, and detailed description thereof is omitted.
  • the modification applied about the structure similar to each said embodiment and a part is applied similarly about this embodiment.
  • FIG. 31 is a functional block diagram of the speaker system SY according to the fourth embodiment.
  • the arrows in the figure indicate the flow of audio data and the flow of power.
  • functions related to power supply will be mainly described.
  • the audio transmission device 1 includes an audio input unit 103, an audio transmission unit 131, a power supply unit 132, and a charging control unit 133 as main functional configurations.
  • the voice input unit 103 inputs voice data from each voice input source 4.
  • the audio transmission unit 131 packs each input audio data into a frame and transmits the daisy chain. Note that the audio transmission unit 131 functions as the transmission control units 106, 113, and 124 of the first to third embodiments.
  • the power supply unit 132 charges the secondary batteries 232 in the plurality of speakers 2 connected in a daisy chain via the cable 3.
  • the charging control unit 133 controls charging to each speaker 2 by switching target speakers to be specified as charging targets among the plurality of speakers 2 in a time-sharing manner (sequentially switching every predetermined time). Specifically, the elapsed time from the start of charging the target speaker is counted by a counter, an RTC (Real Time Clock) or the like, and when a predetermined time has elapsed, it is determined that charging is complete and the target speaker is switched.
  • RTC Real Time Clock
  • the speaker 2 includes a data separation unit 202, a daisy chain output unit 203, an audio output unit 207, a power control unit 231 and a secondary battery 232 as main functional configurations.
  • the flow of audio data in the data separation unit 202, daisy chain output unit 203, and audio output unit 207 is the same as in each of the above embodiments.
  • the power control unit 231 receives power supply from the audio transmission device 1 (charge control unit 133) via the cable 3 and charges the secondary battery 232. In addition, when the power control unit 231 is not designated as a target speaker to be charged, the power control unit 231 holds the charge stop state of the secondary battery 232.
  • the secondary battery 232 various rechargeable batteries such as a lithium ion secondary battery and a lithium ion polymer secondary battery are applicable.
  • the power control unit 231 includes a remaining amount detection unit 231a and a charge stop request transmission unit 231b.
  • the remaining amount detection unit 231a detects the remaining amount of the secondary battery 232 of the speaker 2 itself.
  • the charge stop request transmission unit 231b detects that the remaining amount of the secondary battery 232 exceeds a predetermined amount (first predetermined amount) by the remaining amount detection unit 231a, the charging stop request transmission unit 231b connects the cable 3 to the audio transmission device 1.
  • a charge stop request is transmitted via
  • the consumption ratio becomes a predetermined value or less (for example, 1/5 or less of the total battery capacity)
  • the charging stop request may be transmitted at the same time.
  • FIG. 32A is a diagram showing pin assignment of a straight connection type LAN cable. As shown in the figure, generally, only two pairs of Nos. 1, 2, 3, and 6 are used lines. For this reason, in this embodiment, as shown in FIG. 4B, the audio data is transmitted and the power is supplied using two unused pairs of lines 4, 5, 7, and 8.
  • FIG. 33 is a flowchart showing the charging process on the voice transmission device 1 side.
  • the audio transmission device 1 designates the target speaker to be charged by the speaker ID and notifies the target speaker (S141). Thereafter, the reception timeout of the ACK signal is determined (S142). If the timeout has occurred (S142: Yes), error processing is performed (S143). Specifically, the error is notified by voice output or screen display. On the other hand, when the time-out does not occur (S142: No) and an ACK signal is received from the target speaker (S144: Yes), charging is started (S145).
  • the charging counter is set simultaneously with the start of charging (S146), and when the counting is ended (S147: Yes), charging is terminated and a charging timeout is notified to the target speaker (S148). Thereafter, the speaker ID is incremented (S149), and S141 and subsequent steps are repeated for the next-stage speaker 2.
  • the charging is also terminated (S150), and the speaker ID is incremented (S149).
  • FIG. 34 is a flowchart showing the charging process on the speaker 2 side.
  • the speaker 2 target speaker determines the consistency between the received speaker ID and the ID stored in the speaker 2 (S162). If they match (if the speaker IDs match, S162: Yes), the ACK signal is returned to the audio transmission device 1 and the supply of power transmitted from the audio transmission device 1 is started (to the secondary battery 232). Charging is performed, S164). Thereafter, the remaining battery level of the secondary battery 232 is checked (S165), and when it is determined that the battery has become full (when it is detected that the remaining battery level of the secondary battery 232 exceeds a predetermined amount) (S166: Yes).
  • the voice transmission device 1 is notified of a charge stop request (S167), and the charging process is terminated. On the other hand, also when the charging timeout notification is received from the audio transmission device 1 as the interrupt processing (S168), the charging processing is terminated.
  • transmission and reception of various notifications and ACK signals use two pairs of lines 1, 2, 3, and 6 of the LAN cable (see FIG. 32).
  • each speaker 2 can be achieved with only one cable 3. Can be stably supplied. As a result, power supply is not performed randomly, and the problem of allowable current capacity and the problem of voltage drop and heat generation due to the impedance of the power supply line can be solved.
  • the target speaker to be charged is transmitted by transmitting the speaker ID used at the time of audio transmission, the target speaker can be easily and reliably switched.
  • the system configuration can be further reduced.
  • the charging stop state of the secondary battery 232 is maintained, so that more stable power supply can be performed as the entire system.
  • the audio transmission device 1 since the audio transmission device 1 counts the elapsed time from the start of charging and switches the target speaker, the power transmission to each speaker 2 is performed in an appropriate cycle for maintaining a sufficiently charged state on average. It can be carried out.
  • a charge stop request is received from the speaker 2 being charged, this is used as an interrupt process to switch the target speaker, so that overcharging can be prevented and power supply to each speaker 2 can be efficiently performed. Can do.
  • charging is performed by designating target speakers one by one.
  • target speakers may be designated by a plurality of units.
  • the charging order may be from the downstream speaker 2 to the upstream side instead of going from the upstream speaker 2 to the downstream side, and the charging is performed in a random order regardless of the arrangement order of the speakers 2. May be.
  • the target speaker may perform switching control for only the speaker 2 belonging to a specific zone (for example, the speaker 2 to be reproduced).
  • a general-purpose LAN cable is used as a transmission path for supplying power, but a dedicated cable may be used.
  • a dedicated speaker ID assigned for power supply may be used instead of the speaker ID used for audio transmission.
  • each speaker 2 when it is detected that the remaining amount of the secondary battery 232 is equal to or less than a predetermined amount (second predetermined amount) during non-charging, You may transmit a charge start request
  • the charging control unit 133 receives a charging start request from the non-charging speaker 2, the charging control unit 133 stops charging the charging speaker 2 or waits for the charging of the charging speaker 2 to be completed. It is preferable to switch the speaker 2 that has requested charging start to the target speaker.
  • the speaker 2 is set as the next target speaker 2, and thus insufficient charging can be prevented. Thereby, the stable audio
  • FIG. 35 is a system configuration diagram showing a first modification.
  • cables 3a and 3b extend in two directions from the audio transmission device 1.
  • two output terminals 12 for performing daisy chain output are required on the back surface of the audio transmission device 1 (see FIG. 3A).
  • the audio data to be transmitted need not be distinguished for each of the cables 3a and 3b, and the same audio data may be transmitted.
  • the degree of freedom of daisy chain connection can be improved by providing a plurality of output terminals 12 and only flowing the same data to both terminals.
  • the number of output terminals 12 provided on the back surface of the audio transmission device 1 is arbitrary, and may be three or more.
  • FIG. 36 is a system configuration diagram showing a second modification.
  • This modification is characterized by using an omnidirectional speaker 2.
  • a plurality of multi-channel spaces can be formed in one room.
  • the television 81 and the audio transmission device 1 are connected.
  • the source source of the projector 82 and the audio transmission device 1 are also connected.
  • the omnidirectional speaker 2 for example, watching sports on the television 81 at some times and watching a movie on the projector 82 at some times can change the wiring. Instead, it can be realized only by changing the setting of the voice transmission device 1.
  • the speaker 2a is used as a surround speaker (right side) placed behind the viewer, and in the latter case, it is used as a front speaker (left side) placed on the left and right of the screen.
  • the speaker 2b is used as a surround speaker (left side) in the former and latter cases. Thereby, the same speaker 2 can be shared in a plurality of trial listening environments.
  • the room can be easily redesigned.
  • the usage of the speakers 2 can be changed simply by changing the settings of the audio transmission device 1 as described above.
  • the speaker 2 used for the viewing environment on the TV 81 side is arranged, and when changing the pattern or adding a trial listening environment (for the projector 82, etc.), the corresponding speaker 2 is added and audio transmission is performed.
  • two listening environments can be created.
  • speaker system SY which combined these suitably may be realized.
  • the program can be provided by being stored in various recording media (CD-ROM, flash memory, etc.). That is, a program for causing a computer to function as each part of the audio transmission device 1 or the speaker 2 and a recording medium on which the program is recorded are also included in the scope of the right of the present invention.
  • the system configuration of the speaker system SY, the device configurations of the audio transmission device 1 and the speaker 2, processing steps, and the like can be appropriately changed without departing from the gist of the present invention, regardless of the above embodiment.
  • Charge control unit 201 ... ID storage unit 202 ... Data selection acquisition unit 203 ... Daisy chain output unit 204 ... Data buffer 2 5 ... Interpolation data generation unit 206 ... Selection circuit 207 ... Audio output unit 208 ... Speaker side memory 211 ... Phase synchronization unit 212 ... Counter unit 213 ... Standby control unit 221 ... Speaker side buffer 222 ... Buffer control unit 223 ... Resync setting unit 224 ... Synchronization processing unit 231 ... Power control unit 231a ... Remaining amount detection unit 231b ... Charge stop request transmission unit 232 ... Secondary battery SY ... Speaker system

Abstract

The purpose of the present invention is to simultaneously reproduce multiple kinds of audio data with different sampling rates by using one daisy-chained speaker system. The speaker system (SY) is provided with: multiple speakers (2) that are connected in series by means of one transmission line; and an audio transmission device (1) that transmits audio data, which are input from multiple audio input sources, to the respective speakers (2) via the transmission line. The audio transmission device (1) is provided with: multiple frequency conversion units (111) that are provided so as to correspond to the respective audio input sources, and that convert recovered clocks, which have been recovered from the respective audio input sources, into clocks with predetermined frequencies; a device-side memory (112) for buffering the respective audio data that are frequency-converted by the respective frequency conversion units (111); and a transmission control unit (113) that packs the respective audio data read from the device-side memory (112) into one frame and transmits the frame using a single clock that corresponds to a clock with a predetermined frequency.

Description

スピーカーシステム、音声伝送装置、スピーカー、音声伝送方法およびプログラムSpeaker system, audio transmission device, speaker, audio transmission method and program
　本発明は、１本の伝送路で直列接続された複数のスピーカーと、伝送路を介して各スピーカーに音声データを伝送可能な音声伝送装置と、を備えたスピーカーシステム、音声伝送装置、スピーカー、音声伝送方法およびプログラムに関する。 The present invention provides a speaker system, a sound transmission device, a speaker, and a plurality of speakers connected in series via a single transmission line, and a sound transmission device capable of transmitting sound data to each speaker via the transmission line. The present invention relates to an audio transmission method and program.
　従来、デイジーチェーン接続（１本の伝送路による直列接続，数珠繋ぎ接続）された複数のスピーカーに音声データを伝送するデイジーチェーンスピーカーシステムが知られている（例えば、特許文献１および特許文献２）。当該デイジーチェーンスピーカーシステムは、従来のように、１の音声（ＡＶ）アンプに対して複数のスピーカーを個々に接続する構成と比較して、伝送路数を削減できる、音声アンプ側の出力端子数を減らすことができる、接続作業が容易である、などのメリットがある。また、チャンネル数は年々増加傾向にあるが、デイジーチェーンスピーカーシステムの場合、スピーカーの追加を柔軟に行うことができるため、多チャンネル化に対応しやすいといったメリットもある。 Conventionally, there is known a daisy chain speaker system that transmits audio data to a plurality of speakers connected in a daisy chain (series connection by a single transmission line, daisy chain connection) (for example, Patent Document 1 and Patent Document 2). This daisy chain speaker system can reduce the number of transmission paths compared to a conventional configuration in which a plurality of speakers are individually connected to one audio (AV) amplifier as in the prior art. There are merits such that the connection work can be reduced. The number of channels tends to increase year by year, but in the case of a daisy chain speaker system, speakers can be added flexibly, and there is also an advantage that it is easy to cope with the increase in number of channels.
特開２００５－１７５７４５号公報JP 2005-175745 A 特開２００９－２５１８９１号公報JP 2009-251891 A
　ところで、上記のようなデイジーチェーンスピーカーシステムを、複数の視聴環境を有する建物内（自宅など）に構築したいといった要望がある。ところが、従来のデイジーチェーンスピーカーシステムは、１つのコンテンツ（楽曲）を再生する用途のみで用いられており、複数ゾーン（視聴エリア）で楽曲を同時に再生することができなかった。これは、異なる音声入力源から入力されたサンプリングレートが異なる音声データや同じサンプリングレートでも異なる音声入力源のデジタル音声データを、単一クロックのデイジーチェーンで同時に伝送できないためである。このため、各個人が同じ時刻に、自分の好きな楽曲をそれぞれ自分の部屋だけで楽しむためには、自宅内に複数のデイジーチェーンスピーカーシステムを構築する必要があり、現実的ではなかった。 By the way, there is a demand for building a daisy chain speaker system as described above in a building (such as a home) having a plurality of viewing environments. However, the conventional daisy chain speaker system is used only for the purpose of reproducing one content (music), and cannot reproduce the music simultaneously in a plurality of zones (viewing areas). This is because audio data with different sampling rates input from different audio input sources and digital audio data with different audio input sources even at the same sampling rate cannot be transmitted simultaneously in a single clock daisy chain. For this reason, in order for each individual to enjoy their favorite music only in their own room at the same time, it is necessary to construct a plurality of daisy chain speaker systems in the home, which is not realistic.
　本発明は、上記の問題に鑑み、１つのデイジーチェーンスピーカーシステムでサンプリングレートが異なる複数種類の音声データを同時に再生可能なスピーカーシステム、音声伝送装置、スピーカー、音声伝送方法およびプログラムを提供することを目的とする。 In view of the above problems, the present invention provides a speaker system, an audio transmission device, a speaker, an audio transmission method, and a program capable of simultaneously reproducing a plurality of types of audio data having different sampling rates in one daisy chain speaker system. Objective.
　本発明のスピーカーシステムは、１本の伝送路で直列接続された複数のスピーカーと、複数の音声入力源から入力された音声データを、伝送路を介して各スピーカーに伝送する音声伝送装置と、を備えたスピーカーシステムであって、音声伝送装置は、各音声入力源にそれぞれ対応して設けられ、各音声入力源の再生クロックを一定の周波数クロックに変換する複数の周波数変換部と、各周波数変換部による周波数変換後の各音声データをバッファリングする装置側メモリと、装置側メモリから読み出した各音声データを、１フレームにパッキングし、一定の周波数クロックに対応する単一クロックで伝送する伝送制御部と、を備えたことを特徴とする。 The speaker system of the present invention includes a plurality of speakers connected in series via a single transmission line, an audio transmission device that transmits audio data input from a plurality of audio input sources to each speaker via the transmission line, The audio transmission device is provided corresponding to each audio input source, and converts a reproduction clock of each audio input source into a constant frequency clock, and each frequency. Device-side memory for buffering each audio data after frequency conversion by the converter, and each audio data read from the device-side memory are packed into one frame and transmitted with a single clock corresponding to a fixed frequency clock And a control unit.
　本発明の音声伝送方法は、１本の伝送路で直列接続された複数のスピーカーに対し、複数の音声入力源から入力された音声データを、伝送路を介して伝送する音声伝送方法であって、各音声入力源にそれぞれ対応して設けられた複数の周波数変換部により、各音声入力源の再生クロックを一定の周波数クロックに変換する周波数変換工程と、周波数変換後の各音声データを装置側メモリにバッファリングするバッファリング工程と、装置側メモリから読み出した各音声データを、１フレームにパッキングし、一定の周波数クロックに対応する単一クロックで伝送する伝送制御工程と、を実行することを特徴とする。 The audio transmission method of the present invention is an audio transmission method for transmitting audio data input from a plurality of audio input sources to a plurality of speakers connected in series on a single transmission line via the transmission line. A frequency conversion step of converting a reproduction clock of each audio input source into a fixed frequency clock by a plurality of frequency conversion units provided corresponding to each audio input source, and each audio data after frequency conversion on the device side A buffering step of buffering in a memory, and a transmission control step of packing each audio data read from the device-side memory into one frame and transmitting it with a single clock corresponding to a constant frequency clock. Features.
　これらの構成によれば、音声入力源ごとに周波数変換部を設け、各音声入力源の再生クロックを一定の周波数クロックに変換するため、サンプリングレートが異なる複数種類の音声データを、１つのデイジーチェーンスピーカーシステムで同時に再生することができる。また、周波数変換後の各音声データを装置側メモリにバッファリングするため、音声データの出力タイミングを合わせる（同期をとる）ことができる。また、複数の音声データを１フレームにパッキングして伝送するため、伝送制御を容易に行うことができる。
　なお、「音声データ」とは、各音声入力源から入力されたサンプリングデータを指す。 According to these configurations, a frequency conversion unit is provided for each audio input source, and the reproduction clock of each audio input source is converted into a constant frequency clock. Can be played simultaneously on the speaker system. In addition, since the audio data after frequency conversion is buffered in the device-side memory, the output timing of the audio data can be synchronized (synchronized). Further, since a plurality of audio data is packed and transmitted in one frame, transmission control can be easily performed.
“Audio data” refers to sampling data input from each audio input source.
　上記のスピーカーシステムにおいて、音声伝送装置が、音声入力源からアナログ信号を入力する場合、Ａ／Ｄコンバーターに使用する周波数クロックは、一定の周波数クロックであることを特徴とする。 In the above speaker system, when the audio transmission device inputs an analog signal from an audio input source, the frequency clock used for the A / D converter is a fixed frequency clock.
　この構成によれば、アナログ信号を出力する音声入力源（マイクなど）を用いた場合、当該音声入力源に対応する周波数変換部を不要とすることができる。ここで「一定の周波数クロック」とは、アナログ信号のサンプリングに使用する周波数クロックがサンプリングレート変換後のクロックと同一であることを示す。 According to this configuration, when a voice input source (such as a microphone) that outputs an analog signal is used, a frequency conversion unit corresponding to the voice input source can be made unnecessary. Here, the “constant frequency clock” indicates that the frequency clock used for sampling the analog signal is the same as the clock after the sampling rate conversion.
　上記のスピーカーシステムにおいて、各スピーカーは、音声伝送装置により伝送されたフレームから、シンクデータおよび自スピーカー用の音声データを分離するデータ選択取得部と、選択取得されたシンクデータに同期させて内部クロックを生成する位相同期部と、を備えたことを特徴とする。 In the speaker system, each speaker has a data selection / acquisition unit that separates the sync data and the audio data for the speaker from the frame transmitted by the audio transmission device, and an internal clock in synchronization with the selected / acquired sync data. And a phase synchronizer for generating.
　この構成によれば、選択取得されたシンクデータに同期させて内部クロックを生成するため、他のスピーカーとの同期を図ることができる。 According to this configuration, the internal clock is generated in synchronization with the selected and acquired sync data, so that synchronization with other speakers can be achieved.
　上記のスピーカーシステムにおいて、各スピーカーは、選択取得された音声データをバッファリングするスピーカー側メモリと、スピーカー側メモリからの音声データの出力タイミングを計測するカウンタ部と、をさらに備えたことを特徴とする。 In the speaker system, each speaker further includes a speaker-side memory for buffering the selected and acquired audio data, and a counter unit for measuring the output timing of the audio data from the speaker-side memory, To do.
　この構成によれば、スピーカー側メモリに一旦音声データを格納し、カウンタ部にて出力タイミングをカウントするため、１サンプリング周波数内のデイジーチェーン伝送によるスピーカー間の音出しのずれを解消することができる（音出しのタイミングを合わせることができる）。 According to this configuration, since the audio data is temporarily stored in the speaker-side memory and the output timing is counted by the counter unit, it is possible to eliminate the deviation of sound output between the speakers due to daisy chain transmission within one sampling frequency. (The timing of sound output can be adjusted).
　上記のスピーカーシステムにおいて、フレームとして伝送される各音声データには、該当データの有効／無効を示すイネーブルフラグが定義されており、各スピーカーは、イネーブルフラグ「無効」の検出が一定時間続いた場合、自スピーカーをスタンバイ状態にするスタンバイ制御部をさらに備えたことを特徴とする。 In the above speaker system, an enable flag indicating validity / invalidity of the corresponding data is defined for each audio data transmitted as a frame, and each speaker detects the enable flag “invalid” for a certain period of time. The apparatus further includes a standby control unit that puts the speaker in a standby state.
　この構成によれば、イネーブルフラグ「無効」の検出が一定時間続いた場合、そのスピーカーは使用されないものと判断できるため、スタンバイ状態にする（音声アンプ、Ｄ／Ａコンバーター、カウンタ、メモリ等を休止させる）ことで、消費電力を抑えることができる。但し、スタンバイ状態時でも、デイジーチェーンの伝送だけは行い得ることが前提である。 According to this configuration, when the detection of the enable flag “invalid” continues for a certain period of time, it can be determined that the speaker is not used, so the standby state is set (the audio amplifier, the D / A converter, the counter, the memory, etc. are suspended. Power consumption can be suppressed. However, it is assumed that only daisy chain transmission can be performed even in the standby state.
　上記のスピーカーシステムにおいて、各スピーカーは、音声アンプ機能を備えたことを特徴とする。 In the above speaker system, each speaker has a sound amplifier function.
　この構成によれば、スピーカー側に音声アンプ機能を備えることで、それぞれ独立して制御できるため、音声再生が必要なチャンネルのスピーカーのみアンプ駆動すればよく、アンプ部分で使用する電力を抑えることができる。 According to this configuration, since the sound amplifier function is provided on the speaker side, each can be controlled independently, so that only the speaker of the channel that requires sound reproduction needs to be driven and the power used in the amplifier portion can be suppressed. it can.
　本発明の音声伝送装置は、上記のスピーカーシステムに用いられることを特徴とする。 The audio transmission device of the present invention is used in the above speaker system.
　本発明のスピーカーは、上記のスピーカーシステムに用いられることを特徴とする。 The speaker of the present invention is used in the above speaker system.
　本発明のプログラムは、コンピューターに、上記の音声伝送方法における各工程を実行させることを特徴とする。 The program of the present invention causes a computer to execute each step in the above-described audio transmission method.
　これらを用いることにより、１つのデイジーチェーンスピーカーシステムでサンプリングレートが異なる複数種類の音声データや同じサンプリング周波数でも異なる音声入力源のデジタル音声データを同時に再生可能なスピーカーシステムおよび音声伝送方法を実現できる。 By using these, it is possible to realize a speaker system and an audio transmission method capable of simultaneously reproducing a plurality of types of audio data having different sampling rates and digital audio data of different audio input sources even at the same sampling frequency in one daisy chain speaker system.
本発明の一実施形態に係るスピーカーシステムのシステム構成図である。1 is a system configuration diagram of a speaker system according to an embodiment of the present invention. スピーカーの配置例を示す図である。It is a figure which shows the example of arrangement | positioning of a speaker. 音声伝送装置の背面図、並びにスピーカーの前面図および背面図である。It is the rear view of an audio | voice transmission apparatus, and the front view and rear view of a speaker. 第１実施形態に係るスピーカーシステム（多チャンネル再生関連）の機能ブロック図である。It is a functional block diagram of the speaker system (related to multi-channel playback) according to the first embodiment. デイジーチェーンＩＤ初期化処理を示すフローチャートである。It is a flowchart which shows a daisy chain ID initialization process. パワーオン処理を示すフローチャートである。It is a flowchart which shows a power-on process. ゾーン数の設定処理およびチャンネル数の設定処理を示すフローチャートである。It is a flowchart which shows the setting process of the number of zones, and the setting process of the number of channels. スピーカーアサイン設定処理を示すフローチャートである。It is a flowchart which shows a speaker assignment setting process. デイジーチェーン伝送に用いるデータプロトコルの説明図である。It is explanatory drawing of the data protocol used for daisy chain transmission. 音声データシリアル伝送の概念図である。It is a conceptual diagram of audio | voice data serial transmission. チャンネル設定フラグの説明図である。It is explanatory drawing of a channel setting flag. 第１実施形態に係る音声データシリアル伝送の概念図である。It is a conceptual diagram of the audio | voice data serial transmission which concerns on 1st Embodiment. 第１実施形態に係る再生処理を示すフローチャートである。It is a flowchart which shows the reproduction | regeneration processing which concerns on 1st Embodiment. 第２実施形態に係るスピーカーシステム（同期再生（１））の機能ブロック図である。It is a functional block diagram of the speaker system (synchronous reproduction (1)) concerning a 2nd embodiment. 第２実施形態に係るSync_dataパックおよびNomal_dataパックの説明図である。It is explanatory drawing of the Sync_data pack and Normal_data pack which concern on 2nd Embodiment. 第２実施形態に係る音声伝送装置の制御ブロック図である。It is a control block diagram of the audio transmission apparatus according to the second embodiment. 第２実施形態の変形例に係る音声伝送装置の制御ブロック図である。It is a control block diagram of the audio | voice transmission apparatus which concerns on the modification of 2nd Embodiment. 第２実施形態に係るスピーカーの制御ブロック図である。It is a control block diagram of the speaker which concerns on 2nd Embodiment. サンプリングレート違いの入力データとＳＲＣ出力のデータを示す図である。It is a figure which shows the input data and sampling data of a sampling rate difference. ＳＲＣ出力のデータとデイジーチェーン上の転送イメージを示す図である。It is a figure which shows the data of SRC output, and the transfer image on a daisy chain. 第３実施形態に係るスピーカーシステム（同期再生（２））の機能ブロック図である。It is a functional block diagram of the speaker system (synchronous reproduction (2)) concerning a 3rd embodiment. 第３実施形態に係る音声データのパッキング例を示す図である。It is a figure which shows the example of packing of the audio | voice data which concern on 3rd Embodiment. 第３実施形態に係るSync_dataパックおよびNomal_dataパックの説明図である。It is explanatory drawing of the Sync_data pack and Normal_data pack which concern on 3rd Embodiment. オプションＩＤアサイン設定処理を示すフローチャートである。It is a flowchart which shows an option ID assignment setting process. 音声伝送装置側の同期リセット処理を示すフローチャートである。It is a flowchart which shows the synchronous reset process by the audio | voice transmission apparatus side. スピーカー側の同期リセット処理を示すフローチャートである。It is a flowchart which shows the speaker side synchronous reset process. 第３実施形態に係る音声伝送装置の制御ブロック図である。It is a control block diagram of the audio | voice transmission apparatus which concerns on 3rd Embodiment. 第３実施形態に係るスピーカーの制御ブロック図である。It is a control block diagram of the speaker which concerns on 3rd Embodiment. サンプリングレート違いの入力データとＦｉＦｏ出力のデータを示す図である。It is a figure which shows the input data of a sampling rate difference, and the data of a FiFo output. サンプリングレート違いの入力データとデイジーチェーン上の転送イメージを示す図である。It is a figure which shows the transfer image on the input data and daisy chain with a different sampling rate. 第４実施形態に係るスピーカーシステム（電力供給関連）の機能ブロック図である。It is a functional block diagram of a speaker system (related to power supply) according to a fourth embodiment. ＬＡＮケーブルピンアサインと、これに対する第４実施形態の適用例を示す図である。It is a figure which shows the example of application of 4th Embodiment with respect to a LAN cable pin assignment. 音声伝送装置側の充電処理を示すフローチャートである。It is a flowchart which shows the charging process by the audio | voice transmission apparatus side. スピーカー側の充電処理を示すフローチャートである。It is a flowchart which shows the charging process by the side of a speaker. 第１の変形例を示すシステム構成図である。It is a system configuration figure showing the 1st modification. 第２の変形例を示すシステム構成図である。It is a system configuration figure showing the 2nd modification.
　［第１実施形態］
　以下、添付の図面を参照し、本発明の一実施形態について説明する。まず、図１～図１３を参照し、第１実施形態について説明する。図１は、スピーカーシステムＳＹのシステム構成図である。同図に示すように、スピーカーシステムＳＹは、ＡＶセンターユニット（ＡＶレシーバー）として機能する音声伝送装置１と、複数個のスピーカー２と、各スピーカー２を直列接続するためのデイジーチェーンケーブル３（伝送路，以下単に「ケーブル３」と表記する）と、音声データを入力するための音声入力源４と、から成る。 [First Embodiment]
Hereinafter, an embodiment of the present invention will be described with reference to the accompanying drawings. First, the first embodiment will be described with reference to FIGS. FIG. 1 is a system configuration diagram of the speaker system SY. As shown in the figure, the speaker system SY includes an audio transmission device 1 that functions as an AV center unit (AV receiver), a plurality of speakers 2, and a daisy chain cable 3 (transmission) for connecting the speakers 2 in series. Path, hereinafter simply referred to as “cable 3”) and an audio input source 4 for inputting audio data.
　音声入力源４は、音声伝送装置１と有線／無線接続され、主にネットワーク経由のコンテンツ（ストリーミング再生含む）、ＣＤプレーヤー、ブルーレイプレーヤー、スマートフォン、ネットワークプレーヤー等が用いられる。 The audio input source 4 is wired / wirelessly connected to the audio transmission device 1 and mainly uses content via a network (including streaming playback), a CD player, a Blu-ray player, a smartphone, a network player, and the like.
　音声伝送装置１は、各音声入力源４に対応したデコーダー４１（図１６等参照）を内蔵しており、各音声入力源４から入力されたマルチチャンネルの音声データを、ケーブル３を介して各スピーカー２に伝送する。ケーブル３には、各チャンネルの音声データをパックしたものがシリアルに伝送される。また、音声伝送装置１は、電力供給装置としても機能し、ケーブル３を介して各スピーカー２に電力供給を行う。なお、電力の供給については、第４実施形態にて説明する。また、ケーブル３としては、汎用ケーブルなど（図３２参照）を用いる。 The audio transmission device 1 incorporates a decoder 41 (see FIG. 16 and the like) corresponding to each audio input source 4, and multi-channel audio data input from each audio input source 4 is transmitted via the cable 3 to each Transmit to the speaker 2. A cable 3 in which audio data of each channel is packed is serially transmitted. The audio transmission device 1 also functions as a power supply device, and supplies power to each speaker 2 via the cable 3. The supply of power will be described in the fourth embodiment. As the cable 3, a general-purpose cable or the like (see FIG. 32) is used.
　スピーカー２は、アンプ５６（音声アンプ機能，図１８等参照）を内蔵しており、音量調節およびチャンネル選択指定は、ケーブル３のコントロール線を用いて行う。また、スピーカー２は、１本のケーブル３（物理的にはスピーカー２間をつなぐスピーカー数分のケーブル３）を介して複数個が直列接続され、自宅や館内に配置される。図２は、自宅内の２つ部屋にスピーカーシステムＳＹを構築した場合のスピーカー２の配置例を示す図である。同図の例では、音声伝送装置１からリビング内に配置された一部のスピーカー２、キッチン内に配置されたスピーカー２、さらにリビング内に配置された残りのスピーカー２の順で、合計１２個のスピーカー２がデイジーチェーン接続されている。このように、本実施形態のスピーカーシステムＳＹは、複数の部屋に跨ってスピーカーシステムＳＹを導入可能となっている。したがって、音声伝送装置１は、自宅内に１台だけ設置することにより、複数のゾーン（再生エリア）で同時に異なったコンテンツ再生が可能となる。また、音声伝送装置１は、上記の通り複数の音声入力源４と接続され、各音声入力源４から入力された音声データを同時に再生できるようになっている。これにより、リビングでは、ＣＤプレーヤーの音楽を楽しみ、キッチンでは、ブルーレイプレーヤーの音楽を楽しむ、といった使い方が可能である。また、自宅内にスピーカーシステムＳＹを導入することで、部屋毎に音声伝送装置１を配置する必要がなくなり、ケーブル数を削減できるため、オーディオ視聴環境をシンプルに構築できる。また、１台の音声伝送装置１からスピーカー２ごとにケーブル３を接続する従来のシステムと比較して、ケーブル数を大幅に削減でき、清掃時等にケーブル３が邪魔になるといった問題も解消できる。 The speaker 2 has a built-in amplifier 56 (sound amplifier function, see FIG. 18 and the like), and volume control and channel selection designation are performed using the control line of the cable 3. A plurality of speakers 2 are connected in series via one cable 3 (physically, the number of speakers 3 connecting between the speakers 2), and are arranged at home or in a hall. FIG. 2 is a diagram illustrating an arrangement example of the speakers 2 when the speaker system SY is constructed in two rooms at home. In the example of the figure, a total of 12 speakers are arranged in the order of a part of speakers 2 arranged in the living room from the audio transmission device 1, a speaker 2 arranged in the kitchen, and the remaining speakers 2 arranged in the living room. Are connected in a daisy chain. Thus, the speaker system SY of this embodiment can introduce the speaker system SY across a plurality of rooms. Therefore, by installing only one audio transmission apparatus 1 at home, different contents can be reproduced simultaneously in a plurality of zones (reproduction areas). In addition, the audio transmission device 1 is connected to a plurality of audio input sources 4 as described above, and can simultaneously reproduce audio data input from the audio input sources 4. Accordingly, it is possible to enjoy the music of the CD player in the living room and the music of the Blu-ray player in the kitchen. In addition, by introducing the speaker system SY in the home, it is not necessary to arrange the audio transmission device 1 for each room, and the number of cables can be reduced, so that an audio viewing environment can be simply constructed. Moreover, compared with the conventional system which connects the cable 3 for every speaker 2 from the one audio | voice transmission apparatus 1, the number of cables can be reduced significantly and the problem that the cable 3 becomes obstructive at the time of cleaning etc. can also be eliminated. .
　次に、図３を参照し、音声伝送装置１およびスピーカー２の装置構成について説明する。図３（ａ）は、音声伝送装置１の背面図である。音声伝送装置１の背面には、ＡＣ電源からの電源供給を受ける電源供給口１１と、デイジーチェーン出力を行うための１個の出力端子１２と、が設けられている。従来のＡＶセンターユニットのように、スピーカー２ごとに出力端子を備えていた構成と比較すると、出力端子１２が１個だけのシンプルな構成となっており、これにより筐体の小型化を図ることができる。 Next, with reference to FIG. 3, device configurations of the audio transmission device 1 and the speaker 2 will be described. FIG. 3A is a rear view of the audio transmission device 1. On the rear surface of the audio transmission device 1, a power supply port 11 that receives power supply from an AC power supply and one output terminal 12 for performing daisy chain output are provided. Compared with the configuration in which each speaker 2 has an output terminal as in the conventional AV center unit, the output terminal 12 has a simple configuration, thereby reducing the size of the housing. Can do.
　図３（ｂ）は、スピーカー２の前面図および背面図である。スピーカー２の前面には、振動部材２１が設けられている。また、背面には、デイジーチェーン入出力を行うための各１個の入力端子２２および出力端子２３が設けられている。なお、デイジーチェーン接続される全てのスピーカー２の背面構成は同じである。 FIG. 3B is a front view and a rear view of the speaker 2. A vibration member 21 is provided on the front surface of the speaker 2. Also, on the back side, one input terminal 22 and one output terminal 23 for performing daisy chain input / output are provided. The rear configuration of all speakers 2 connected in a daisy chain is the same.
　次に、図４を参照し、第１実施形態に係るスピーカーシステムＳＹの機能構成について説明する。第１実施形態では、主に多チャンネル再生に関する機能について説明する。なお、同図の矢印は、音声データの流れを示している。音声伝送装置１は、主な機能構成として、ＩＤ初期化部１０１、スピーカーアサイン設定部１０２、音声入力部１０３、チャンネル情報取得部１０４、チャンネル情報設定部１０５および伝送制御部１０６を備えている。 Next, the functional configuration of the speaker system SY according to the first embodiment will be described with reference to FIG. In the first embodiment, functions related to multi-channel playback will be mainly described. In addition, the arrow of the figure has shown the flow of audio | voice data. The audio transmission device 1 includes an ID initialization unit 101, a speaker assignment setting unit 102, an audio input unit 103, a channel information acquisition unit 104, a channel information setting unit 105, and a transmission control unit 106 as main functional configurations.
　ＩＤ初期化部１０１は、デイジーチェーン接続されたスピーカー２の構成を認識させるための初期設定を行う。ＩＤ初期化部１０１は、各スピーカー２に対して固有のスピーカーＩＤを発行すると共に、発行したスピーカーＩＤを発行順序にしたがって内部メモリ（図示省略）に記憶する。詳細については、図５にて後述する。 The ID initialization unit 101 performs initial setting for recognizing the configuration of the speakers 2 connected in a daisy chain. The ID initialization unit 101 issues a unique speaker ID to each speaker 2 and stores the issued speaker ID in an internal memory (not shown) according to the issue order. Details will be described later with reference to FIG.
　スピーカーアサイン設定部１０２は、ユーザーに対し、スピーカー２のチャンネルアサインを促すアサイン処理を行う。なお、音声伝送装置１は、ディスプレイ、並びに操作パネルやリモートコントローラー等のユーザーインターフェースが設けられているものとする。本実施形態では、ゾーン（マルチ試聴エリア）ごとに該当するスピーカー２をアサイン可能となっている。詳細については、図７および図８にて後述する。 The speaker assignment setting unit 102 performs assignment processing that prompts the user to assign a channel to the speaker 2. Note that the audio transmission device 1 is provided with a display and a user interface such as an operation panel or a remote controller. In the present embodiment, the corresponding speaker 2 can be assigned to each zone (multi-listening area). Details will be described later with reference to FIGS.
　音声入力部１０３は、各音声入力源４から音声データ（オーディオサンプリングデータ）を入力する。なお、各音声入力源４から、それぞれ複数チャンネルの音声データを入力することも可能である。 The voice input unit 103 inputs voice data (audio sampling data) from each voice input source 4. It is also possible to input a plurality of channels of audio data from each audio input source 4.
　チャンネル情報取得部１０４は、再生コンテンツ内にチャンネル情報が埋め込まれている場合、これを取得する。当該チャンネル情報とは、コンテンツの再生チャンネル数および間引き対象／間引き非対象となるチャンネルを示す情報である。後に詳述するが、本実施形態では、各スピーカー２に送信する音声データを時間軸方向で間引くことで、伝送周波数を上げることなく、従来よりも伝送許容チャンネル数を増やした音声再生を実現している。したがって、「間引き対象となるチャンネル」とは、例えば１３チャンネルを実現する場合の「第４～第１３チャンネル」などと設定される。この場合、第４～第１３の合計１０チャンネルを、第４～第８と第９～第１３との２グループに分割して交互に音声データを伝送する。これにより、最大８チャンネル分のデータ伝送量で、１３チャンネルのコンテンツ再生を実現する。 If the channel information is embedded in the playback content, the channel information acquisition unit 104 acquires it. The channel information is information indicating the number of content reproduction channels and channels to be thinned / not thinned. As will be described in detail later, in this embodiment, the audio data to be transmitted to each speaker 2 is thinned out in the time axis direction, thereby realizing audio reproduction with an increased number of allowable transmission channels than before without increasing the transmission frequency. ing. Accordingly, the “channel to be thinned out” is set to, for example, “fourth to thirteenth channels” when 13 channels are realized. In this case, the fourth to thirteenth total 10 channels are divided into two groups of fourth to eighth and ninth to thirteenth, and audio data is transmitted alternately. As a result, 13-channel content reproduction is realized with a data transmission amount for a maximum of 8 channels.
　チャンネル情報設定部１０５は、音声伝送装置１に設けられたユーザーインターフェースを用いて、チャンネル情報（間引き対象／間引き非対象となるチャンネル）を設定する。なお、再生コンテンツ内にチャンネル情報が埋め込まれている場合であってチャンネル情報設定部１０５による設定も行われている場合、チャンネル情報取得部１０４にて取得したチャンネル情報を優先しても良い。また、どちらを優先させるかをユーザーが設定可能としても良いし、再生コンテンツ内にチャンネル情報が埋め込まれていない場合のみ、ユーザーにチャンネル情報の設定を促す構成としても良い。後者の場合、以下のような設定方法が考えられる。まず、ユーザーのアサイン処理によって使用するスピーカー２の数が設定されると、音声伝送装置１は、コンテンツに含まれる再生チャンネル数との比較により、分割伝送が必要か否かを判別する。分割伝送が必要と判定した場合は、分割させたくないスピーカー２または分割させても良いスピーカー２の選択をユーザーに促す。ここで、ユーザーが分割させても良いスピーカー２を選択する都度、音声伝送装置１は、伝送許容レート内に収まるか否かを判別し（分割対象となるスピーカー２数が、伝送可能な必要個数となったか否かを判別し）、伝送許容レート内に収まった時点で、分割伝送設定の完了をユーザーに通知する。このように、ユーザーがチャンネル情報を設定することにより、視聴環境に合った視聴環境設定を行うことができる。 The channel information setting unit 105 sets channel information (thinning target / non-thinning target channel) using a user interface provided in the audio transmission device 1. Note that when channel information is embedded in the playback content and setting is also performed by the channel information setting unit 105, the channel information acquired by the channel information acquisition unit 104 may be prioritized. In addition, the user may be able to set which is prioritized, or may be configured to prompt the user to set channel information only when channel information is not embedded in the playback content. In the latter case, the following setting method can be considered. First, when the number of speakers 2 to be used is set by the user assignment process, the audio transmission device 1 determines whether division transmission is necessary by comparing with the number of reproduction channels included in the content. When it is determined that the divided transmission is necessary, the user is prompted to select the speaker 2 that is not desired to be divided or the speaker 2 that may be divided. Here, every time the user selects a speaker 2 that may be divided, the audio transmission device 1 determines whether or not the transmission rate is within the allowable transmission rate (the number of speakers 2 to be divided is the necessary number that can be transmitted). When the result falls within the allowable transmission rate, the user is notified of the completion of the divided transmission setting. As described above, the user can set the viewing environment suitable for the viewing environment by setting the channel information.
　伝送制御部１０６は、複数の音声入力源４から入力された複数の音声データを、１フレームにパッキングして伝送する。例えば図１２では、チャンネル１～８の音声データを１フレーム、チャンネル１～３，９～１３を１フレームとして、各フレーム構成を交互に伝送する場合を示している。具体的には、デイジーチェーン接続された複数のスピーカー２の少なくとも一部を２個のスピーカー群に分割し、分割した各スピーカー群に対して、２回のサンプリングのうち１回の割合で音声データを順次伝送する（間欠伝送制御を行う）。例えば図１２では、チャンネル４～８を第１グループ、チャンネル９～１３を第２グループとしてグループ分けしている。この場合、奇数回目のサンプリングでは、元のコンテンツに存在していたデータからチャンネル９～１３に対応するデータを破棄し、偶数回目のサンプリングでは、チャンネル４～８に対応するデータを破棄する。なお、チャンネル１～３はグループ分けの対象とならない非分割グループに属する。これら非分割グループのスピーカー２に対しては、常時音声データを伝送する。この非分割グループでは、フロントＬ，Ｒ、センターチェンネルなどユーザーがクオリティの差異を感じやすい比較的重要なスピーカーなどを設定する。 The transmission control unit 106 packs a plurality of audio data input from the plurality of audio input sources 4 into one frame for transmission. For example, FIG. 12 shows a case where the audio data of channels 1 to 8 is 1 frame, and channels 1 to 3 and 9 to 13 are 1 frame, and each frame configuration is transmitted alternately. Specifically, at least a part of the plurality of speakers 2 connected in a daisy chain is divided into two speaker groups, and the audio data is divided into two pieces of sampling for each divided speaker group. Are transmitted sequentially (intermittent transmission control is performed). For example, in FIG. 12, channels 4 to 8 are grouped as a first group, and channels 9 to 13 are grouped as a second group. In this case, in the odd-numbered sampling, data corresponding to channels 9 to 13 is discarded from the data existing in the original content, and in the even-numbered sampling, data corresponding to channels 4 to 8 is discarded. Channels 1 to 3 belong to non-divided groups that are not grouped. Audio data is always transmitted to the speakers 2 of these non-divided groups. In this non-divided group, relatively important speakers such as front L, R, and center channel, which are easy for the user to feel a difference in quality, are set.
　一方、スピーカー２は、主な制御構成として、ＩＤ記憶部２０１、データ選択取得部２０２、デイジーチェーン出力部２０３、データバッファ２０４、補間データ生成部２０５、選択回路２０６および音声出力部２０７を備えている。 On the other hand, the speaker 2 includes an ID storage unit 201, a data selection acquisition unit 202, a daisy chain output unit 203, a data buffer 204, an interpolation data generation unit 205, a selection circuit 206, and an audio output unit 207 as main control configurations. Yes.
　ＩＤ記憶部２０１は、ＩＤ初期化部１０１により発行されたスピーカーＩＤを記憶するものであり、不揮発性のメモリにより実現される。データ選択取得部２０２は、音声伝送装置１から伝送されたフレームから、スタートコードパケットや自スピーカー２用の音声データを選択取得する。なお、自スピーカー２用であるか否かは、ＩＤ記憶部２０１に記憶されたスピーカーＩＤを用いて判別する。 The ID storage unit 201 stores the speaker ID issued by the ID initialization unit 101, and is realized by a nonvolatile memory. The data selection / acquisition unit 202 selects and acquires a start code packet and audio data for the own speaker 2 from the frame transmitted from the audio transmission device 1. Whether the speaker 2 is used is determined by using the speaker ID stored in the ID storage unit 201.
　デイジーチェーン出力部２０３は、下流側に接続された次段のスピーカー２に対してデイジー出力を行う。つまり、上流側のケーブル３から伝送されたフレームを下流側のケーブル３へ出力する。 The daisy chain output unit 203 performs daisy output to the next-stage speaker 2 connected downstream. That is, the frame transmitted from the upstream cable 3 is output to the downstream cable 3.
　データバッファ２０４は、連続する３サンプリング分の音声データを、各部分バッファ２０４ａ，２０４ｂ，２０４ｃに格納する。補間データ生成部２０５は、データバッファ２０４の中間部分バッファ２０４ｂのデータが欠けている場合、その前後の部分バッファ２０４ａ，２０４ｃに格納されたリアルデータ（実際に伝送された音声データ）を、例えば足して１／２倍することで補間データを生成する。なお、補間データ生成部２０５は、データの欠けを、フレームのスタートコードパケットに含まれるチャンネル設定フラグに基づいて判別する。つまり、チャンネル設定フラグが「無効」を示す場合、そのタイミングでの音声データは欠けていると判別し、補間データを生成する。 The data buffer 204 stores audio data for three consecutive samples in the partial buffers 204a, 204b, and 204c. When the data in the intermediate partial buffer 204b of the data buffer 204 is missing, the interpolation data generation unit 205 adds the real data (actually transmitted audio data) stored in the preceding and subsequent partial buffers 204a and 204c, for example. Interpolation data is generated by halving. The interpolation data generation unit 205 determines the lack of data based on the channel setting flag included in the start code packet of the frame. That is, when the channel setting flag indicates “invalid”, it is determined that the audio data at that timing is missing, and interpolation data is generated.
　選択回路２０６は、チャンネル設定フラグに応じて、補間データを選択するかリアルデータを選択するかを選択する。チャンネル設定フラグが「無効」を示す場合、補間データを選択し、チャンネル設定フラグが「有効」を示す場合、データバッファ２０４に格納されているリアルデータを選択する。音声出力部２０７は、選択回路２０６の選択結果に基づいて、リアルデータまたは補間データを出力する。なお、本実施形態の場合（２グループに分割して伝送を行う場合）、分割対象となるスピーカー２は、リアルデータおよび補間データを交互に出力する。 The selection circuit 206 selects whether to select interpolation data or real data according to the channel setting flag. When the channel setting flag indicates “invalid”, the interpolation data is selected. When the channel setting flag indicates “valid”, the real data stored in the data buffer 204 is selected. The audio output unit 207 outputs real data or interpolation data based on the selection result of the selection circuit 206. In the case of the present embodiment (when transmission is performed by dividing into two groups), the speaker 2 to be divided alternately outputs real data and interpolation data.
　次に、図５を参照し、デイジーチェーンＩＤ初期化処理について説明する。音声伝送装置１は、電源ＯＮすると（Ｓ１１，但し各スピーカー２はデイジーアウトＯＦＦ状態であるものとする）、先頭のスピーカー２（最上流のスピーカー２）にアクセスする（Ｓ１２）。先頭のスピーカー２からＡＣＫ信号を受信した場合は（Ｓ１３：Ｙｅｓ）、そのスピーカー２のＩＤ記憶部２０１をクリアにして（Ｓ１４）、スピーカーＩＤを付与する（Ｓ１５）。さらに、そのスピーカー２に対してデイジーアウトのアクティブ設定を行い（Ｓ１６）、次の（下流側の）スピーカー２にアクセスする（Ｓ１７）。当該次のスピーカー２からもＡＣＫ信号を受信した場合は（Ｓ１８：Ｙｅｓ）、Ｓ１４以降を繰り返す。 Next, the daisy chain ID initialization process will be described with reference to FIG. When the power is turned on (S11, where each speaker 2 is in a daisy-out OFF state), the audio transmission device 1 accesses the head speaker 2 (the most upstream speaker 2) (S12). When an ACK signal is received from the head speaker 2 (S13: Yes), the ID storage unit 201 of the speaker 2 is cleared (S14), and a speaker ID is assigned (S15). Further, daisy-out active setting is performed for the speaker 2 (S16), and the next (downstream) speaker 2 is accessed (S17). When an ACK signal is received from the next speaker 2 (S18: Yes), the steps after S14 are repeated.
　一方、Ｓ１３：Ｎｏの場合およびＳ１８：Ｎｏの場合（ＡＣＫ信号を受信することなくタイムアウトした場合）、デイジーアウトのインアクティブ設定を行う（Ｓ１９）。その後、全チェーンの初期化を終了したか否かを判別し（Ｓ２０）、終了した場合は（Ｓ２０：Ｙｅｓ）、デイジーチェーンＩＤ初期化処理を終了する。また、Ｓ２０：Ｎｏの場合は、次のチェーンについて、Ｓ１２以降を繰り返す（Ｓ２１）。なお、Ｓ２０，Ｓ２１の工程は、１本のデイジーチェーン構成の場合（図１，図２の構成の場合）、省略可能である。 On the other hand, in the case of S13: No and in the case of S18: No (when a time-out occurs without receiving an ACK signal), daisy-out inactive setting is performed (S19). Thereafter, it is determined whether or not the initialization of all the chains has been completed (S20), and if completed (S20: Yes), the daisy chain ID initialization process is terminated. If S20: No, S12 and subsequent steps are repeated for the next chain (S21). Note that the steps S20 and S21 can be omitted in the case of a single daisy chain configuration (in the case of the configuration of FIGS. 1 and 2).
　次に、図６を参照し、パワーオン処理について説明する。ここでは、デイジーチェーンＩＤ初期化設定後、一度電源ＯＦＦした後に電源ＯＮした場合の処理について説明する。音声伝送装置１は、電源ＯＮすると（但し各スピーカーは、デイジーアウトＯＦＦ状態で起動するものとする）、先頭のスピーカー２にアクセスし（Ｓ３１）、ＡＣＫ信号の受信を待つ。ＡＣＫ信号を受信した場合は（Ｓ３２：Ｙｅｓ）、そのスピーカー２のスピーカーＩＤを取得し（Ｓ３３）、期待値であるか否かを判別する（Ｓ３４）。つまり、記憶しておいた順序どおりにスピーカーＩＤを取得したか否かを判別する。ここで、スピーカーＩＤが期待値でなかった場合は（Ｓ３４：Ｎｏ）、スピーカー２の入れ替えや順序変更が行われた可能性があるため、図５に示したデイジーチェーンＩＤ初期化処理を行う（Ｓ３５）。 Next, the power-on process will be described with reference to FIG. Here, processing when the power is turned on after the power is turned off once after the daisy chain ID initialization setting will be described. When the power is turned on (assuming that each speaker is activated in the daisy-out OFF state), the audio transmission device 1 accesses the head speaker 2 (S31) and waits for reception of an ACK signal. When the ACK signal is received (S32: Yes), the speaker ID of the speaker 2 is acquired (S33), and it is determined whether or not the value is an expected value (S34). That is, it is determined whether or not the speaker IDs are acquired in the stored order. If the speaker ID is not the expected value (S34: No), there is a possibility that the speaker 2 has been replaced or the order has been changed, so the daisy chain ID initialization process shown in FIG. 5 is performed ( S35).
　また、スピーカーＩＤが期待値であった場合は（Ｓ３４：Ｙｅｓ）、そのスピーカー２に対してデイジーアウトのアクティブ設定を行い（Ｓ３６）、次のスピーカー２にアクセスする（Ｓ３７）。当該次のスピーカー２からもＡＣＫ信号を受信した場合は（Ｓ３８：Ｙｅｓ）、Ｓ３３以降を繰り返す。また、Ｓ３２：Ｎｏ，Ｓ３８：Ｎｏの場合（ＡＣＫ信号を受信することなくタイムアウトした場合）、デイジーアウトのインアクティブ設定を行い（Ｓ３９）、パワーオン処理を終了する。なお、次のチェーンが存在する場合は、次のチェーンのパワーオン処理を行う。 If the speaker ID is the expected value (S34: Yes), daisy-out active setting is performed for the speaker 2 (S36), and the next speaker 2 is accessed (S37). When an ACK signal is also received from the next speaker 2 (S38: Yes), S33 and subsequent steps are repeated. If S32: No, S38: No (timeout without receiving an ACK signal), daisy-out inactive setting is performed (S39), and the power-on process is terminated. If the next chain exists, the power-on process for the next chain is performed.
　次に、図７および図８を参照し、スピーカー２のアサイン処理について説明する。図７（ａ）は、ゾーン数の設定処理を示すフローチャートである。音声伝送装置１は、まずユーザーに対し、ゾーン数の入力を促す（Ｓ４１）。具体的には、ディスプレイへの表示または音声出力によって、入力が必要である旨をユーザーに通知する。ここで、ゾーン数が入力された場合は（Ｓ４２：Ｙｅｓ）、その値が有効であるか否かを判別し（Ｓ４３）、有効である場合は（Ｓ４３：Ｙｅｓ）、設定値を内部メモリに記憶する（Ｓ４４）。また、Ｓ４２：Ｎｏの場合は入力待機し、タイムアウトした場合は（Ｓ４５：Ｙｅｓ）、エラー処理を行う（Ｓ４６）。また、タイムアウトしていなければ（Ｓ４５：Ｎｏ）、Ｓ４２に戻る。さらに、入力されたゾーン数が無効な値である場合は（Ｓ４３：Ｎｏ）、エラー処理を行う（Ｓ４６）。 Next, the assignment process of the speaker 2 will be described with reference to FIGS. FIG. 7A is a flowchart showing the zone number setting process. The voice transmission device 1 first prompts the user to input the number of zones (S41). Specifically, the user is notified that input is required by display on the display or audio output. If the number of zones is input (S42: Yes), it is determined whether or not the value is valid (S43). If it is valid (S43: Yes), the set value is stored in the internal memory. Store (S44). If S42: No, input standby is performed, and if a time-out occurs (S45: Yes), error processing is performed (S46). If the time has not expired (S45: No), the process returns to S42. Furthermore, when the input zone number is an invalid value (S43: No), error processing is performed (S46).
　図７（ｂ）は、チャンネル数（スピーカー２の数）の設定処理を示すフローチャートである。当該処理では、まずユーザーに対し、チャンネル数の入力を促す（Ｓ５１）。ここでも、ディスプレイへの表示や音声出力によって、入力が必要である旨をユーザーに通知する。ここで、チャンネル数が入力された場合は（Ｓ５２：Ｙｅｓ）、その値が有効であるか否かを判別し（Ｓ５３）、有効である場合は（Ｓ５３：Ｙｅｓ）、設定値を内部メモリに記憶する（Ｓ５４）。また、Ｓ５２：Ｎｏの場合は入力待機し、タイムアウトした場合は（Ｓ５５：Ｙｅｓ）、エラー処理を行う（Ｓ５６）。また、タイムアウトしていなければ（Ｓ５５：Ｎｏ）、Ｓ５２に戻る。さらに、入力されたチャンネル数が無効な値である場合は（Ｓ５３：Ｎｏ）、エラー処理を行う（Ｓ５６）。 FIG. 7B is a flowchart showing the setting process of the number of channels (number of speakers 2). In this process, first, the user is prompted to input the number of channels (S51). In this case as well, the user is notified that input is required by display on the display or audio output. If the number of channels is input (S52: Yes), it is determined whether the value is valid (S53). If it is valid (S53: Yes), the set value is stored in the internal memory. Store (S54). If S52: No, input standby is performed, and if a time-out occurs (S55: Yes), error processing is performed (S56). If the time has not expired (S55: No), the process returns to S52. Furthermore, when the input channel number is an invalid value (S53: No), error processing is performed (S56).
　図８は、スピーカーアサイン設定処理を示すフローチャートである。当該処理は、図７（ａ），（ｂ）の設定処理後に行われるものとする。音声伝送装置１は、ＩＤ設定済み且つアサイン未設定のスピーカー２が存在するか否かを判別する（Ｓ６１）。存在しない場合は（Ｓ６１：Ｎｏ）、エラー処理を行う（Ｓ６２）。また、ＩＤ設定済み且つアサイン未設定のスピーカー２が存在する場合は（Ｓ６１：Ｙｅｓ）、設定するスピーカーＩＤ（最初は「１」）をセットし（Ｓ６３）、順次、アサインループ処理を開始する（Ｓ６４）。 FIG. 8 is a flowchart showing the speaker assignment setting process. This processing is assumed to be performed after the setting processing of FIGS. 7 (a) and 7 (b). The audio transmission device 1 determines whether or not there is a speaker 2 with an ID set and an unassigned setting (S61). If it does not exist (S61: No), error processing is performed (S62). If there is a speaker 2 with an ID set and no assignment (S61: Yes), the speaker ID to be set (initially “1”) is set (S63), and the assignment loop process is started sequentially (S63). S64).
　アサインループ処理では、まず設定対象となるスピーカー２から識別信号（音あるいはインジケータ（ＬＥＤ））を出力し（Ｓ６５）ユーザーに、現在どこに配置しているスピーカー２の設定を行うかを認識させ、スピーカーアサインの入力を促す（Ｓ６６）。ここでは、音声伝送装置１に接続された表示装置（ＴＶモニターなど）上でのＧＵＩを用いてゾーン番号を入力させる。また、識別信号の発生は、識別信号Gen６７（図２８参照）にて行われる。 In the assignment loop processing, first, an identification signal (sound or indicator (LED)) is output from the speaker 2 to be set (S65), and the user is made aware of where to set the speaker 2 currently arranged. The user is prompted to input an assignment (S66). Here, a zone number is input using a GUI on a display device (such as a TV monitor) connected to the audio transmission device 1. The generation of the identification signal is performed by the identification signal Gen67 (see FIG. 28).
　ユーザーにより、スピーカーアサインが入力された場合は（Ｓ６７：Ｙｅｓ）、スピーカー２の識別信号の出力を停止し（Ｓ６８）、有効な値であるか否かを判別する（Ｓ６９）。有効な値である場合は（Ｓ６９：Ｙｅｓ）、設定値を内部メモリに記憶し（Ｓ７０）、アサインループ処理を終了する（Ｓ７３）。また、スピーカーアサインが入力されず（Ｓ６７：Ｎｏ）、タイムアウトした場合は（Ｓ７１：Ｙｅｓ）、エラー処理を行う（Ｓ７２）。さらに、入力されたスピーカーアサインが有効な値でなかった場合も（Ｓ６９：Ｎｏ）、エラー処理を行う（Ｓ７２）。 When the speaker assignment is input by the user (S67: Yes), the output of the identification signal of the speaker 2 is stopped (S68), and it is determined whether or not it is a valid value (S69). If it is a valid value (S69: Yes), the set value is stored in the internal memory (S70), and the assignment loop process is terminated (S73). If no speaker assignment is input (S67: No) and time-out occurs (S71: Yes), error processing is performed (S72). Further, if the input speaker assignment is not a valid value (S69: No), error processing is performed (S72).
　アサインループ処理を終了した後は、設定するスピーカーＩＤをインクリメントし（Ｓ７４）、そのＩＤが有効である場合は（全てのスピーカー２のアサインが完了していない場合は，Ｓ７５：Ｙｅｓ）、Ｓ６４以降を繰り返す。また、ＩＤが無効である場合は（全てのスピーカー２のアサインを完了した場合は，Ｓ７５：Ｎｏ）、スピーカーアサイン設定処理を終了する。 After completing the assignment loop process, the speaker ID to be set is incremented (S74). If the ID is valid (S75: Yes if assignment of all speakers 2 has not been completed), S64 and thereafter. repeat. If the ID is invalid (when assignment of all speakers 2 is completed, S75: No), the speaker assignment setting process is terminated.
　次に、図９を参照し、デイジーチェーン伝送に用いるデータプロトコルの一例について説明する。本実施形態では、同図に示すように３本の信号を用いてデータ伝送を行う。また、複数チャンネルのデータをひとまとめにし、１フレームとしてデータ伝送を行う。例えば、Sync_data（スタートコードパケット）からSync_dataの間に、各チャンネル最大３２ビットのデータを埋め込む。各チャンネルの区切りは、ch_div信号で区切る。各チャンネルに対応するスピーカー２は、各チャンネルのデータを、毎フレーム取り込み、Sync_dataで同期のタイミングを測り、clkにて同期タイミングをとる。なお、Sync_dataとしては特定のパターンを予め決めておくことで検出する（図１５（ａ）等参照）。なお、同図のデータプロトコルは一例であり、他のデータプロトコルを用いても良い。また、ビット長も任意であり、４８，６４ビット等のデータを埋め込んでも良い。 Next, an example of a data protocol used for daisy chain transmission will be described with reference to FIG. In this embodiment, data transmission is performed using three signals as shown in FIG. In addition, data of a plurality of channels is collected and data transmission is performed as one frame. For example, data of up to 32 bits for each channel is embedded between Sync_data (start code packet) and Sync_data. Each channel is separated by a ch_div signal. The speaker 2 corresponding to each channel captures the data of each channel every frame, measures the synchronization timing with Sync_data, and takes the synchronization timing with clk. Sync_data is detected by determining a specific pattern in advance (see FIG. 15A, etc.). In addition, the data protocol of the figure is an example, and other data protocols may be used. The bit length is also arbitrary, and data such as 48, 64 bits may be embedded.
　次に、図１０を参照し、音声（オーディオ）データシリアル伝送について説明する。同図において、“Ｓ”は、Sync_data、各数字は、チャンネル番号を示している。同図に示すように、各チャンネルの音声データ（Nomal_data）は、時系列で伝送される。また、全チャンネル数分の音声データ（同図の例では、８チャンネル）は、次のSync_dataが送られてくるまでの間（１／ｆｓ内）に全チャンネル数分の音声データの伝送を完了する必要がある。この全チャンネル数分の音声データが揃ったところで、各スピーカー２を同期して駆動させることで、その瞬間ごとの（サンプリング周波数単位の）音場再生が可能となる。 Next, voice (audio) data serial transmission will be described with reference to FIG. In the figure, “S” indicates Sync_data, and each number indicates a channel number. As shown in the figure, the audio data (Nomal_data) of each channel is transmitted in time series. Also, audio data for all channels (8 channels in the example in the figure) has been transmitted for all channels until the next Sync_data is sent (within 1 / fs). There is a need to. When the audio data for all the channels has been prepared, each speaker 2 is driven in synchronization, so that sound field reproduction can be performed for each moment (in sampling frequency units).
　次に、図１１ないし図１３を参照し、第１実施形態に係る音声データシリアル伝送について説明する。図１１は、チャンネル設定フラグの説明図である。同図に示すように、Sync_data内には、ペイロードやチャンネル設定フラグが含まれる。ペイロードには、以下に続くデータの各種情報が含まれる。また、チャンネル設定フラグは、チャンネルごと（スピーカー２）ごとに有効／無効を示すものである。当該チャンネル設定フラグの設定（組み込み）は、伝送制御部１０６にて行われる。スピーカー２（補間データ生成部２０５）は、このチャンネル設定フラグの設定にしたがって、伝送データを認識し、補間が必要と判断した場合（自スピーカー２用のチャンネル設定フラグが「無効」を示す場合）、補間データの生成を行う。 Next, the audio data serial transmission according to the first embodiment will be described with reference to FIGS. FIG. 11 is an explanatory diagram of a channel setting flag. As shown in the figure, the Sync_data includes a payload and a channel setting flag. The payload includes various types of data information that follows. The channel setting flag indicates validity / invalidity for each channel (speaker 2). The setting (incorporation) of the channel setting flag is performed by the transmission control unit 106. Speaker 2 (interpolation data generation unit 205) recognizes transmission data according to the setting of the channel setting flag and determines that interpolation is necessary (when the channel setting flag for the speaker 2 indicates “invalid”). Generate interpolation data.
　図１２は、本実施形態に係る音声データシリアル伝送の概念図である。上記の通り本実施形態では、各チャンネルの音声データをシリアル伝送するに当たり、一部のチャンネルに対し、間欠的にデータ伝送を行う。図１２の例では、全１３チャンネルのうち、チャンネル４～８を第１グループ、チャンネル９～１３を第２グループとして分割し、各グループについて交互にデータ伝送を行う。なお、チャンネル１～３は、グループ分けの対象（間欠伝送の対象）とならない非分割グループに属する。この場合、奇数回目のフレームとして、非分割グループ＋第１グループのデータを伝送し、偶数回目のフレームとして非分割グループ＋第２グループのデータを伝送する。 FIG. 12 is a conceptual diagram of audio data serial transmission according to the present embodiment. As described above, in the present embodiment, when audio data of each channel is serially transmitted, data transmission is intermittently performed for some channels. In the example of FIG. 12, among all 13 channels, channels 4 to 8 are divided into a first group and channels 9 to 13 are divided into a second group, and data transmission is performed alternately for each group. Channels 1 to 3 belong to a non-divided group that is not a grouping target (intermittent transmission target). In this case, the non-divided group + first group data is transmitted as an odd-numbered frame, and the non-divided group + second group data is transmitted as an even-numbered frame.
　一方、スピーカー２側では、例えば、第２フレームの音声出力タイミングでは、非分割グループおよび第２グループの各チャンネルが、フレームとして伝送された音声データであるリアルデータを再生する。これに対し、第１グループの各チャンネル（チャンネル４～８）は、第１フレームで伝送された音声データと第３フレームで伝送された音声データとを、例えば足して１／２倍することで補間データを生成し、これを第２フレームの音声出力タイミングに同期して出力する。また、第３フレームの音声出力タイミングでは、非分割グループおよび第１グループの各チャンネルがリアルデータを再生する。これに対し、第２グループの各チャンネル（チャンネル９～１３）は、第２フレームで伝送された音声データと第４フレームで伝送された音声データとを、例えば足して１／２倍することで補間データを生成し、これを第３フレームの音声出力タイミングに同期して出力する。これを繰り返すことにより、各フレーム単位で見ると常時８チャンネル分しか伝送していないが、第１グループおよび第２グループにおいて、リアルデータと補間データを交互に出力することで全１３チャンネルの常時再生が可能となる。 On the other hand, on the speaker 2 side, for example, at the audio output timing of the second frame, each channel of the non-divided group and the second group reproduces real data that is audio data transmitted as a frame. On the other hand, each channel (channels 4 to 8) of the first group is obtained by multiplying, for example, the audio data transmitted in the first frame and the audio data transmitted in the third frame by a factor of 1/2. Interpolated data is generated and output in synchronization with the audio output timing of the second frame. In addition, at the audio output timing of the third frame, each channel of the non-divided group and the first group reproduces real data. On the other hand, each channel (channels 9 to 13) of the second group is obtained by multiplying, for example, the audio data transmitted in the second frame and the audio data transmitted in the fourth frame by a factor of 1/2. Interpolated data is generated and output in synchronization with the audio output timing of the third frame. By repeating this, only 8 channels are always transmitted when viewed in units of frames. However, in the first group and the second group, real data and interpolated data are alternately output to constantly reproduce all 13 channels. Is possible.
　図１３は、本実施形態に係る再生処理を示すフローチャートである。スピーカー２は、フレームに含まれるSync_dataを取得すると（Ｓ８１）、チャンネル設定フラグが「１：有効」を示すか否かを判別し（Ｓ８２）、「１：有効」を示す場合は（Ｓ８２：Ｙｅｓ）、選択回路２０６にてリアルデータを選択する（Ｓ８３）。一方、「０：無効」を示す場合は（Ｓ８２：Ｎｏ）、補間データを生成・選択する（Ｓ８４）。Ｓ８３およびＳ８４により再生データの準備が完了すると（Ｓ８５）、これを再生する（Ｓ８６）。 FIG. 13 is a flowchart showing the reproduction processing according to the present embodiment. When the speaker 2 acquires Sync_data included in the frame (S81), the speaker 2 determines whether or not the channel setting flag indicates “1: valid” (S82), and when “1: valid” is indicated (S82: Yes). ), Real data is selected by the selection circuit 206 (S83). On the other hand, when “0: invalid” is indicated (S82: No), interpolation data is generated and selected (S84). When the preparation of the reproduction data is completed by S83 and S84 (S85), it is reproduced (S86).
　以上説明したとおり、第１実施形態によれば、音声伝送装置１側において、分割対象となる各スピーカー群に対し交互にデータを伝送し、スピーカー２側では、自スピーカー２で取得した時間軸上の前後の音声データから、未送信の音声データを補間するための補間データを生成するため、搬送周波数を上げることなく、再生可能なスピーカー数（チャンネル数）を増やすことができる。また、両隣のスピーカー２のサンプリングデータから、その間のスピーカー２の補間データを生成する方式ではないため、複数の楽曲再生を行う場合や隣り合うチャンネルの相関関係が低い場合であっても、音場の定位感を保ったまま、多チャンネル数分の音声データを伝送することができる。また、ディスクメディアなどの規格上で決められているチャンネル数上限の制限を超えた多チャンネルのコンテンツ作成が可能となり、コンテンツ作成者の、より多チャンネルで表現したいという要望に応えることができる。 As described above, according to the first embodiment, on the audio transmission apparatus 1 side, data is alternately transmitted to each speaker group to be divided, and on the speaker 2 side, on the time axis acquired by the own speaker 2. Since interpolated data for interpolating untransmitted audio data is generated from the audio data before and after the above, the number of reproducible speakers (number of channels) can be increased without increasing the carrier frequency. Further, since it is not a method of generating interpolated data of the speaker 2 between the sampling data of the adjacent speakers 2, the sound field can be obtained even when a plurality of music reproductions are performed or when the correlation between adjacent channels is low. Thus, it is possible to transmit audio data for a number of channels while maintaining a sense of localization. In addition, it is possible to create multi-channel content that exceeds the upper limit on the number of channels determined by the standards of disk media and the like, and it is possible to meet the demand of content creators to express with more channels.
　また、Sync_dataには、スピーカー２ごとに有効／無効を示すチャンネル設定フラグが定義されているため、スピーカー２側で、補間データを生成すべきか否かを正確に判定することができる。また、常時サンプリングデータを伝送する非分割グループを設定できるため、音質の低下を防止できる。特に、フロントスピーカーなど、直接音で重要なスピーカーを非分割のスピーカーとして設定することで、聴感上の音質影響を抑えることができる。 Also, since the channel setting flag indicating validity / invalidity is defined for each speaker 2 in Sync_data, it is possible to accurately determine whether interpolation data should be generated on the speaker 2 side. In addition, since a non-divided group that constantly transmits sampling data can be set, it is possible to prevent deterioration in sound quality. In particular, by setting speakers that are important for direct sound, such as front speakers, as non-divided speakers, it is possible to suppress the influence of sound quality on hearing.
　なお、上記の実施形態では、全チャンネルの一部を２つのグループに分割し、交互に音声データの伝送を行ったが、３個以上のグループ（Ｎ個のグループ）に分割しても良い。この場合、分割対象となるグループ内の各スピーカー２に対し、常時伝送時に比べ１／Ｎ回の割合で音声データが伝送される。また、スピーカー２側には、バッファとして（Ｎ＋１）個の部分バッファを備え、中間部分バッファが空の場合、前後の部分バッファ内のデータを用いて補間データを生成する。 In the above embodiment, a part of all the channels is divided into two groups and the audio data is transmitted alternately. However, it may be divided into three or more groups (N groups). In this case, audio data is transmitted to each speaker 2 in the group to be divided at a rate of 1 / N times as compared with the case of constant transmission. On the speaker 2 side, (N + 1) partial buffers are provided as buffers, and when the intermediate partial buffer is empty, interpolation data is generated using data in the preceding and subsequent partial buffers.
　また、上記の実施形態では、ユーザーがチャンネル情報を設定する場合、ユーザーが分割させても良いスピーカー２を選択する都度、音声伝送装置１が、伝送許容レート内に収まるか否かを判別し、伝送許容レート内に収まった時点で、分割伝送設定の完了をユーザーに通知するものとしたが（Ｎ＝２を前提としたが）、ユーザーが分割させても良いスピーカー２を任意数選択した段階で、音声伝送装置１が、伝送許容レート内に収まるように分割数（Ｎグループに分割する場合のＮの数）を決定する構成としても良い。 In the above embodiment, when the user sets the channel information, each time the user selects the speaker 2 that may be divided, it is determined whether or not the audio transmission device 1 falls within the allowable transmission rate. When the transmission rate falls within the allowable transmission rate, the user is notified of the completion of the divided transmission setting (assuming N = 2), but the user has selected any number of speakers 2 that may be divided. Thus, the voice transmission device 1 may be configured to determine the number of divisions (the number of N when dividing into N groups) so as to be within the allowable transmission rate.
　また、上記の実施形態では、補間データの生成方法として、前後のデータを足して１／２倍する方法を例示したが、他のフィルタリング方法を採用しても良い。また、各チャンネルの分割方法も、必ずしもスピーカーの接続順序にしたがってグループ分けされる必要はなく、例えば奇数スピーカーと偶数スピーカーとでグループ分けする方法なども考えられる。 In the above embodiment, the interpolation data generation method is exemplified by a method of adding 1/2 of the preceding and succeeding data, but other filtering methods may be employed. Further, the dividing method of each channel does not necessarily need to be grouped according to the connection order of the speakers. For example, a method of grouping with odd speakers and even speakers may be considered.
　［第２実施形態］
　次に、図１４～図２０を参照し、本発明の第２実施形態について説明する。本実施形態は、ＳＲＣ４３（Sampling　Rate　Converter）（図１６等参照）を用いて、サンプリングレートが異なる複数種類の音声データの同期再生を実現することを特徴とする。以下、第１実施形態と異なる点のみ説明する。なお、本実施形態において、第１実施形態と同様の構成部分については同様の符号を付し、詳細な説明を省略する。また、第１実施形態と同様の構成、部分について適用される変形例は、本実施形態についても同様に適用される。 [Second Embodiment]
Next, a second embodiment of the present invention will be described with reference to FIGS. The present embodiment is characterized in that the SRC 43 (Sampling Rate Converter) (see FIG. 16 and the like) is used to realize synchronized reproduction of a plurality of types of audio data having different sampling rates. Only differences from the first embodiment will be described below. In the present embodiment, the same components as those in the first embodiment are denoted by the same reference numerals, and detailed description thereof is omitted. Moreover, the modification applied about the structure and part similar to 1st Embodiment are applied similarly about this embodiment.
　図１４は、第２実施形態に係るスピーカーシステムＳＹの機能ブロック図である。第２実施形態では、ＳＲＣ４３を用いた同期再生（同期再生（１））に関する機能について説明する。音声伝送装置１は、主な機能構成として、音声入力部１０３、周波数変換部１１１、装置側メモリ１１２および伝送制御部１１３を備えている。 FIG. 14 is a functional block diagram of the speaker system SY according to the second embodiment. In the second embodiment, functions related to synchronous reproduction (synchronous reproduction (1)) using the SRC 43 will be described. The audio transmission device 1 includes an audio input unit 103, a frequency conversion unit 111, a device-side memory 112, and a transmission control unit 113 as main functional configurations.
　音声入力部１０３は、各音声入力源４から音声データを入力する。周波数変換部１１１は、各音声入力源４の信号入力部にそれぞれ対応して設けられたＳＲＣ４３（図１６参照）により、各音声入力源４の再生クロックを一定の周波数クロックに変換する。なお、アナログ信号の信号入力部については、周波数変換部１１１は設けなくても良い。装置側メモリ１１２は、各周波数変換部１１１による周波数変換後の各音声データをバッファリングするものであり、図１６のメモリ４６に相当する。 The voice input unit 103 inputs voice data from each voice input source 4. The frequency conversion unit 111 converts the reproduction clock of each audio input source 4 into a constant frequency clock by the SRC 43 (see FIG. 16) provided corresponding to the signal input unit of each audio input source 4. Note that the frequency converter 111 may not be provided for the analog signal input unit. The device-side memory 112 buffers each audio data after frequency conversion by each frequency conversion unit 111, and corresponds to the memory 46 in FIG.
　伝送制御部１１３は、装置側メモリ１１２から読み出した各音声データを、１フレームにパッキングし、一定の周波数クロックに対応する単一クロックで伝送する。「一定の周波数クロックに対応する」とは、単一クロックが、一定の周波数クロックそのものであること、または一定の周波数クロック（ＳＲＣ４３の読み出し側のクロック）を逓倍したクロックであることを意味する。これにより、サンプリングレートが同じで各音声入力源４の再生クロックが微妙に異なる場合やサンプリングレートが異なる場合に発生するデータ転送のエラーを防止できる。なお、伝送制御部１１３は、図１６のデイジーフォーマットGen４５およびＣＰＵ４８を主要部とする。 The transmission control unit 113 packs each piece of audio data read from the device-side memory 112 into one frame and transmits it with a single clock corresponding to a certain frequency clock. “Corresponding to a fixed frequency clock” means that the single clock is the fixed frequency clock itself or a clock obtained by multiplying the fixed frequency clock (the clock on the reading side of the SRC 43). As a result, it is possible to prevent an error in data transfer that occurs when the sampling rate is the same and the reproduction clock of each audio input source 4 is slightly different or when the sampling rate is different. The transmission control unit 113 includes the daisy format Gen45 and the CPU 48 of FIG. 16 as main parts.
　一方、スピーカー２は、主な機能構成として、データ選択取得部２０２、デイジーチェーン出力部２０３、スピーカー側メモリ２０８、位相同期部２１１、カウンタ部２１２、音声出力部２０７およびスタンバイ制御部２１３を備えている。データ選択取得部２０２およびデイジーチェーン出力部２０３については、第１実施形態と同様であるため説明を省略する。 On the other hand, the speaker 2 includes a data selection acquisition unit 202, a daisy chain output unit 203, a speaker side memory 208, a phase synchronization unit 211, a counter unit 212, an audio output unit 207, and a standby control unit 213 as main functional configurations. Yes. Since the data selection acquisition unit 202 and the daisy chain output unit 203 are the same as those in the first embodiment, description thereof is omitted.
　スピーカー側メモリ２０８は、データ選択取得部２０２により選択取得された音声データをバッファリングするものであり、図１８のメモリ５３に相当する。位相同期部２１１は、データ選択取得部２０２により分離されたSync_dataに同期させて内部クロックを生成するものであり、図１８のＰＬＬ５２（Phase-locked　loop）に相当する。カウンタ部２１２は、スピーカー側メモリ２０８からの音声データの出力タイミングを計測するものであり、図１８のカウンタ５４に相当する。 The speaker side memory 208 buffers the audio data selected and acquired by the data selection acquisition unit 202, and corresponds to the memory 53 of FIG. The phase synchronization unit 211 generates an internal clock in synchronization with Sync_data separated by the data selection / acquisition unit 202, and corresponds to the PLL 52 (Phase-locked loop) in FIG. The counter unit 212 measures the output timing of audio data from the speaker-side memory 208, and corresponds to the counter 54 in FIG.
　音声出力部２０７は、スピーカー側メモリ２０８から読み出された音声データに基づいて音声出力するものであり、図１８の音声出力機構５７に相当する。スタンバイ制御部２１３は、イネーブルフラグ（第１実施形態のチャンネル設定フラグに相当）の「無効」の検出が一定時間続いた場合、自スピーカー２をスタンバイ状態にする。ここで、イネーブルフラグとは、各チャンネルの音声データに対応して設定されるフラグであり、該当データの有効／無効を示す。したがって、イネーブルフラグ「無効」の検出が一定時間続いた場合、そのスピーカー２は使用されないものと判断できるため、スタンバイ状態にする（アンプ５６等を休止させる）ことで、消費電力を抑えることができる。但し、スピーカー２がスタンバイ状態時でも、デイジーチェーンの伝送だけは行い得るようになっている。 The audio output unit 207 outputs audio based on audio data read from the speaker-side memory 208, and corresponds to the audio output mechanism 57 in FIG. The standby control unit 213 sets the speaker 2 to the standby state when detection of “invalid” of the enable flag (corresponding to the channel setting flag of the first embodiment) continues for a certain time. Here, the enable flag is a flag set corresponding to the audio data of each channel, and indicates the validity / invalidity of the corresponding data. Therefore, when the detection of the enable flag “invalid” continues for a certain period of time, it can be determined that the speaker 2 is not used. Therefore, the power consumption can be suppressed by setting the standby state (pausing the amplifier 56 and the like). . However, even when the speaker 2 is in a standby state, only daisy chain transmission can be performed.
　次に、図１５を参照し、第２実施形態に係るSync_dataパックおよびNomal_dataパックについて説明する。同図（ａ）に示すように、本実施形態のSync_dataは、ヘッダ（０から７ビット目）を“１０１１１１１１”とし、当該ヘッダの検出によりSync_dataであるか否かの判別を行う。また、９ビット目には、リセットを定義している。リセットが「１：ＯＮ」の場合、各スピーカー２はリセット処理を行う。 Next, the Sync_data pack and the Normal_data pack according to the second embodiment will be described with reference to FIG. As shown in FIG. 9A, the Sync_data of this embodiment sets the header (0th to 7th bits) to “10111111”, and determines whether it is Sync_data by detecting the header. The ninth bit defines reset. When the reset is “1: ON”, each speaker 2 performs a reset process.
　同図（ｂ）は、Nomal_dataパックのアサイン例を示している。同図の例では、０ビット目にイネーブル（イネーブルフラグ）を定義している。イネーブルは、該当データが有効か否かを示し、各スピーカー２は、当該イネーブル「０：無効」の検出が一定時間続いた場合スタンバイ処理を行う。また、デイジーチェーン上にある全てのスピーカー２に向けた音声データを伝送しても、次のシンクタイミングまで時間が余る場合、残りのNomal_dataパックのイネーブルフラグを「０：無効」に設定する。なお、同図のNomal_dataパック例では、最大３２ビットの例を示したが、チャンネル当たりのビット数に制限はない。 (B) in the figure shows an assignment example of the Normal_data pack. In the example of the figure, an enable (enable flag) is defined at the 0th bit. “Enable” indicates whether or not the corresponding data is valid, and each speaker 2 performs standby processing when the detection of the enable “0: invalid” continues for a predetermined time. Further, even if audio data is transmitted to all the speakers 2 on the daisy chain, if there is time remaining until the next sync timing, the remaining Normal_data pack enable flags are set to “0: invalid”. In the Normal_data pack example in the figure, an example of a maximum of 32 bits is shown, but the number of bits per channel is not limited.
　次に、図１６ないし図１８を参照し、音声伝送装置１およびスピーカー２の制御構成について説明する。図１６は、音声伝送装置１の制御ブロック図である。音声伝送装置１は、ＣＤプレーヤー、ブルーレイプレーヤー、ネットワークプレーヤー等の各音声入力源４に対応したデコーダー４１ａ～４１ｃ、各音声入力源４に対応したＰＬＬ４２ａ～４２ｃ、各音声入力源４に対応したＳＲＣ４３ａ～４３ｃ、システムクロックを生成するクロック生成部４４、ＳＲＣ４３ａ～４３ｃによる周波数変換後の各音声データを、音出しのタイミングを合わせて伝送するデイジーフォーマットGen４５、各音声データを格納するメモリ４６、各種制御プログラムや制御データを記憶する不揮発性メモリ４７、当該不揮発性メモリ４７に記憶されている制御プログラムにしたがって各種演算処理を行うＣＰＵ４８（Central　Processing　Unit）を備えている。 Next, the control configuration of the audio transmission device 1 and the speaker 2 will be described with reference to FIGS. FIG. 16 is a control block diagram of the audio transmission device 1. The audio transmission apparatus 1 includes a decoder 41a to 41c corresponding to each audio input source 4 such as a CD player, a Blu-ray player, a network player, a PLL 42a to 42c corresponding to each audio input source 4, and an SRC 43a corresponding to each audio input source 4. 43c, a clock generation unit 44 for generating a system clock, a daisy format Gen45 for transmitting each sound data after frequency conversion by the SRCs 43a to 43c in synchronism with the sound output timing, a memory 46 for storing each sound data, various controls A nonvolatile memory 47 for storing programs and control data, and a CPU 48 (Central Processing Unit) for performing various arithmetic processes according to the control program stored in the nonvolatile memory 47 are provided.
　デイジーフォーマットGen４５は、図９に示したようなデータプロトコルへの変換を行う。具体的には、ＳＲＣ４３ａ～４３ｃから出力された各音声データを転送する順序に並び替えたり、これをデイジーチェーン伝送用の周波数クロック（ＳＲＣ４３の読み出し側のクロックに同期した伝送用クロック）に載せ変えたり、Sync_dataを埋め込んだり、などの処理を行う。当該構成により、音声伝送装置１は、サンプリングレートが異なる音声データを、１本のデイジーチェーンに乗せて送り出すことが可能となっている。 Daisy format Gen 45 performs conversion to a data protocol as shown in FIG. Specifically, the audio data output from the SRCs 43a to 43c is rearranged in the order of transfer, or is replaced with a frequency clock for daisy chain transmission (a transmission clock synchronized with the clock on the reading side of the SRC 43). Or embed Sync_data. With this configuration, the audio transmission device 1 can send out audio data having different sampling rates on a single daisy chain.
　図１７は、図１６の変形例を示す制御ブロック図である。図１７は、マイク信号を入力する場合の例を示している。このようにアナログ信号の入力がある場合、Ａ／Ｄコンバーター４９でデジタル化するが、その際Ａ／Ｄコンバーター４９で使用するクロックは、ＳＲＣ４３ａ，４３ｂの読み出し側の周波数クロック若しくは、それを逓倍または分周した周波数クロックである。当該構成により、アナログ信号入力部に対応するＳＲＣ４３を省略できる。 FIG. 17 is a control block diagram showing a modification of FIG. FIG. 17 shows an example of inputting a microphone signal. When an analog signal is input in this way, it is digitized by the A / D converter 49. At this time, the clock used by the A / D converter 49 is the frequency clock on the reading side of the SRCs 43a and 43b, or is multiplied or This is a divided frequency clock. With this configuration, the SRC 43 corresponding to the analog signal input unit can be omitted.
　図１８は、スピーカー２の制御ブロック図である。同図に示すように、スピーカー２は、フレームからSync_dataやNomal_dataを分離するデイジーデータ分離部５１、位相の同期を行うＰＬＬ５２、音声データを一時的に蓄えるメモリ５３、音出しタイミングを測るカウンタ５４、ＤＡコンバーター５５、アンプ５６、音声出力機構５７、各種制御プログラムや各種制御データを記憶する不揮発性メモリ５８、スピーカー２全体を制御するコントロール部５９、デイジー出力のアクティブ設定／インアクティブ設定を行うデイジーアクティブ制御部６０を備えている。 FIG. 18 is a control block diagram of the speaker 2. As shown in the figure, the speaker 2 includes a daisy data separation unit 51 that separates Sync_data and Normal_data from a frame, a PLL 52 that performs phase synchronization, a memory 53 that temporarily stores audio data, a counter 54 that measures sound output timing, DA converter 55, amplifier 56, audio output mechanism 57, non-volatile memory 58 for storing various control programs and various control data, control unit 59 for controlling the entire speaker 2, daisy active for daisy output active setting / inactive setting A control unit 60 is provided.
　このように、スピーカー２は、Daisy　Inから入力されたデータのSync_dataを分離し、これに同期する形でＰＬＬ５２を組むことで内部のシステムクロックを生成し動作する。つまり、Sync_dataを検出したタイミングでパルスを発生させ、当該パルスを原発としてＰＬＬ５２の位相比較器に入力する。これにより、ＤＡコンバーター５５の動作用クロックをＰＬＬ５２で生成することができる。なお、Sync_dataはサンプリング周波数の周期で伝送されるため、ＰＬＬ５２では、サンプリング周波数を逓倍した周波数クロックを生成することとなる。また、スピーカー２は、入力されたデータを一旦メモリ５３に蓄えることで音出しのタイミングを合わせている（１サンプリング周波数内のデイジーチェーン伝送によるずれに対するスピーカー２間の音出しを合わせている）。 Thus, the speaker 2 operates by generating an internal system clock by separating the Sync_data of the data input from Daisy In and assembling the PLL 52 in synchronization with the data. That is, a pulse is generated at the timing at which Sync_data is detected, and the pulse is input to the phase comparator of the PLL 52 as a primary source. Thereby, the operation clock of the DA converter 55 can be generated by the PLL 52. Since Sync_data is transmitted at a sampling frequency cycle, the PLL 52 generates a frequency clock obtained by multiplying the sampling frequency. In addition, the speaker 2 synchronizes the timing of sound output by temporarily storing the input data in the memory 53 (the sound output between the speakers 2 is adjusted with respect to the shift due to daisy chain transmission within one sampling frequency).
　次に、図１９および図２０を参照し、サンプリングレート違いの入力データとＳＲＣ出力のデータ、並びにデイジーチェーン上の転送イメージについて説明する。図１９は、サンプリングレート違いの入力データとＳＲＣ出力のデータの関係を示す図である。同図では、簡素化のため各音声入力源４から１チャンネルのみ入力された場合を示している（実際は、音声入力源４ごとに複数チャンネルの入力が可能である）。同図において上段は、入力側の信号入力を示し、下段は、ＳＲＣ４３の出力信号を示している。このように、ＳＲＣ４３を通すことにより、各音声入力源４の再生クロックが一定の周波数クロックに変換される。 Next, input data with different sampling rates, SRC output data, and transfer images on the daisy chain will be described with reference to FIG. 19 and FIG. FIG. 19 is a diagram illustrating the relationship between input data with different sampling rates and SRC output data. In the figure, for simplification, only one channel is input from each audio input source 4 (actually, a plurality of channels can be input for each audio input source 4). In the figure, the upper part shows the signal input on the input side, and the lower part shows the output signal of the SRC 43. In this way, by passing through the SRC 43, the reproduction clock of each audio input source 4 is converted into a constant frequency clock.
　図２０は、ＳＲＣ出力のデータとデイジーチェーン上の転送イメージの関係を示す図である。同図下段において、“Ｓ”はSync_dataを示し、“ｄ＊”で表される数字は、周期（フレームの番号）を示す。このように、Sync_dataに続いて、各チャンネルのNomal_data（“ｄ１”、“ｄ２”、“ｄ３”などで示すデータ）が時系列にしたがい、単一クロックで順次伝送される。 FIG. 20 is a diagram showing the relationship between the SRC output data and the transfer image on the daisy chain. In the lower part of the figure, “S” indicates Sync_data, and the number represented by “d *” indicates the period (frame number). In this way, following Sync_data, Normal_data (data indicated by “d1”, “d2”, “d3”, etc.) of each channel is sequentially transmitted in a single clock according to time series.
　以上説明したとおり、第２実施形態によれば、音声入力源４ごとにＳＲＣ４３を設け、各音声入力源４の再生クロックを一定の周波数クロックに変換するため、サンプリングレートが異なる複数種類の音声データを、１つのデイジーチェーンで同時に再生することができる。また、周波数変換後の各音声データをメモリ４６にバッファリングするため、音声データの出力タイミングを合わせる（同期をとる）ことができる。また、複数の音声データを１フレームにパッキングして伝送するため、伝送制御を容易に行うことができる。つまり、パッキングしない場合と比較して、Sync_data（該当チャンネルのデータを見分けるための情報）が１つだけで済むためデータ量を少なくすることができ、１フレームのデータを３２bitでアラインさせておくことで、Sync_dataの検出および各チャンネルのデータの分離を容易に行うことができる。なお、３２bitの例を説明したが本願の趣旨から逸脱しない限り、他のbit数でも適用できる。 As described above, according to the second embodiment, the SRC 43 is provided for each audio input source 4 and the reproduction clock of each audio input source 4 is converted into a constant frequency clock. Can be played simultaneously in one daisy chain. Also, since the audio data after frequency conversion is buffered in the memory 46, the output timing of the audio data can be synchronized (synchronized). Further, since a plurality of audio data is packed and transmitted in one frame, transmission control can be easily performed. In other words, compared to the case where packing is not performed, only one Sync_data (information for identifying data of the corresponding channel) is required, so the amount of data can be reduced, and one frame of data is aligned with 32 bits. Therefore, Sync_data detection and data separation of each channel can be easily performed. In addition, although the example of 32 bits was demonstrated, as long as it does not deviate from the meaning of this application, other numbers of bits are applicable.
　また、スピーカー２側では、ＰＬＬ５２を備え、分離されたSync_dataに同期させて内部クロックを生成するため、他のスピーカー２との同期を図ることができる。また、メモリ５３に一旦音声データを格納し、カウンタ５４にて出力タイミングをカウントするため、１サンプリング周波数内のデイジーチェーン伝送によるスピーカー間の音出しのずれを解消することができる。 In addition, the speaker 2 is provided with a PLL 52 and generates an internal clock in synchronization with the separated Sync_data, so that synchronization with other speakers 2 can be achieved. Further, since the audio data is temporarily stored in the memory 53 and the output timing is counted by the counter 54, it is possible to eliminate the deviation of the sound output between the speakers due to the daisy chain transmission within one sampling frequency.
　また、スピーカー２は、イネーブルフラグ「無効」の検出が一定時間続いた場合、アンプ５６等を休止させてスタンバイ状態にするため、消費電力を抑えることができる。また、スピーカー２側にアンプ５６を備えることで、チャンネル単位で独立して制御できるため、音声伝送装置１側にアンプを備える構成と比較して、消費電力を抑えることができる。 Further, when the enable flag “invalid” continues to be detected for a certain period of time, the speaker 2 pauses the amplifier 56 and the like to enter a standby state, thereby reducing power consumption. In addition, since the amplifier 56 is provided on the speaker 2 side, control can be performed independently for each channel, so that power consumption can be suppressed as compared with a configuration in which the amplifier is provided on the audio transmission device 1 side.
　［第３実施形態］
　次に、図２１～図３０を参照し、本発明の第３実施形態について説明する。本実施形態は、ＦｉＦｏ６１（First-In　First-Out）（図２７参照）を用いて、サンプリングレートが異なる複数種類の音声データの同期再生を実現することを特徴とする。以下、第１，第２実施形態と異なる点のみ説明する。なお、本実施形態において、第１，第２実施形態と同様の構成部分については同様の符号を付し、詳細な説明を省略する。また、第１，第２実施形態と同様の構成、部分について適用される変形例は、本実施形態についても同様に適用される。 [Third Embodiment]
Next, a third embodiment of the present invention will be described with reference to FIGS. The present embodiment is characterized in that synchronized playback of a plurality of types of audio data having different sampling rates is realized using FiFo61 (First-In First-Out) (see FIG. 27). Only differences from the first and second embodiments will be described below. In the present embodiment, the same components as those in the first and second embodiments are denoted by the same reference numerals, and detailed description thereof is omitted. Also, modifications applied to the same configurations and parts as those in the first and second embodiments are similarly applied to this embodiment.
　図２１は、第３実施形態に係るスピーカーシステムＳＹの機能ブロック図である。第３実施形態では、ＦｉＦｏ６１を用いた同期再生（同期再生（２））に関する機能について説明する。音声伝送装置１は、主な機能構成として、オプションＩＤアサイン設定部１２１、音声入力部１０３、装置側バッファ１２２、バッファ監視部１２３、伝送制御部１２４およびリシンク設定指令部１２５を備えている。 FIG. 21 is a functional block diagram of the speaker system SY according to the third embodiment. In the third embodiment, functions related to synchronized playback (synchronized playback (2)) using FiFo 61 will be described. The audio transmission device 1 includes an option ID assignment setting unit 121, an audio input unit 103, a device side buffer 122, a buffer monitoring unit 123, a transmission control unit 124, and a resync setting command unit 125 as main functional configurations.
　オプションＩＤアサイン設定部１２１は、オプションＩＤのアサイン処理を行う。ここでは、デイジーチェーン伝送されるフレーム内において、Option_dataがSync_dataから何番目のパックに存在するのかを認識させるため（図２２参照）、各スピーカー２のスピーカーＩＤとオプションＩＤとの紐付けを行う。詳細については、図２４にて後述する。 The option ID assignment setting unit 121 performs an option ID assignment process. Here, in order to recognize in which frame the Option_data exists from the Sync_data in the daisy chain transmitted frame (see FIG. 22), the speaker ID of each speaker 2 is associated with the option ID. Details will be described later with reference to FIG.
　音声入力部１０３は、各音声入力源４から音声データを入力する。装置側バッファ１２２は、各音声入力源４から入力された音声データを、それぞれバッファリングするものであり、図２７のＦｉＦｏ６１に相当する。バッファ監視部１２３は、各装置側バッファ１２２のデータ量を監視するものであり、図２７のＦｉＦｏデータ監視部６２に相当する。 The voice input unit 103 inputs voice data from each voice input source 4. The apparatus-side buffer 122 buffers the audio data input from each audio input source 4, and corresponds to the FiFo 61 in FIG. The buffer monitoring unit 123 monitors the data amount of each device-side buffer 122 and corresponds to the FiFo data monitoring unit 62 in FIG.
　伝送制御部１２４は、各バッファ監視部１２３の監視結果に基づいて、各装置側バッファ１２２から読み出す各音声データのデータ数を決定し、それぞれ決定したデータ数の各音声データを１フレームにパッキングして、単一クロックで伝送する。なお、伝送制御部１２４は、図２７のデイジーフォーマットGen４５およびＣＰＵ４８を主要部とする。 The transmission control unit 124 determines the number of pieces of audio data to be read from each device-side buffer 122 based on the monitoring result of each buffer monitoring unit 123, and packs each piece of audio data having the determined number of data into one frame. And transmit with a single clock. The transmission control unit 124 has the daisy format Gen 45 and the CPU 48 of FIG. 27 as main parts.
　ところで、図２２に示すように、本実施形態のフレームは、各スピーカー２に対応した複数のNomal_data（チャンネルデータ領域）から成るチャンネルデータ領域群（“Ｃｈ１・・・Ｃｈ　Ｌａｓｔ”）と、各スピーカー２に対応した複数のOption_data（オプションデータ領域）から成るオプションデータ領域群（“Ｏｐ１・・・Ｏｐ　Ｌａｓｔ”）とを含む構成となっている。このため、伝送制御部１２４は、各装置側バッファ１２２のデータ量が上限閾値を超えた場合、チャンネルデータ領域およびオプションデータ領域に合計２サンプリング分（２データ分）の音声データを組み込み、各装置側バッファ１２２のデータ量が上限閾値以下且つ下限閾値以上の場合、チャンネルデータ領域のみに１サンプリング分（１データ分）の音声データを組み込む。また、伝送制御部１２４は、各装置側バッファ１２２のデータ量が下限閾値を下回った場合、各データ領域へのデータ組み込みを行わない。 By the way, as shown in FIG. 22, the frame of this embodiment includes a channel data area group (“Ch1... Ch Last”) composed of a plurality of Normal_data (channel data areas) corresponding to each speaker 2 and each speaker. 2 includes an option data area group (“Op1... Op Last”) composed of a plurality of Option_data (option data areas) corresponding to 2. For this reason, when the data amount of each device-side buffer 122 exceeds the upper limit threshold, the transmission control unit 124 embeds a total of two samplings (two data) of audio data in the channel data area and the option data area, When the data amount of the side buffer 122 is equal to or lower than the upper threshold and equal to or higher than the lower threshold, audio data for one sampling (one data) is incorporated only in the channel data area. In addition, when the data amount of each device-side buffer 122 falls below the lower limit threshold, the transmission control unit 124 does not incorporate data into each data area.
　また、伝送制御部１２４は、各スピーカー２に対応する各チャンネルデータ領域に、各スピーカー２に対応する各オプションデータ領域内のデータの有無をオプションフラグで表す。同様に、各スピーカー２に対応する各チャンネルデータ領域に、該当データの有効／無効をイネーブルフラグで表す。これらのフラグによって、スピーカー２に対し、伝送したデータの所在を通知する。また、伝送制御部１２４は、所定の条件を満たした場合、フレームのSync_dataパック内に、リシンクフラグで表す。ここで、「所定の条件を満たした場合」とは、音声入力切替、入力される音声データのサンプリングレートの変更、音声伝送装置の電源ＯＮ、音声データの出力開始などを指す。 In addition, the transmission control unit 124 indicates, in an option flag, whether each channel data area corresponding to each speaker 2 has data in each option data area corresponding to each speaker 2. Similarly, the validity / invalidity of the corresponding data is represented by an enable flag in each channel data area corresponding to each speaker 2. With these flags, the speaker 2 is notified of the location of the transmitted data. In addition, when a predetermined condition is satisfied, the transmission control unit 124 represents a resync flag in the Sync_data pack of the frame. Here, “when a predetermined condition is satisfied” refers to switching of audio input, changing the sampling rate of input audio data, turning on the power of the audio transmission device, starting output of audio data, and the like.
　リシンク設定指令部１２５は、複数のスピーカー２をゾーンごと（例えば、部屋ごと）に分類し、同期処理の対象となるゾーンの各スピーカー２に対し、リシンク設定を指令する。スピーカー２側では、リシンク設定指令部１２５の指令およびリシンクフラグの検出に基づいて同期処理を行う。詳細については、図２５，図２６にて後述する。 The resync setting command unit 125 classifies the plurality of speakers 2 for each zone (for example, for each room) and commands the resync setting to each speaker 2 in the zone to be subjected to the synchronization process. On the speaker 2 side, synchronization processing is performed based on the command of the resync setting command unit 125 and the detection of the resync flag. Details will be described later with reference to FIGS.
　一方、スピーカー２は、主な制御構成として、データ選択取得部２０２、デイジーチェーン出力部２０３、スピーカー側バッファ２２１、バッファ制御部２２２、音声出力部２０７、スタンバイ制御部２１３、リシンク設定部２２３および同期処理部２２４を備えている。 On the other hand, the speaker 2 has, as main control configurations, a data selection acquisition unit 202, a daisy chain output unit 203, a speaker side buffer 221, a buffer control unit 222, an audio output unit 207, a standby control unit 213, a resync setting unit 223, and a synchronization. A processing unit 224 is provided.
　スピーカー側バッファ２２１は、音声伝送装置１から伝送された音声データをバッファリングするものであり、図２８のＦｉＦｏ６４に相当する。バッファ制御部２２２は、スピーカー側バッファ２２１に音声データを書き込むと共に、書き込まれたデータ量を監視し、当該監視結果に基づいて読み出しクロックの可変制御を行うものであり、図２８のコントロール部５９に相当する。また、バッファ制御部２２２は、自スピーカー２に対応したチャンネルデータ領域のイネーブルフラグが「有効」を示す場合、チャンネルデータ領域内のデータをスピーカー側バッファ２２１に書き込み、チャンネルデータ領域のオプションフラグが「有」を示す場合、自スピーカー２に対応したオプションデータ領域内のデータもスピーカー側バッファ２２１に書き込む。 The speaker-side buffer 221 buffers audio data transmitted from the audio transmission device 1, and corresponds to FiFo64 in FIG. The buffer control unit 222 writes audio data to the speaker side buffer 221 and monitors the amount of written data, and performs variable control of the read clock based on the monitoring result. The control unit 59 in FIG. Equivalent to. Further, when the enable flag of the channel data area corresponding to the speaker 2 indicates “valid”, the buffer control unit 222 writes the data in the channel data area to the speaker side buffer 221, and the option flag of the channel data area is “ When “present” is indicated, the data in the option data area corresponding to the speaker 2 is also written in the speaker-side buffer 221.
　音声出力部２０７は、スピーカー側バッファ２２１から読み出された音声データに基づいて音声出力を行うものであり、図２８の音声出力機構５７に相当する。スタンバイ制御部２１３は、第２実施形態と同様に、イネーブルフラグ「無効」の検出が一定時間続いた場合、自スピーカー２をスタンバイ状態にする。 The audio output unit 207 outputs audio based on audio data read from the speaker-side buffer 221 and corresponds to the audio output mechanism 57 in FIG. As in the second embodiment, the standby control unit 213 sets the speaker 2 to the standby state when the detection of the enable flag “invalid” continues for a certain period of time.
　リシンク設定部２２３は、リシンク設定指令部１２５の指令にしたがって、自スピーカー２のリシンク設定を行う。具体的には、リシンクレジスタにリシンク設定を行う。同期処理部２２４は、リシンク設定部２２３によりリシンク設定されており、且つSync_dataパックからリシンクフラグ「ＯＮ」を検出したとき、同期処理を行う。 The resync setting unit 223 performs the resync setting of the speaker 2 according to the command of the resync setting command unit 125. Specifically, resync is set in the resync register. The synchronization processing unit 224 performs synchronization processing when the resync is set by the resync setting unit 223 and the resync flag “ON” is detected from the Sync_data pack.
　次に、図２３を参照し、第３実施形態に係るSync_dataパックおよびNomal_dataパックについて説明する。同図（ａ）に示すように、本実施形態のSync_dataは、０から７ビット目にヘッダを定義し、９ビット目に、リセットを定義している。これらは第２実施形態と同様であるが、８ビット目にリシンクを定義している点で異なる。リシンクは、上記の通り同期を取り直す必要がある場合、スピーカー２に対し再同期処理を実行させるためのものである。音声入力に変化がない場合などは再同期処理が必要ないため、通常のSync_dataのリシンクフラグは「０：ＯＦＦ」に設定されている。 Next, a Sync_data pack and a Normal_data pack according to the third embodiment will be described with reference to FIG. As shown in FIG. 5A, Sync_data of this embodiment defines a header at 0 to 7 bits and a reset at 9 bits. These are the same as in the second embodiment, but differ in that resync is defined at the eighth bit. The resynchronization is for causing the speaker 2 to execute resynchronization processing when it is necessary to regain synchronization as described above. Since there is no need for resynchronization processing when there is no change in the audio input, the normal Sync_data resync flag is set to “0: OFF”.
　同図（ｂ）は、Nomal_dataパックのアサイン例を示している。本実施形態のNomal_dataは、０ビット目にイネーブルを定義している点は、第２実施形態と同様であるが、１ビット目にオプション（オプションフラグ）を定義している点で異なる。オプションは、オプションデータ領域に音声データが組み込まれている場合「１：有」と設定され、オプションデータ領域に音声データが組み込まれていない場合「１：無」と設定される。 (B) in the figure shows an assignment example of the Normal_data pack. The Normal_data of this embodiment is similar to the second embodiment in that enable is defined in the 0th bit, but differs in that an option (option flag) is defined in the 1st bit. The option is set to “1: Yes” if audio data is incorporated in the option data area, and “1: None” if audio data is not incorporated in the option data area.
　次に、図２４を参照し、オプションＩＤアサイン設定処理について説明する。図２２に示したとおり、デイジーチェーン上では、Sync_data、Nomal_data、Option_dataが１フレームとなって伝送される。Nomal_dataは、デイジーチェーン内に存在するスピーカー数と同じ数が１フレームに存在する。Option_dataは、Nomal_dataのパックが続いた後に存在するため、スピーカー２の数が分かった段階でOption_dataがSync_dataから何番目のパックに存在するのかを設定する必要がある。同図に示すオプションＩＤアサイン設定処理は、Option_dataの位置を示すためのオプションＩＤを設定する処理である。 Next, the option ID assignment setting process will be described with reference to FIG. As shown in FIG. 22, on the daisy chain, Sync_data, Normal_data, and Option_data are transmitted as one frame. As for Nomal_data, the same number as the number of speakers existing in the daisy chain exists in one frame. Since Option_data exists after the pack of Normal_data continues, it is necessary to set in which pack the Option_data exists from Sync_data when the number of speakers 2 is known. The option ID assignment setting process shown in the figure is a process for setting an option ID for indicating the position of Option_data.
　音声伝送装置１は、スピーカーＩＤが設定され、且つオプションＩＤが未設定のスピーカー２が存在するか否かを判別する（Ｓ９１）。存在しない場合は（Ｓ９１：Ｎｏ）、オプションＩＤアサイン設定処理の必要がないため、エラー処理を行う（Ｓ９２）。また、存在する場合は（Ｓ９１：Ｙｅｓ）、設定するスピーカーＩＤ（最初は、「１」）をセットし（Ｓ９３）、順次アサインループ処理を開始する（Ｓ９４）。 The audio transmission device 1 determines whether or not there is a speaker 2 for which the speaker ID is set and the option ID is not set (S91). If it does not exist (S91: No), there is no need for the option ID assignment setting process, so an error process is performed (S92). If it exists (S91: Yes), the speaker ID to be set (initially “1”) is set (S93), and assignment loop processing is started sequentially (S94).
　アサインループ処理では、設定するオプションＩＤをセットし（スピーカー数にスピーカーＩＤを加算したもの，Ｓ９５）、セットしたオプションＩＤを記憶する（Ｓ９６）。以上のアサインループ処理を終了すると（Ｓ９７）、設定するスピーカーＩＤをインクリメントし（Ｓ９８）、スピーカーＩＤが有効か否かを判別する（Ｓ９９）。スピーカーＩＤが有効の場合は（全てのスピーカー２のアサインが完了していない場合は，Ｓ９９：Ｙｅｓ）、Ｓ９４以降を繰り返す。また、スピーカーＩＤが無効である場合は（全てのスピーカー２のアサインが完了した場合は，Ｓ９９：Ｎｏ）、オプションＩＤアサイン設定処理を終了する。 In the assign loop process, the option ID to be set is set (the number of speakers plus the speaker ID, S95), and the set option ID is stored (S96). When the above assignment loop processing is completed (S97), the speaker ID to be set is incremented (S98), and it is determined whether the speaker ID is valid (S99). When the speaker ID is valid (S99: Yes when assignment of all the speakers 2 is not completed), S94 and subsequent steps are repeated. If the speaker ID is invalid (if all the speakers 2 have been assigned, S99: No), the option ID assignment setting process is terminated.
　次に、図２５および図２６を参照し、同期リセット処理について説明する。図２５は、音声伝送装置１側の同期リセット処理を示すフローチャートである。音声伝送装置１は、入力データのサンプリングレートが変更になった場合や入力切替が行われた場合など同期リセット処理が必要と判定した場合（Ｓ１１１：Ｙｅｓ）、該当するゾーン内のスピーカーＩＤを読み出し（Ｓ１１２）、該当ゾーン内におけるスピーカー２のリシンク前処理ループを開始する（Ｓ１１３）。リシンク前処理ループでは、該当スピーカー２をミュート設定し（Ｓ１１４）、各スピーカー２に対しリシンク設定を行う（Ｓ１１５）。つまり、該当スピーカー２に対して、ミュート指令を行うと共に、リシンクレジスタへの設定を指令する。ミュート設定は、リシンク動作によるノイズが発生するのを防止するためである。リシンク前処理ループが終了すると（Ｓ１１６）、該当ゾーン内の次のスピーカーＩＤを読み出し（Ｓ１１７）、該当ゾーンに残りのスピーカー２が存在するか否かを判別する（Ｓ１１８）。存在する場合は（Ｓ１１８：Ｙｅｓ）、Ｓ１１２以降を繰り返す。また、存在しない場合は（Ｓ１１８：Ｎｏ）、リシンクフラグ「１：ＯＮ」のSync_dataを入力する（Ｓ１１９）。 Next, the synchronous reset process will be described with reference to FIGS. FIG. 25 is a flowchart showing a synchronization reset process on the voice transmission device 1 side. If the audio transmission device 1 determines that a synchronous reset process is necessary, such as when the sampling rate of input data is changed or when input switching is performed (S111: Yes), the audio transmission device 1 reads the speaker ID in the corresponding zone. (S112), the resync preprocessing loop of the speaker 2 in the corresponding zone is started (S113). In the resync preprocessing loop, the corresponding speaker 2 is set to mute (S114), and the resync setting is performed for each speaker 2 (S115). That is, the mute command is issued to the corresponding speaker 2 and the setting to the resync register is commanded. The mute setting is for preventing noise due to the resync operation. When the resync preprocessing loop is completed (S116), the next speaker ID in the corresponding zone is read (S117), and it is determined whether or not the remaining speaker 2 exists in the corresponding zone (S118). When it exists (S118: Yes), S112 and subsequent steps are repeated. If not (S118: No), Sync_data of the resync flag “1: ON” is input (S119).
　その後、該当ゾーン内における先頭スピーカー２のスピーカーＩＤを読み出し（Ｓ１２０）、リシンク後処理ループを開始する（Ｓ１２１）。リシンク後処理ループでは、該当スピーカー２のリシンク設定の解除およびミュート設定の解除を行う（Ｓ１２２，Ｓ１２３）。このように、リシンク設定の解除を行うことで、他のゾーンへのリシンク動作による誤動作を防ぐことができる。リシンク後処理ループが終了すると（Ｓ１２４）、該当ゾーン内の次のスピーカーＩＤを読み出し（Ｓ１２５）、該当ゾーンに残りのスピーカー２が存在するか否かを判別する（Ｓ１２６）。存在する場合は（Ｓ１２６：Ｙｅｓ）、Ｓ１２０以降を繰り返す。また、存在しない場合は（Ｓ１２６：Ｎｏ）、同期リセット処理を終了する。 Thereafter, the speaker ID of the first speaker 2 in the corresponding zone is read (S120), and a resync post-processing loop is started (S121). In the resync post-processing loop, the resync setting of the corresponding speaker 2 is canceled and the mute setting is canceled (S122, S123). Thus, by canceling the resync setting, it is possible to prevent malfunction due to the resync operation to another zone. When the post-resync processing loop is completed (S124), the next speaker ID in the corresponding zone is read (S125), and it is determined whether or not the remaining speaker 2 exists in the corresponding zone (S126). When it exists (S126: Yes), S120 and subsequent steps are repeated. If not (S126: No), the synchronization reset process is terminated.
　次に、図２６を参照し、スピーカー２側の同期リセット処理について説明する。スピーカー２は、リシンクレジスタを参照してリシンク設定が行われているか否かを判別し（Ｓ１３１）、リシンク設定が行われている場合は（Ｓ１３１：Ｙｅｓ）、ミュート設定を行う（Ｓ１３２）。また、リシンクフラグ「１：ＯＮ」のSync_dataを入力すると（Ｓ１３３：Ｙｅｓ）、リシンク動作を行う（Ｓ１３４）。その後、音声伝送装置１からミュート設定の解除指令を取得すると（Ｓ１３５：Ｙｅｓ）、ミュート設定を解除し（Ｓ１３６）、同期リセット処理を終了する。以上のように、図２５および図２６のフローにしたがって同期リセット処理を行うことで、各音声データを１フレームとして送信する本実施形態の場合でも（フレーム内にSync_dataが１つしか存在しない場合でも）、ゾーンごとにリシンクをかけることができる。 Next, the synchronization reset processing on the speaker 2 side will be described with reference to FIG. The speaker 2 refers to the resync register to determine whether or not the resync setting is performed (S131). If the resync setting is performed (S131: Yes), the mute setting is performed (S132). When Sync_data of the resync flag “1: ON” is input (S133: Yes), the resync operation is performed (S134). Thereafter, when a mute setting cancel command is acquired from the audio transmission device 1 (S135: Yes), the mute setting is canceled (S136), and the synchronization reset process is terminated. As described above, by performing the synchronization reset processing according to the flow of FIG. 25 and FIG. 26, even in the present embodiment in which each audio data is transmitted as one frame (even when only one Sync_data exists in the frame) ), You can resync for each zone.
　次に、図２７，図２８を参照し、第３実施形態に係る音声伝送装置１およびスピーカー２の制御構成について説明する。図２７は、音声伝送装置１の制御ブロック図である。音声伝送装置１は、ＣＤプレーヤーやブルーレイプレーヤー等の各音声入力源４に対応したデコーダー４１ａ，４１ｂ、各音声入力源４に対応したＰＬＬ４２ａ，４２ｂ、各音声入力源４に対応したＦｉＦｏ６１ａ，６１ｂ、各ＦｉＦｏ６１ａ，６１ｂ内のデータ量を監視するＦｉＦｏデータ監視部６２ａ，６２ｂ、システムクロックを生成するクロック生成部４４、各ＦｉＦｏ６１ａ，６１ｂから読み出した各音声データを伝送するデイジーフォーマットGen４５、各音声データを格納するメモリ４６、リシンク設定を行うリシンクタイミングGen６３、各種制御プログラムや制御データを記憶する不揮発性メモリ４７、当該不揮発性メモリ４７に記憶されている制御プログラムにしたがって各種演算処理を行うＣＰＵ４８を備えている。 Next, the control configuration of the audio transmission device 1 and the speaker 2 according to the third embodiment will be described with reference to FIGS. FIG. 27 is a control block diagram of the audio transmission device 1. The audio transmission device 1 includes decoders 41a and 41b corresponding to each audio input source 4 such as a CD player and a Blu-ray player, PLLs 42a and 42b corresponding to each audio input source 4, FiFo 61a and 61b corresponding to each audio input source 4, FiFo data monitoring units 62a and 62b that monitor the amount of data in each FiFo 61a and 61b, a clock generation unit 44 that generates a system clock, a daisy format Gen45 that transmits each audio data read from each FiFo 61a and 61b, and each audio data A memory 46 for storing, a resync timing Gen 63 for performing resync setting, a non-volatile memory 47 for storing various control programs and control data, and a CPU 48 for performing various arithmetic processes in accordance with the control program stored in the non-volatile memory 47 are provided. Yes.
　当該構成により、音声伝送装置１は、例えばＦｉＦｏデータ監視部６２によりチャンネル１に対応するＦｉＦｏ６１内のデータ量が上限閾値を超えたことを検出した場合、チャンネル１の入力がデイジーチェーン伝送より速い（入力レートが転送レートを上回っている）と判断し、「Ｃh１」のチャンネルデータ領域に１サンプリング分のデータを格納すると共に、「Ｃh１」のオプションデータ領域に次のサンプリングデータを格納する（図２２参照）。これにより通常の２倍のデータまで伝送可能となり、入力部分のＦｉＦｏ６１があふれてしまうことを防止できる。なお、ＦｉＦｏ６１内のデータ量が上限閾値を超えたことを検出した場合、デイジーフォーマットGen４５は、Nomal_dataパックの０ビット目にイネーブル「１：有効」、１ビット目にオプション「１：有」を組み込む。逆に、例えばチャンネル１に対応するＦｉＦｏ６１内のデータ量が下限閾値を下回ったことを検出した場合、デイジーフォーマットGen４５は、チャンネル１の入力がデイジーチェーン伝送より遅いと判断し、「Ｃh１」のチャンネルデータ領域への組み込みを行わない。なお、この場合デイジーフォーマットGen４５は、Nomal_dataパックの０ビット目をイネーブル「０：無効」とする。 With this configuration, for example, when the audio transmission apparatus 1 detects that the amount of data in the FiFo 61 corresponding to the channel 1 exceeds the upper threshold by the FiFo data monitoring unit 62, the input of the channel 1 is faster than the daisy chain transmission ( The input rate exceeds the transfer rate), data for one sampling is stored in the channel data area of “Ch1”, and the next sampling data is stored in the option data area of “Ch1” (FIG. 22). reference). As a result, it is possible to transmit up to twice as much data as normal, and it is possible to prevent the FiFo 61 in the input portion from overflowing. When it is detected that the amount of data in the FiFo 61 has exceeded the upper limit threshold, the daisy format Gen45 incorporates the enable “1: valid” in the 0th bit of the Normal_data pack and the option “1: present” in the 1st bit. . Conversely, for example, when it is detected that the amount of data in FiFo 61 corresponding to channel 1 has fallen below the lower threshold, daisy format Gen 45 determines that the input of channel 1 is slower than daisy chain transmission, and the channel of “Ch1” Do not include in the data area. In this case, the daisy format Gen 45 enables the 0th bit of the Normal_data pack “0: invalid”.
　一方、リシンクタイミングGen６３は、Sync_dataのリシンクフラグを設定するタイミングを作る。具体的には、ＣＰＵ４８にて入力データのサンプリングレートが変更になったり入力切替が行われたことを検出した場合、該当するスピーカー２のリシンク設定を行う。また、全てのスピーカー２のリシンク設定が終わったところで、Sync_dataの埋め込みのタイミングでリシンクタイミングGen６３がタイミング信号を出力し、この信号に合わせてデイジーフォーマットGen４５がリシンクフラグを「１：ＯＮ」にしたSync_dataを生成する。 On the other hand, the resync timing Gen63 creates the timing for setting the sync_data resync flag. Specifically, when the CPU 48 detects that the sampling rate of input data has been changed or input switching has been performed, resync setting of the corresponding speaker 2 is performed. When the resync settings for all the speakers 2 have been completed, the resync timing Gen63 outputs a timing signal at the timing of Sync_data embedding, and the daisy format Gen45 sets the resync flag to “1: ON” in accordance with this signal. Is generated.
　図２８は、スピーカー２の制御ブロック図である。同図に示すように、スピーカー２は、フレームからSync_dataやNomal_dataを分離するデイジーデータ分離部５１、Sync_dataとの同期を行うためのＰＬＬ５２、音声データを一時的に蓄えるメモリ５３、音だしタイミングを測るカウンタ５４、音声データをバッファリングするＦｉＦｏ６４、ＦｉＦｏ６４内のデータ量を監視するＦｉＦｏデータ監視部６５、ＦｉＦｏ６４の読み出しクロックを可変するためのＰＬＬ６６、チャンネルアサインの音（識別信号）を出すための発信源となる識別信号Gen６７、識別信号と通常の音声を切り替えるためのセレクター６８、ＤＡコンバーター５５、同期リセット処理時に音声信号をMuteするMute回路６９、アンプ５６、音声出力機構５７、各種制御プログラムや各種制御データを記憶する不揮発性メモリ５８、スピーカー２全体を制御するコントロール部５９、デイジー出力のアクティブ設定／インアクティブ設定を行うデイジーアクティブ制御部６０を備えている。 FIG. 28 is a control block diagram of the speaker 2. As shown in the figure, the speaker 2 measures a sound output timing, a daisy data separation unit 51 that separates Sync_data and Normal_data from a frame, a PLL 52 for synchronizing with Sync_data, a memory 53 that temporarily stores audio data. Counter 54, FiFo64 for buffering audio data, FiFo data monitoring unit 65 for monitoring the amount of data in FiFo64, PLL 66 for changing the readout clock of FiFo64, and a transmission source for outputting channel assignment sound (identification signal) Identification signal Gen67, selector 68 for switching between identification signal and normal sound, DA converter 55, mute circuit 69 for muting the audio signal during synchronization reset processing, amplifier 56, audio output mechanism 57, various control programs and various controls Non-volatile memory 58 for storing data, Manufacturers 2 controller 59 that controls the whole, and a daisy active control unit 60 for active set / inactive settings daisy output.
　当該構成により、スピーカー２は、デイジーデータ分離部５１によりDaisy　Inから入力されたデータ（フレーム）から「Ch１」のデータを分離し、取り込む。取り込んだ「Ch１」データのイネーブルフラグが「１：有効」に設定されていればＦｉＦｏ６４に書き込む。また、オフションフラグが「１：有」に設定されているときは、「Op１」のデータも書き込む。一方、イネーブルフラグが「０：無効」に設定されているときはＦｉＦｏ６４に書き込みを行わない。また、オフションフラグが「０：無」に設定されているときも、「Op１」のデータを書き込まない。さらに、ＦｉＦｏデータ監視部６５によりＦｉＦｏ６４のデータ量を監視し、上限閾値を超えた場合は、ＰＬＬ６６を制御してＦｉＦｏ６４の読み出し側の周波数クロックを高くする。逆に、下限閾値を下回った場合は、ＰＬＬ６６を制御してＦｉＦｏ６４の読み出し側の周波数クロックを低くする。これにより、ＦｉＦｏ６４を空にすることやあふれさせることなく、間接的に音声伝送装置１の入力に同期させて音声再生を行うことができる。 With this configuration, the speaker 2 separates and captures “Ch1” data from the data (frame) input from Daisy In by the daisy data separation unit 51. If the enable flag of the fetched “Ch1” data is set to “1: Valid”, the data is written into the FiFo64. In addition, when the option flag is set to “1: present”, the data of “Op1” is also written. On the other hand, when the enable flag is set to “0: invalid”, writing to the FiFo 64 is not performed. Further, even when the option flag is set to “0: None”, the data of “Op1” is not written. Further, the data amount of FiFo64 is monitored by the FiFo data monitoring unit 65. When the upper limit threshold is exceeded, the PLL 66 is controlled to increase the frequency clock on the reading side of the FiFo64. On the other hand, when the value falls below the lower limit threshold, the PLL 66 is controlled to lower the frequency clock on the reading side of the FiFo64. As a result, it is possible to perform audio reproduction in synchronization with the input of the audio transmission device 1 indirectly without emptying or overflowing the FiFo64.
　次に、図２９および図３０を参照し、サンプリングレート違いの入力データとＦｉＦｏ出力のデータ、並びにデイジーチェーン上の転送イメージについて説明する。図２９は、サンプリングレート違いの入力データとＦｉＦｏ出力のデータの関係を示す図である。同図では、３種類の信号が入力された場合を示している。また、同図において上段は、入力側の信号入力を示し、下段は、スピーカー２に設けられたＦｉＦｏ６４の読み出し側の出力信号を示している。このように、スピーカー２では、入力された音声入力源４の周波数クロックと同等の周波数クロックで再生が行われる。 Next, with reference to FIG. 29 and FIG. 30, input data with different sampling rates, FiFo output data, and a transfer image on the daisy chain will be described. FIG. 29 is a diagram illustrating a relationship between input data with different sampling rates and data of FiFo output. In the figure, a case where three types of signals are input is shown. Further, in the same figure, the upper part shows the signal input on the input side, and the lower part shows the output signal on the reading side of the FiFo 64 provided in the speaker 2. As described above, the speaker 2 performs reproduction with a frequency clock equivalent to the frequency clock of the input audio input source 4.
　図３０は、サンプリングレート違いの入力データとデイジーチェーン上の転送イメージの関係を示す図である。同図下段において、“Ｓ”はSync_dataを示し、“ｄ＊”で表される数字は、周期（フレームの番号）を示す。また、“ｄｘ”はイネーブルフラグの「０：無効」を示し、“ｏｘ”はオプションフラグの「０：無」を示す。同図の例では、例えばネットワークプレーヤーの場合、最初のサンプリングデータを第１フレームのチャンネルデータ領域、２番目のサンプリングデータを第２フレームのチャンネルデータ領域、３番目のサンプリングデータを第２フレームのオプションデータデータ領域、４番目のサンプリングデータを第３フレームのチャンネルデータ領域・・・にそれぞれ組み込みを行い、データ伝送を行っている。 FIG. 30 is a diagram showing a relationship between input data with different sampling rates and a transfer image on the daisy chain. In the lower part of the figure, “S” indicates Sync_data, and the number represented by “d *” indicates the period (frame number). “Dx” indicates an enable flag “0: Invalid”, and “ox” indicates an option flag “0: None”. In the example of the figure, for example, in the case of a network player, the first sampling data is the channel data area of the first frame, the second sampling data is the channel data area of the second frame, and the third sampling data is the option of the second frame. The data data area, the fourth sampling data are incorporated into the channel data area of the third frame, and data transmission is performed.
　以上説明したとおり、第３実施形態によれば、音声伝送装置１側で、各音声入力源４に対応するＦｉＦｏ６１のデータ量を監視し、その監視結果に基づいて、１フレームに含める各音声データの有り/無しおよび追加（オプション）データの有り/無しを決定するため、サンプリングレートが異なる複数種類の音声データを伝送することができる。また、スピーカー２側では、伝送された音声データを一旦ＦｉＦｏ６４に書き込み、そのデータ量に応じて読み出しクロックの可変制御を行うため、１つのデイジーチェーンスピーカーシステムで同時に複数種類の音声データを再生することができる As described above, according to the third embodiment, the audio transmission apparatus 1 side monitors the data amount of the FiFo 61 corresponding to each audio input source 4, and each audio data included in one frame based on the monitoring result. In order to determine the presence / absence of data and the presence / absence of optional data (optional), a plurality of types of audio data having different sampling rates can be transmitted. On the speaker 2 side, the transmitted audio data is once written in the FiFo 64, and the read clock is variably controlled according to the amount of data, so a plurality of types of audio data are reproduced simultaneously with one daisy chain speaker system. Can
　また、音声伝送装置１側では、ＦｉＦｏ６１のデータ量が、上限閾値を超えた場合、上限閾値以下且つ下限閾値以上の場合、下限閾値を下回った場合の３つのパターンに場合分けして、フレームへの音声データの記録を行い、各パターンをオプションフラグおよびイネーブルフラグで示すため、スピーカー２側で各パターンを正確に判別し、バッファ書き込み処理および再生クロック生成を行うことができる。 Also, on the voice transmission device 1 side, when the data amount of FiFo 61 exceeds the upper threshold, when it is lower than the upper threshold and equal to or higher than the lower threshold, it is divided into three patterns when it falls below the lower threshold, and is transferred to the frame. Since the audio data is recorded and each pattern is indicated by an option flag and an enable flag, each pattern can be accurately discriminated on the speaker 2 side, and buffer writing processing and reproduction clock generation can be performed.
　また、Sync_dataパック内でリシンクフラグの設定が可能であるため、音声入力切替時、入力される音声データのサンプリングレートの変更時、音声伝送装置１の電源ＯＮ時、音声データの出力開始時など、必要なタイミングで再同期処理を行うことができる。また、リシンク設定指令部１２５により、同期処理の対象となるゾーンの各スピーカー２に対しリシンク設定を指令できるため、再同期処理が必要なゾーン内のスピーカー２だけを対象に効率的な再同期処理を行うことができる。 In addition, since the resync flag can be set in the Sync_data pack, when the audio input is switched, when the sampling rate of the input audio data is changed, when the audio transmission device 1 is turned on, when the output of the audio data is started, etc. Resynchronization processing can be performed at a necessary timing. In addition, since the resync setting command unit 125 can command the resync setting to each speaker 2 in the zone to be synchronized, an efficient resynchronization process only for the speakers 2 in the zone requiring the resynchronization process. It can be performed.
　［第４実施形態］
　次に、図３１～図３４を参照し、本発明の第４実施形態について説明する。本実施形態は、音声伝送装置１から各スピーカー２への電力供給を行うことを特徴とする。以下、上記の各実施形態と異なる特徴点のみ説明する。なお、本実施形態において、上記の各実施形態と同様の構成部分については同様の符号を付し、詳細な説明を省略する。また、上記の各実施形態と同様の構成、部分について適用される変形例は、本実施形態についても同様に適用される。 [Fourth Embodiment]
Next, a fourth embodiment of the present invention will be described with reference to FIGS. The present embodiment is characterized in that power is supplied from the audio transmission device 1 to each speaker 2. Only the feature points different from the above embodiments will be described below. In the present embodiment, the same components as those in the above-described embodiments are denoted by the same reference numerals, and detailed description thereof is omitted. Moreover, the modification applied about the structure similar to each said embodiment and a part is applied similarly about this embodiment.
　図３１は、第４実施形態に係るスピーカーシステムＳＹの機能ブロック図である。なお、同図の矢印は、音声データの流れおよび電力の流れを示している。第４実施形態では、主に電力供給に関する機能について説明する。音声伝送装置１は、主な機能構成として、音声入力部１０３、音声伝送部１３１、電力供給部１３２および充電制御部１３３を備えている。 FIG. 31 is a functional block diagram of the speaker system SY according to the fourth embodiment. The arrows in the figure indicate the flow of audio data and the flow of power. In the fourth embodiment, functions related to power supply will be mainly described. The audio transmission device 1 includes an audio input unit 103, an audio transmission unit 131, a power supply unit 132, and a charging control unit 133 as main functional configurations.
　音声入力部１０３は、各音声入力源４から音声データを入力する。また、音声伝送部１３１は、入力された各音声データをフレームにパッキングし、デイジーチェーン伝送する。なお、音声伝送部１３１は、第１～第３実施形態の伝送制御部１０６，１１３，１２４として機能する。 The voice input unit 103 inputs voice data from each voice input source 4. In addition, the audio transmission unit 131 packs each input audio data into a frame and transmits the daisy chain. Note that the audio transmission unit 131 functions as the transmission control units 106, 113, and 124 of the first to third embodiments.
　電力供給部１３２は、ケーブル３を介して、デイジーチェーン接続された複数のスピーカー２内の２次電池２３２に充電を行う。また、充電制御部１３３は、複数のスピーカー２のうち充電対象として指定する対象スピーカーを時分割で切り替える（所定時間ごとに順次切り替える）ことにより、各スピーカー２への充電を制御する。具体的には、対象スピーカーへの充電開始からの経過時間をカウンタやＲＴＣ（Real　Time　Clock）等によってカウントし、所定の時間が経過した場合、充電完了と判定して、対象スピーカーを切り替える。また、充電中のスピーカー２から充電停止要求を受信した場合（２次電池２３２の残量が所定量を超えた場合）、充電完了と判定して対象スピーカーを切り替える。なお、対象スピーカーの指定は、デイジーチェーンＩＤ初期化処理（図５参照）によって付与したスピーカーＩＤを用いる。 The power supply unit 132 charges the secondary batteries 232 in the plurality of speakers 2 connected in a daisy chain via the cable 3. In addition, the charging control unit 133 controls charging to each speaker 2 by switching target speakers to be specified as charging targets among the plurality of speakers 2 in a time-sharing manner (sequentially switching every predetermined time). Specifically, the elapsed time from the start of charging the target speaker is counted by a counter, an RTC (Real Time Clock) or the like, and when a predetermined time has elapsed, it is determined that charging is complete and the target speaker is switched. Further, when a charge stop request is received from the speaker 2 being charged (when the remaining amount of the secondary battery 232 exceeds a predetermined amount), it is determined that charging is complete and the target speaker is switched. Note that the speaker ID assigned by the daisy chain ID initialization process (see FIG. 5) is used to specify the target speaker.
　一方、スピーカー２は、主な機能構成として、データ分離部２０２、デイジーチェーン出力部２０３、音声出力部２０７、電力制御部２３１および２次電池２３２を備えている。データ分離部２０２、デイジーチェーン出力部２０３および音声出力部２０７の音声データの流れについては、上記の各実施形態と同様である。 On the other hand, the speaker 2 includes a data separation unit 202, a daisy chain output unit 203, an audio output unit 207, a power control unit 231 and a secondary battery 232 as main functional configurations. The flow of audio data in the data separation unit 202, daisy chain output unit 203, and audio output unit 207 is the same as in each of the above embodiments.
　電力制御部２３１は、音声伝送装置１（充電制御部１３３）からケーブル３を介して電力供給を受け、２次電池２３２に充電する。また、電力制御部２３１は、充電対象となる対象スピーカーとして指定されていない場合、２次電池２３２の充電停止状態を保持する。２次電池２３２としては、リチウムイオン二次電池、リチウムイオンポリマー二次電池など各種充電式電池を適用可能である。 The power control unit 231 receives power supply from the audio transmission device 1 (charge control unit 133) via the cable 3 and charges the secondary battery 232. In addition, when the power control unit 231 is not designated as a target speaker to be charged, the power control unit 231 holds the charge stop state of the secondary battery 232. As the secondary battery 232, various rechargeable batteries such as a lithium ion secondary battery and a lithium ion polymer secondary battery are applicable.
　また、電力制御部２３１は、残量検出部２３１ａおよび充電停止要求送信部２３１ｂを有している。残量検出部２３１ａは、自スピーカー２の２次電池２３２の残量を検出する。充電停止要求送信部２３１ｂは、残量検出部２３１ａにより、２次電池２３２の残量が所定量（第１所定量）を超えたことを検出した場合、音声伝送装置１に対し、ケーブル３を介して充電停止要求を送信する。なお、２次電池２３２の残量が所定量を超えたことを検出した場合に代えて、またはこれに加え、消費割合が所定値以下となった場合（例えば、全電池容量の１／５以下となった場合など）に充電停止要求を送信しても良い。 Also, the power control unit 231 includes a remaining amount detection unit 231a and a charge stop request transmission unit 231b. The remaining amount detection unit 231a detects the remaining amount of the secondary battery 232 of the speaker 2 itself. When the charge stop request transmission unit 231b detects that the remaining amount of the secondary battery 232 exceeds a predetermined amount (first predetermined amount) by the remaining amount detection unit 231a, the charging stop request transmission unit 231b connects the cable 3 to the audio transmission device 1. A charge stop request is transmitted via In addition, instead of or in addition to the case where it is detected that the remaining amount of the secondary battery 232 exceeds a predetermined amount, when the consumption ratio becomes a predetermined value or less (for example, 1/5 or less of the total battery capacity) The charging stop request may be transmitted at the same time.
　次に、図３２を参照し、伝送路としてＬＡＮケーブルを流用する場合の例を、ＬＡＮケーブルピンアサインと、これに対する本実施形態の適用例について説明する。上記の通り、本実施形態のスピーカーシステムＳＹでは、ケーブル３として汎用のＬＡＮケーブル（10BASE-Tや100BASE-TXなど）を用いる。図３２（ａ）は、ストレート結線方式ＬＡＮケーブルのピンアサインを示す図である。同図に示すように一般的には、１，２，３，６番の２ペアのみが使用ラインとなっている。このため、本実施形態では、同図（ｂ）に示すように、未使用の４，５，７，８番の２ペアのラインを用いて、音声データの伝送および電力供給を行う。 Next, with reference to FIG. 32, an example of diverting a LAN cable as a transmission path, a LAN cable pin assignment, and an application example of this embodiment to this will be described. As described above, in the speaker system SY of the present embodiment, a general-purpose LAN cable (10BASE-T, 100BASE-TX, etc.) is used as the cable 3. FIG. 32A is a diagram showing pin assignment of a straight connection type LAN cable. As shown in the figure, generally, only two pairs of Nos. 1, 2, 3, and 6 are used lines. For this reason, in this embodiment, as shown in FIG. 4B, the audio data is transmitted and the power is supplied using two unused pairs of lines 4, 5, 7, and 8.
　次に、図３３および図３４のフローチャートを参照し、充電処理の流れについて説明する。図３３は、音声伝送装置１側の充電処理を示すフローチャートである。音声伝送装置１は、充電対象となる対象スピーカーをスピーカーＩＤによって指定し、対象スピーカーに通知する（Ｓ１４１）。その後、ＡＣＫ信号の受信タイムアウトの判別を行い（Ｓ１４２）、タイムアウトした場合は（Ｓ１４２：Ｙｅｓ）、エラー処理を行う（Ｓ１４３）。具体的には、音声出力または画面表示よってエラーを報知する。一方、タイムアウトせず（Ｓ１４２：Ｎｏ）、対象スピーカーからＡＣＫ信号を受信した場合は（Ｓ１４４：Ｙｅｓ）、充電を開始する（Ｓ１４５）。また、充電開始と同時に充電カウンタをセットし（Ｓ１４６）、カウントを終了した場合は（Ｓ１４７：Ｙｅｓ）、充電を終了して充電タイムアウトを対象スピーカーに通知する（Ｓ１４８）。その後、スピーカーＩＤをインクリメントし（Ｓ１４９）、次段のスピーカー２に対して、Ｓ１４１以降を繰り返す。一方、割り込み処理として、充電中のスピーカー２から充電停止要求を受信した場合も充電を終了し（Ｓ１５０）、スピーカーＩＤをインクリメントする（Ｓ１４９）。 Next, the flow of the charging process will be described with reference to the flowcharts of FIGS. FIG. 33 is a flowchart showing the charging process on the voice transmission device 1 side. The audio transmission device 1 designates the target speaker to be charged by the speaker ID and notifies the target speaker (S141). Thereafter, the reception timeout of the ACK signal is determined (S142). If the timeout has occurred (S142: Yes), error processing is performed (S143). Specifically, the error is notified by voice output or screen display. On the other hand, when the time-out does not occur (S142: No) and an ACK signal is received from the target speaker (S144: Yes), charging is started (S145). Further, the charging counter is set simultaneously with the start of charging (S146), and when the counting is ended (S147: Yes), charging is terminated and a charging timeout is notified to the target speaker (S148). Thereafter, the speaker ID is incremented (S149), and S141 and subsequent steps are repeated for the next-stage speaker 2. On the other hand, as an interruption process, when a charge stop request is received from the speaker 2 being charged, the charging is also terminated (S150), and the speaker ID is incremented (S149).
　図３４は、スピーカー２側の充電処理を示すフローチャートである。スピーカー２（対象スピーカー）は、スピーカーＩＤを受信すると（Ｓ１６１）、受信したスピーカーＩＤと自スピーカー２内に記憶しているＩＤとの整合性を判別する（Ｓ１６２）。整合した場合は（スピーカーＩＤが一致した場合は、Ｓ１６２：Ｙｅｓ）、ＡＣＫ信号を音声伝送装置１に返信し、音声伝送装置１から伝送された電力の供給を開始する（２次電池２３２への充電を行う，Ｓ１６４）。その後、２次電池２３２の電池残量をチェックし（Ｓ１６５）、充電Ｆｕｌｌになったと判定した場合（２次電池２３２の残量が所定量を超えたことを検出した場合）は（Ｓ１６６：Ｙｅｓ）、音声伝送装置１に対し充電停止要求を通知し（Ｓ１６７）、充電処理を終了する。一方、割り込み処理として、音声伝送装置１から充電タイムアウト通知を受信した場合も（Ｓ１６８）、充電処理を終了する。なお、図３３および図３４のフローチャートにおいて、各種通知およびＡＣＫ信号の送受信は、ＬＡＮケーブルの１，２，３，６番の２ペアのラインを用いる（図３２参照）。 FIG. 34 is a flowchart showing the charging process on the speaker 2 side. Upon receiving the speaker ID (S161), the speaker 2 (target speaker) determines the consistency between the received speaker ID and the ID stored in the speaker 2 (S162). If they match (if the speaker IDs match, S162: Yes), the ACK signal is returned to the audio transmission device 1 and the supply of power transmitted from the audio transmission device 1 is started (to the secondary battery 232). Charging is performed, S164). Thereafter, the remaining battery level of the secondary battery 232 is checked (S165), and when it is determined that the battery has become full (when it is detected that the remaining battery level of the secondary battery 232 exceeds a predetermined amount) (S166: Yes). ) The voice transmission device 1 is notified of a charge stop request (S167), and the charging process is terminated. On the other hand, also when the charging timeout notification is received from the audio transmission device 1 as the interrupt processing (S168), the charging processing is terminated. In the flowcharts of FIG. 33 and FIG. 34, transmission and reception of various notifications and ACK signals use two pairs of lines 1, 2, 3, and 6 of the LAN cable (see FIG. 32).
　以上説明したとおり、第４実施形態によれば、音声伝送装置１側で、充電対象として指定する対象スピーカー２を時分割で切り替える制御を行うことにより、１本のケーブル３のみで、各スピーカー２への電力供給を安定して行うことができる。これにより、無秩序に電力供給が行われることがなくなり、許容電流容量の問題や、給電ラインのインピーダンスによる電圧降下および発熱の問題を解消できる。また、充電対象となる対象スピーカーを、音声伝送時に用いるスピーカーＩＤの送信によって行うため、対象スピーカーの切り替えを容易且つ確実に行うことができる。 As described above, according to the fourth embodiment, by controlling the target speaker 2 designated as the charging target in a time-division manner on the audio transmission device 1 side, each speaker 2 can be achieved with only one cable 3. Can be stably supplied. As a result, power supply is not performed randomly, and the problem of allowable current capacity and the problem of voltage drop and heat generation due to the impedance of the power supply line can be solved. In addition, since the target speaker to be charged is transmitted by transmitting the speaker ID used at the time of audio transmission, the target speaker can be easily and reliably switched.
　また、電力供給を行うための伝送路として汎用のＬＡＮケーブルを用いるため、システム構成の更なる低廉化を図ることができる。また、各スピーカー２は、対象スピーカーとして指定されていない場合、２次電池２３２の充電停止状態を保持するため、システム全体としてより安定した電力供給を行うことができる。 In addition, since a general-purpose LAN cable is used as a transmission path for supplying power, the system configuration can be further reduced. In addition, when each speaker 2 is not designated as a target speaker, the charging stop state of the secondary battery 232 is maintained, so that more stable power supply can be performed as the entire system.
　また、音声伝送装置１は、充電開始からの経過時間をカウントして対象スピーカーを切り替えるため、平均的に十分充電されている状態を維持するための適切なサイクルで各スピーカー２への電力供給を行うことができる。また、充電中のスピーカー２から充電停止要求を受信した場合、これを割り込み処理として対象スピーカーを切り替えるため、過充電を防止することができると共に、各スピーカー２への電力供給を効率的に行うことができる。 In addition, since the audio transmission device 1 counts the elapsed time from the start of charging and switches the target speaker, the power transmission to each speaker 2 is performed in an appropriate cycle for maintaining a sufficiently charged state on average. It can be carried out. In addition, when a charge stop request is received from the speaker 2 being charged, this is used as an interrupt process to switch the target speaker, so that overcharging can be prevented and power supply to each speaker 2 can be efficiently performed. Can do.
　なお、上記の実施形態では、１台ずつ対象スピーカーを指定して充電を行うものとしたが、複数台ずつ対象スピーカーを指定しても良い。また、充電する順序も、上流側のスピーカー２から下流側に向かうのではなく、下流側のスピーカー２から上流側に向かっても良いし、スピーカー２の並び順に拘わらずランダムな順序で充電を行っても良い。また、対象スピーカーは、特定のゾーンに属するスピーカー２（例えば、再生対象となるスピーカー２）のみを対象として切替制御を行っても良い。 In the above embodiment, charging is performed by designating target speakers one by one. However, target speakers may be designated by a plurality of units. Also, the charging order may be from the downstream speaker 2 to the upstream side instead of going from the upstream speaker 2 to the downstream side, and the charging is performed in a random order regardless of the arrangement order of the speakers 2. May be. In addition, the target speaker may perform switching control for only the speaker 2 belonging to a specific zone (for example, the speaker 2 to be reproduced).
　また、上記の実施形態では、電力供給を行うための伝送路として汎用のＬＡＮケーブルを用いたが、専用ケーブルを用いても良い。また、対象スピーカーの指定も、音声伝送に用いるスピーカーＩＤを用いるのではなく、電力供給用に付与した専用スピーカーＩＤを用いても良い。また、スピーカーＩＤを用いて対象スピーカーを指定するのではなく、スピーカー２の並び順にしたがって順次対象スピーカー２を切り替える構成としても良い。 In the above embodiment, a general-purpose LAN cable is used as a transmission path for supplying power, but a dedicated cable may be used. In addition, for the designation of the target speaker, a dedicated speaker ID assigned for power supply may be used instead of the speaker ID used for audio transmission. Moreover, it is good also as a structure which does not designate a target speaker using speaker ID, but switches the target speaker 2 sequentially according to the arrangement order of the speaker 2. FIG.
　また、上記の実施形態には記載していないが、各スピーカー２において、非充電中に、２次電池２３２の残量が所定量（第２所定量）以下となったことを検出した場合、音声伝送装置１に対し、充電開始要求を送信しても良い（充電開始要求送信部）。この場合、充電制御部１３３は、非充電中のスピーカー２から充電開始要求を受信した場合、充電中のスピーカー２への充電を停止し、または充電中のスピーカー２への充電完了を待って、充電開始要求を行ったスピーカー２を対象スピーカーに切り替えることが好ましい。この構成によれば、非充電中のスピーカー２から充電開始要求を受信した場合、当該スピーカー２を次の対象スピーカー２とするため、充電不足を防止できる。これにより、システム全体として、安定した音声出力を行うことができる。 Although not described in the above embodiment, in each speaker 2, when it is detected that the remaining amount of the secondary battery 232 is equal to or less than a predetermined amount (second predetermined amount) during non-charging, You may transmit a charge start request | requirement with respect to the audio | voice transmission apparatus 1 (charge start request transmission part). In this case, when the charging control unit 133 receives a charging start request from the non-charging speaker 2, the charging control unit 133 stops charging the charging speaker 2 or waits for the charging of the charging speaker 2 to be completed. It is preferable to switch the speaker 2 that has requested charging start to the target speaker. According to this configuration, when a charge start request is received from the non-charged speaker 2, the speaker 2 is set as the next target speaker 2, and thus insufficient charging can be prevented. Thereby, the stable audio | voice output can be performed as the whole system.
　次に、図３５および図３６を参照し、本発明の変形例について説明する。図３５は、第１の変形例を示すシステム構成図である。同図に示すように、第１の変形例では、音声伝送装置１から２方向にケーブル３ａ、３ｂが伸びている。この場合、音声伝送装置１の背面には、デイジーチェーン出力を行うための出力端子１２が２個必要となる（図３（ａ）参照）。また、伝送する音声データは、ケーブル３ａ、３ｂごとに区別する必要はなく、同一の音声データを伝送すれば良い。この構成によれば、出力端子１２を複数個備え、両端子に同じデータを流すだけで、デイジーチェーン接続の自由度を向上させることができる。つまり、部屋の出入り口などケーブル３を配線できない箇所を避けることができ、部屋のレイアウトに与える制約を小さくできる。なお、音声伝送装置１の背面に備える出力端子１２の数は任意であり、３個以上であっても良い。 Next, a modification of the present invention will be described with reference to FIGS. FIG. 35 is a system configuration diagram showing a first modification. As shown in the figure, in the first modification, cables 3a and 3b extend in two directions from the audio transmission device 1. In this case, two output terminals 12 for performing daisy chain output are required on the back surface of the audio transmission device 1 (see FIG. 3A). The audio data to be transmitted need not be distinguished for each of the cables 3a and 3b, and the same audio data may be transmitted. According to this configuration, the degree of freedom of daisy chain connection can be improved by providing a plurality of output terminals 12 and only flowing the same data to both terminals. That is, it is possible to avoid places where the cable 3 cannot be routed, such as entrances and exits of rooms, and it is possible to reduce restrictions on the room layout. The number of output terminals 12 provided on the back surface of the audio transmission device 1 is arbitrary, and may be three or more.
　図３６は、第２の変形例を示すシステム構成図である。本変形例は、無指向性のスピーカー２を用いることを特徴とする。この構成により、１つの部屋に複数のマルチチャンネルスペースを形成できる。同図では特に図示しないが、テレビ８１と音声伝送装置１は接続されているものとする。また、プロジェクター８２のソース源と音声伝送装置１も接続されているものとする。同図に示すように、無指向性のスピーカー２を用いることで、例えば、ある時はテレビ８１でスポーツ観戦をし、ある時はプロジェクター８２で映画鑑賞をする、といった使い分けが、配線を変えることなく、音声伝送装置１の設定を変更するだけで実現できる。つまり、スピーカー２ａを、前者の場合、視聴者の背後に置くサラウンドスピーカー（右側）として利用すると共に、後者の場合、画面の左右に置くフロントスピーカー（左側）として利用する。また、スピーカー２ｂを、前者および後者の場合において、サラウンドスピーカー（左側）として利用する。これにより、同じスピーカー２を、複数の試聴環境で共有することができる。 FIG. 36 is a system configuration diagram showing a second modification. This modification is characterized by using an omnidirectional speaker 2. With this configuration, a plurality of multi-channel spaces can be formed in one room. Although not particularly shown in the figure, it is assumed that the television 81 and the audio transmission device 1 are connected. Further, it is assumed that the source source of the projector 82 and the audio transmission device 1 are also connected. As shown in the figure, by using the omnidirectional speaker 2, for example, watching sports on the television 81 at some times and watching a movie on the projector 82 at some times can change the wiring. Instead, it can be realized only by changing the setting of the voice transmission device 1. That is, in the former case, the speaker 2a is used as a surround speaker (right side) placed behind the viewer, and in the latter case, it is used as a front speaker (left side) placed on the left and right of the screen. The speaker 2b is used as a surround speaker (left side) in the former and latter cases. Thereby, the same speaker 2 can be shared in a plurality of trial listening environments.
　また、第２の変形例によれば、部屋の模様替えを手軽に行うことができる。例えば、部屋の中に小型のスピーカー２を壁掛けなどで複数個配置しておけば、上記のように音声伝送装置１の設定を変更するだけで、スピーカー２の使い方を変えることができる。また、設置当初はテレビ８１側の視聴環境に用いるスピーカー２だけを配置しておき、模様替えや試聴環境の追加（プロジェクター８２用など）を行う際に、その分のスピーカー２を追加し、音声伝送装置１の設定を変更することで、２箇所の試聴環境を作ることができる。 Also, according to the second modification, the room can be easily redesigned. For example, if a plurality of small speakers 2 are arranged in a room, such as on a wall, the usage of the speakers 2 can be changed simply by changing the settings of the audio transmission device 1 as described above. In addition, at the beginning of installation, only the speaker 2 used for the viewing environment on the TV 81 side is arranged, and when changing the pattern or adding a trial listening environment (for the projector 82, etc.), the corresponding speaker 2 is added and audio transmission is performed. By changing the setting of the device 1, two listening environments can be created.
　以上、４つの実施形態および２つの変形例を示したが、これらを適宜組み合わせたスピーカーシステムＳＹを実現しても良い。また、上記の各実施形態に示したスピーカーシステムＳＹ（音声伝送装置１またはスピーカー２）における各部および各機能をプログラムとして提供することが可能である。また、そのプログラムを各種記録媒体（ＣＤ－ＲＯＭ、フラッシュメモリ等）に格納して提供することも可能である。すなわち、コンピューターを、音声伝送装置１またはスピーカー２の各部として機能させるためのプログラム、およびそれを記録した記録媒体も、本発明の権利範囲に含まれるものである。また、上記の実施例によらず、スピーカーシステムＳＹのシステム構成、音声伝送装置１およびスピーカー２の装置構成や処理工程等について、本発明の要旨を逸脱しない範囲で、適宜変更が可能である。 As mentioned above, although four embodiment and two modifications were shown, speaker system SY which combined these suitably may be realized. Moreover, it is possible to provide each part and each function in the speaker system SY (the audio transmission device 1 or the speaker 2) shown in each embodiment as a program. Further, the program can be provided by being stored in various recording media (CD-ROM, flash memory, etc.). That is, a program for causing a computer to function as each part of the audio transmission device 1 or the speaker 2 and a recording medium on which the program is recorded are also included in the scope of the right of the present invention. Moreover, the system configuration of the speaker system SY, the device configurations of the audio transmission device 1 and the speaker 2, processing steps, and the like can be appropriately changed without departing from the gist of the present invention, regardless of the above embodiment.
　１…音声伝送装置　２…スピーカー　３…ケーブル　４…音声入力源　１１…電源供給口　１２…出力端子　２１…振動部材　２２…入力端子　２３…出力端子　１０１…ＩＤ初期化部　１０２…スピーカーアサイン設定部　１０３…音声入力部　１０４…チャンネル情報取得部　１０５…チャンネル情報設定部　１０６…伝送制御部　１１１…周波数変換部　１１２…装置側メモリ　１１３…伝送制御部　１２１…オプションＩＤアサイン設定部　１２２…装置側バッファ　１２３…バッファ監視部　１２４…伝送制御部　１２５…リシンク設定指令部　１３１…音声伝送部　１３２…電力供給部　１３３…充電制御部　２０１…ＩＤ記憶部　２０２…データ選択取得部　２０３…デイジーチェーン出力部　２０４…データバッファ　２０５…補間データ生成部　２０６…選択回路　２０７…音声出力部　２０８…スピーカー側メモリ　２１１…位相同期部　２１２…カウンタ部　２１３…スタンバイ制御部　２２１…スピーカー側バッファ　２２２…バッファ制御部　２２３…リシンク設定部　２２４…同期処理部　２３１…電力制御部　２３１ａ…残量検出部　２３１ｂ…充電停止要求送信部　２３２…２次電池ＳＹ…スピーカーシステム DESCRIPTION OF SYMBOLS 1 ... Audio | voice transmission apparatus 2 ... Speaker 3 ... Cable 4 ... Audio | voice input source 11 ... Power supply port 12 ... Output terminal 21 ... Vibrating member 22 ... Input terminal 23 ... Output terminal 101 ... ID initialization part 102 ... Speaker assignment setting part 103 ... voice input unit 104 ... channel information acquisition unit 105 ... channel information setting unit 106 ... transmission control unit 111 ... frequency conversion unit 112 ... device side memory 113 ... transmission control unit 121 ... option ID assignment setting unit 122 ... device side buffer 123 ... Buffer monitoring unit 124 ... Transmission control unit 125 ... Resync setting command unit 131 ... Audio transmission unit 132 ... Power supply unit 133 ... Charge control unit 201 ... ID storage unit 202 ... Data selection acquisition unit 203 ... Daisy chain output unit 204 ... Data buffer 2 5 ... Interpolation data generation unit 206 ... Selection circuit 207 ... Audio output unit 208 ... Speaker side memory 211 ... Phase synchronization unit 212 ... Counter unit 213 ... Standby control unit 221 ... Speaker side buffer 222 ... Buffer control unit 223 ... Resync setting unit 224 ... Synchronization processing unit 231 ... Power control unit 231a ... Remaining amount detection unit 231b ... Charge stop request transmission unit 232 ... Secondary battery SY ... Speaker system

Claims (10)

  1. 　１本の伝送路で直列接続された複数のスピーカーと、
    　複数の音声入力源から入力された音声データを、前記伝送路を介して各スピーカーに伝送する音声伝送装置と、を備えたスピーカーシステムであって、
    　前記音声伝送装置は、
    　前記各音声入力源にそれぞれ対応して設けられ、各音声入力源の再生クロックを一定の周波数クロックに変換する複数の周波数変換部と、
    　前記各周波数変換部による周波数変換後の各音声データをバッファリングする装置側メモリと、
    　前記装置側メモリから読み出した前記各音声データを、１フレームにパッキングし、前記一定の周波数クロックに対応する単一クロックで伝送する伝送制御部と、を備えたことを特徴とするスピーカーシステム。     A plurality of speakers connected in series with one transmission line;
    An audio transmission device that transmits audio data input from a plurality of audio input sources to each speaker via the transmission path, and a speaker system comprising:
    The audio transmission device is
    A plurality of frequency converters provided corresponding to the respective audio input sources, and converting a reproduction clock of each audio input source into a constant frequency clock;
    A device-side memory for buffering each audio data after frequency conversion by each frequency conversion unit;
    A speaker system, comprising: a transmission control unit that packs each of the audio data read from the device-side memory into one frame and transmits the data with a single clock corresponding to the constant frequency clock.
  2. 　前記音声伝送装置が、前記音声入力源からアナログ信号を入力する場合、Ａ／Ｄコンバーターに使用する周波数クロックは、前記一定の周波数クロックであることを特徴とする請求項１に記載のスピーカーシステム。 The speaker system according to claim 1, wherein when the audio transmission device inputs an analog signal from the audio input source, a frequency clock used for an A / D converter is the constant frequency clock.
  3. 　前記各スピーカーは、
    　前記音声伝送装置により伝送されたフレームから、シンクデータおよび自スピーカー用の音声データを分離するデータ選択取得部と、
    　選択取得された前記シンクデータに同期させて内部クロックを生成する位相同期部と、を備えたことを特徴とする請求項１に記載のスピーカーシステム。     Each speaker is
    A data selection / acquisition unit for separating sync data and audio data for the speaker from the frame transmitted by the audio transmission device;
    The speaker system according to claim 1, further comprising: a phase synchronization unit that generates an internal clock in synchronization with the selected and acquired sync data.
  4. 　前記各スピーカーは、
    　選択取得された前記音声データをバッファリングするスピーカー側メモリと、
    　前記スピーカー側メモリからの前記音声データの出力タイミングを計測するカウンタ部と、をさらに備えたことを特徴とする請求項３に記載のスピーカーシステム。     Each speaker is
    Speaker-side memory for buffering the selected and acquired audio data; and
    The speaker system according to claim 3, further comprising a counter unit that measures an output timing of the audio data from the speaker-side memory.
  5. 　前記フレームとして伝送される各音声データには、該当データの有効／無効を示すイネーブルフラグが定義されており、
    　前記各スピーカーは、
    　前記イネーブルフラグ「無効」の検出が一定時間続いた場合、自スピーカーをスタンバイ状態にするスタンバイ制御部をさらに備えたことを特徴とする請求項１に記載のスピーカーシステム。     Each audio data transmitted as the frame is defined with an enable flag indicating validity / invalidity of the corresponding data,
    Each speaker is
    The speaker system according to claim 1, further comprising a standby control unit that puts the speaker in a standby state when detection of the enable flag “invalid” continues for a predetermined time.
  6. 　前記各スピーカーは、音声アンプ機能を備えたことを特徴とする請求項１に記載のスピーカーシステム。 The speaker system according to claim 1, wherein each of the speakers has an audio amplifier function.
  7. 　請求項１ないし６のいずれか１項に記載のスピーカーシステムに用いられることを特徴とする音声伝送装置。 An audio transmission device used in the speaker system according to any one of claims 1 to 6.
  8. 　請求項１ないし６のいずれか１項に記載のスピーカーシステムに用いられることを特徴とするスピーカー。 A speaker used for the speaker system according to any one of claims 1 to 6.
  9. 　１本の伝送路で直列接続された複数のスピーカーに対し、複数の音声入力源から入力された音声データを、前記伝送路を介して伝送する音声伝送方法であって、
    　前記各音声入力源にそれぞれ対応して設けられた複数の周波数変換部により、各音声入力源の再生クロックを一定の周波数クロックに変換する周波数変換工程と、
    　周波数変換後の各音声データを装置側メモリにバッファリングするバッファリング工程と、
    　前記装置側メモリから読み出した前記各音声データを、１フレームにパッキングし、前記一定の周波数クロックに対応する単一クロックで伝送する伝送制御工程と、を実行することを特徴とする音声伝送方法。     An audio transmission method for transmitting audio data input from a plurality of audio input sources to a plurality of speakers connected in series on a single transmission line, via the transmission line,
    A frequency conversion step of converting a reproduction clock of each audio input source into a constant frequency clock by a plurality of frequency conversion units provided corresponding to the respective audio input sources,
    A buffering step of buffering each audio data after frequency conversion in the device-side memory;
    And a transmission control step of packing each of the audio data read out from the device-side memory into one frame and transmitting it with a single clock corresponding to the fixed frequency clock.
  10. 　コンピューターに、請求項９に記載の音声伝送方法における各工程を実行させるためのプログラム。 A program for causing a computer to execute each step in the audio transmission method according to claim 9.
PCT/JP2011/004383 2011-08-03 2011-08-03 Speaker system, audio transmission device, speaker, audio transmission method and program WO2013018139A1 (en)

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CN103929691A (en) * 2014-03-20 2014-07-16 联想(北京)有限公司 Information processing method and electronic equipment

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JP2005175745A (en) * 2003-12-10 2005-06-30 Sony Corp Speaker management information acquisition method in acoustic system, acoustic system, server, and speaker
JP2009272887A (en) * 2008-05-07 2009-11-19 Canon Inc Transmission device and method

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JP2005175745A (en) * 2003-12-10 2005-06-30 Sony Corp Speaker management information acquisition method in acoustic system, acoustic system, server, and speaker
JP2009272887A (en) * 2008-05-07 2009-11-19 Canon Inc Transmission device and method

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Publication number Priority date Publication date Assignee Title
CN103929691A (en) * 2014-03-20 2014-07-16 联想(北京)有限公司 Information processing method and electronic equipment

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