Abstract:
A wireless digital audio distribution system includes a transmitter that performs parallel acquisition of audio data from plural audio data channels. The acquired data is stored to a first buffer during intervals defined by a first clock unit. A radio transmitter provides for the packetization of the parallel collected data and the inclusion of a timing marker in a predetermined packet of data transmitted during an interval. A receiver unit delivers audio output data corresponding to one of the audio data channels. The receiver unit includes a radio receiver for receiving packetized data and storing second parallel collected data to said second buffer. A demultiplexer is coupled to the second buffer to select the audio output data corresponding to the selected one of the plural audio data channels, whereby the audio output data is available for the selective reproduction of the chosen audio data channel.

Description:
[0001]     This application claims the benefit of U.S. Provisional Application No(s). 60/705,723 and 60/705,724, all filed Aug. 4, 2005.  
       CROSS-REFERENCE TO RELATED APPLICATIONS  
       [0002]     The present application is related to High Quality. Controlled Latency Multi-Channel Wireless Digital Audio Distribution System and Methods, Ser. No. ______, filed Aug. 4, 2006 and assigned to the Assignee of the present Application. 
     
    
     BACKGROUND OF THE INVENTION  
       [0003]     1. Field of the Invention  
         [0004]     The present invention is generally related to the wireless distribution of high-quality audio signals and, in particular to a system and methods of distributing high bit rate, multi channel, audio wirelessly while maintaining a constant, low, playback to source latency and channel to channel phase coherency.  
         [0005]     2. Description of the Related Art  
         [0006]     In the audio space there are many places that latency, high quality, and more than two channels are critical to the quality of the experience. It is also difficult to retrofit standard spaces with cables to the support multiple channels of audio. Today&#39;s definition of high end audio in the Home Theater space is 7 channels of audio samples at 48,000 samples per second with 24 bits of data per sample. Further, the marketplace is rapidly maturing from 5.1 (6 channel) to 11.1 (12 channel) sound system requirements.  
         [0007]     Conventional wireless solutions rely on simple, low-cost radio technologies, such as frequency modulation (FM) and basic spread spectrum modulation schemes. The consequence of this is a reduction in the number of bits used for each audio sample, with a corresponding reduction in dynamic range and audio quality.  
         [0008]     A critical requirement exists in both spaces to minimize and establish a constant or fixed latency in the system and to keep all channels aligned in time. Latency refers to time delays measured from audio source-to-output and from channel-to-channel. Source-to-output delays are a problem for all sound venues including, in particular, Home Theater and other video/audio systems, where the audio program material is synchronized to a video screen (“lip-sync”). Acoustics engineers generally consider source-to-output delays greater than 10 milliseconds to be noticeable. As for latency from channel-to-channel, the human ear is extremely sensitive to these phase delays and experts describe audio delivered with channel-to-channel delays greater than 1 millisecond as sounding “disjointed” or “blurry”.  
         [0009]     The same data and sampling rate are in use in recording and sound reinforcement, only the desired number of channels is generally between 8 and 32. In conferencing use, the latency and wireless requirement are compounded by a need for accurate routing of audio paths with intelligent addition of signals and echo cancellation.  
         [0010]     Consequently, there is a clear need to solve all of these problems in a wireless audio distribution system.  
       SUMMARY OF THE INVENTION  
       [0011]     Thus, a general purpose of the present invention is to provide an efficient wireless, high bit rate, multi channel, audio system capable of maintaining constant, low, playback to source latency while further maintaining channel to channel phase coherency.  
         [0012]     This is achieved in the present invention by providing a wireless digital audio distribution system that includes a transmitter that performs parallel acquisition of audio data from plural audio data channels. The acquired data is stored to a first buffer during intervals defined by a first clock unit. A radio transmitter provides for the packetization of the parallel collected data and the inclusion of a timing marker in a predetermined packet of data transmitted during an interval. A receiver unit delivers audio output data corresponding to one of the audio data channels. The receiver unit includes a radio receiver for receiving packetized data and storing second parallel collected data to said second buffer. A demultiplexer is coupled to the second buffer to select the audio output data corresponding to the selected one of the plural audio data channels, whereby the audio output data is available for the selective reproduction of the chosen audio data channel.  
         [0013]     An advantage of the present invention is base configurations are immediately capable of distributing 16 channels of audio with a full 24 bits per sample and 48,000 samples per second.  
         [0014]     Another advantage of the present invention is the initial preferred embodiments are capable of achieving a fixed, repeatable inter-channel differential latency of less than 0.001 millisecond and a fixed, repeatable source to speaker latency of less than 2 milliseconds.  
         [0015]     A further advantage of the present invention is that it enables multi-channel audio sources to be placed “out-of-view”, while supporting a full complement of audio speakers to be installed throughout a room without wires. Costly physical rewiring is not required.  
         [0016]     Still another advantage of the present invention is that the audio playback delays can be precisely adjusted and maintained in fixed relation to “tune” audio phasing for specific listener/speaker positions and room acoustics.  
         [0017]     Yet another advantage of the present invention is that the transmitters and receivers, as implemented in the preferred embodiments, can and will coexist with present wireless networking systems without introducing interference, without loss of audio fidelity, and while meeting all FCC and CSA certification requirements.  
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0018]      FIG. 1  is a block diagram illustrating a system implementation of a preferred embodiment of the present invention;  
         [0019]      FIG. 2  is a flow diagram illustrating the pipeline processing of data through a wireless transmitter and receiver in accordance with a preferred embodiment of the present invention;  
         [0020]      FIG. 3  is a block diagram of a wireless audio packet content transmitter unit constructed in accordance with a preferred embodiment of the present invention;  
         [0021]      FIG. 4  is a block diagram of a wireless audio packet content transmitter unit constructed in accordance with a preferred embodiment of the present invention;  
         [0022]      FIGS. 5A and 5B  are block diagrams illustrating the preferred flow of data through transmitter and receiver units as implemented in a preferred embodiment of the present invention; and  
         [0023]      FIG. 6  is a flow diagram showing the bit parallel packing and of channel data in data words to build audio data packets for transition and corresponding unpacking on reception as implemented in a preferred embodiment of the present invention. 
     
    
     DETAILED DESCRIPTION OF THE INVENTION  
       [0024]     The present invention provides for the packet transmission of audio data from a transmitter, typically coupled to a multiple channel audio data source, to a set of wireless packet data receivers. The receivers are programmable to associate operation with an assigned transmitter. The receivers are further programmable to select and decode a specified channel or channels of the transmitted multiple channel content. In preferred configuration, a separate receiver is provided for each audio reproduction speaker in a sound system and, dependent on the speaker type and placement, selects and decodes a corresponding channel of the audio content. Receivers associated with the center channel, base, various left and right side and rear effects speakers each preferably decode respective audio content channels provided through the transmitter for respective speakers.  
         [0025]     The transmitters and receivers of the present invention preferably support both digital or analog format inputs and outputs for audio data. In particular, the receivers of the present invention provide may be integrated into the speaker enclosures and closely integrated with the speaker amplification system. That is, wireless transmission of audio content while maintaining high audio fidelity enables audio component manufacturers to locate and isolate speaker amplifiers internal to the speaker enclosures. This removes the “hot and heavy” power sources and amplifiers from audio source appliances. Migration of these components out to the speakers themselves enables manufactures to fully implement modern digital switching amplifier topologies, including specifically Class D amplifier designs, in the speakers. This will enable fundamental improvements in sound reproduction while achieving reduced size, cost, power consumption, and EMI radiation in all system components. Users also gain the advantages of flexible installation and reconfiguration.  
         [0026]     The transmitters and receivers used in the preferred embodiments are preferably based on the high-volume commodity radio components used in conventional wireless networking systems, such as IEEE 802.11g and 802.11n. For purposes of implementation, the present invention provides for the replacement of the conventional Media Access Control (MAC) layer with a data processing engine specifically designed to deliver high bit rate isochronous data, such as audio and video, with low latency in accordance with the present invention. Clock capture and alignment by the data processing engine of the present invention is further described in the co-pending application, High Quality Controlled Latency Multi-Channel Wireless Digital Audio Distribution System and Methods, Ser. No. ______, filed concurrently herewith, assigned to the assignee of the present invention, which is hereby incorporated by reference.  
         [0027]     The system and methods of the present invention utilize a basic architecture that can be configured for operation in multiple ways. All configurations are generally based on the same elements. The overall system configuration is shown in  FIG. 1 .  
         [0028]     A set of audio input sources delivers audio signals, encoded either as analog or digital audio, to a transmitter unit. The transmitter unit contains digitization, synchronization, buffering, MAC (Media Access Controller), and RF transmitter elements. A receiver unit contains RF receiver, MAC, buffering, synchronization and audio output elements. The audio output can be decoded for analog speakers through a digital-to-analog signal conversion or appropriately transcoder for digital audio uses. The resulting audio outputsignals are provided as outputs.  
         [0029]      FIG. 2  shows a preferred method of transmission and reception of audio data packets as implemented by a master transmitter unit with receipt and playback on a slave receiver unit. The following describes operation at the corresponding stages illustrated in  FIG. 2 . 
        1) The samples are collected from the CODECS or Digital Audio Interfaces into a Sample Block Buffer. The data in the Sample Block Buffer preferably implements data redundancy injection for Forward Error Correction (FEC) and is organized into a Send Buffer.     2) The Sample Block Buffer (Send  1 ) is transmitted over the radio link as a packet. The Block may be sent more than once (Send  2 ) to provide data redundancy. The first Data Block sent in this mode of operation will have its Sample Block Marker bits set.     3) When the receiver radio and MAC decode a valid Sample Block with the Marker bits set the MAC will trigger a Sample Block Marker at a delay determined during the initialization of the radio link. The delay will provide a Sample Block Marker at Sample Block boundaries.     4) The Sample Block is played starting at the Sample Block Marker generated in step  3 . The received Data Buffer is processed through a convolutional decoder and the resulting data is checked and repaired by use of the FEC methods employed and is returned to being a Sample Block that can the be sequence for playing.     5) The entire sample Block is sequenced out. The three phases of collection, transport, and playback are pipelined such that every step is running simultaneously.          
         [0035]     A preferred embodiment of a transmitter unit, constructed in accordance with the present invention, is shown in  FIG. 3 . A set of CODECs (CODer/DECoders) convert the incoming multi-channel analog signals to serial digital bit streams. If a digital audio input signal is used, the CODECs can be bypassed.  
         [0036]     In the particular embodiment of this invention shown in  FIG. 3 , a 48 kHz clock, derived from an 18.432 MHz crystal oscillator, is used for the sampling of analog audio data. On each sample clock interval, the data present on each of the CODEC serial outputs is captured at the input of the MUX &amp; FIFO. This block multiplexes the multiple codec inputs, and places them into a FIFO buffer for transmission to the radio transmitter MAC interface. Preferably, as data is added to the FIFO buffer, a forward error correction value is calculated and associated with the buffered data. In this particular implementation, the 18.432 MHz crystal is used to provide a bus interface clock to the MAC interface section.  
         [0037]     A 1 kHz Sample Block Marker output is also derived from the crystal oscillator. It is used to indicate the start of a block of samples 1 msec in duration. All data sampled from the CODECs during a Sample Block interval is presented to the Radio MAC during the duration of the current interval. Thus, the 1 msec duration of data collected in one interval is provided to, packetized, and transmitted by the MAC during the following interval. The preamble of the initial packet sent during an interval is marked with the Sample Block Marker. Copies of this packet may be re-transmitted, though without the Sample Block Marker, to provide data redundancy for the receivers. The number of copies re-transmitted may be programmatically defined, preferably provided retransmission can be completed within the balance of the current interval.  
         [0038]     A preferred embodiment of a receiver unit, constructed in accordance with the present invention, is shown in  FIG. 4 . The Radio MAC interface in the receiver delivers a Sample Block Marker that indicates the start of a block of sampled data. In the particular embodiment of this invention described by the illustration, this Sample Block Marker is a 1 kHz signal.  
         [0039]     When the Clock Divider Block receives the Sample Block Marker, the counter/divider from which the Sample Clock is derived is asynchronously reset to zero. In this particular example, it insures a phase alignment for the sample clock of ±54.3 nsec, or one period of the 18.432 MHz clock.  
         [0040]     The Sample Block Marker also indicates the start of a block of sampled audio data to the FIFO &amp; DEMUX section. In this particular implementation, the 18.432 MHz crystal is used to provide a bus interface clock to this section.  
         [0041]     The radio receiver MAC will load the FIFO buffer with new data. Forward error correction will be applied. If the error correction fails to correct the data packet contents, a redundant packet is corrected and loaded into the FIFO instead. If a packet and redundant copies are entirely lost or beyond correct, interpolation is used to cover the data error duration. The data delivered to the FIFO will be demultiplexed for synchronous transmission to the CODECs. All data received from the Radio MAC during the duration of the current Sample Block Interval will be sent to the CODECs during the duration of the current interval. CODECs for only the number of channels supported by a receiver unit need be physically populated in the receiver unit. Where a receiver unit supports only a single channel, as is expected for the typical case, that one channel is demultiplexed from the FIFO and provided to a single CODEC.  
         [0042]     The operational data transfer sequence, as described for the preferred embodiments of the present invention, thus ensures a maximum latency of two intervals. In the particular embodiment of this system described in the illustrations, this limits the overall latency to 2 msec with a 54.3 nsec accuracy for all receiver channels.  
         [0043]     The preferred data sequencing operation of the transmitter unit is shown in  FIG. 5A . The corresponding preferred data sequencing operation of the receiver is shown in  FIG. 5B . For the preferred embodiments of the present invention, the channel data is processed through the multiplexer interface by packing data bits from each channel in parallel into data words that are then collected into a data packet for transmission, as generally shown in  FIG. 6 . In a simple case, sixteen audio input channels are input. The channel data is separately serialized with a bit per channel being stored as a respective bit in parallel in a data word. If the per data channel sample resolution is 12 bits, then twelve data words are used to transmit a single sample from each of the sixteen channels.  
         [0044]     On the receiver side, the channel demultiplexer reconstructs the channel samples. Where, for typical implementations, the receiver unit is intended to support only a single or reduced set of channels, only those selected channels are output from the demultiplexer and processed, as appropriate, through transcoders for production of digital audio data streams, or through CODECs to produce analog audio data signals.  
         [0045]     Thus, a system and methods for providing for the distribution of high bit rate, multi channel, audio wirelessly while maintaining a constant, low, playback to source latency and channel to channel phase coherency operable in multiple configurations has been described.  
         [0046]     In view of the above description of the preferred embodiments of the present invention, many modifications and variations of the disclosed embodiments will be readily appreciated by those of skill in the art. It is therefore to be understood that, within the scope of the appended claims, the invention may be practiced otherwise than as specifically described above.