Patent Abstract:
The invention relates to the field of wireless communications and more specifically to an high quality, low power wireless audio system. More specifically, the invention comprises an audio source for receiving audio signals (e.g. music) and audio status information (e.g. song title) from a first external device (e.g. an MP3 player) and transmitting the audio signals and the audio status information over a wireless connection; and at least one audio sink for receiving the audio signals and said audio status information from the audio source and communicating the audio signals and the audio status information to a second external device (e.g. headphones), wherein a specified one of the at least one audio sink receives audio control information (e.g. pause) from the second external device and transmits said audio control information to said audio source via said wireless connection. Among other features, the wireless audio system of the present invention incorporates dynamic channel selection as well as dynamic adjustment of the transmission interval to ensure enhanced audio quality using the lowest possible power.

Full Description:
BACKGROUND OF THE INVENTION 
     1. Field of Invention 
     The invention relates to the field of wireless communications and more specifically to an enhanced wireless audio system. 
     2. Description of the Related Prior Art 
     As shown in  FIG. 1 , in its simplest form, a typical portable digital wireless audio system comprises two devices: a personal wireless audio player  100  which communicates with a wireless remote control  140  or wireless headphones  120 . The audio data is a unidirectional stream from personal wireless audio player  100 , the audio source, to wireless headphones  120 , the audio sink. In the case of wireless remote control  140 , personal wireless audio player  100 , the audio source, may also send LCD display information to wireless remote control  140 , the audio sink, and the audio sink will return packets containing key-press information (e.g. audio track selection). 
     Current radio technology that has the capacity to carry high quality stereo audio consumes too much power to make the use of wireless headphone  120  or wireless remote control  140  in conjunction with wireless personal audio player  100  successful in the marketplace. Presently, most personal audio players such as CD players, Mini-Disk players, and MP3 players are not wireless with a headphone or remote control being corrected to the player via a hard wire. Such personal audio players are intended to be mobile, (i.e. easily carried by the user) and are powered from a battery to allow for such portability. Currently, the vast majority of such personal audio products use one battery and the subtending headphone (and possibly remote control) receive their power from the battery in the player through the hard wire. 
     The hard wire connecting the player to the headphone and/or remote control is often an inconvenience to the user. For example, when the user wishes to put the player into a pocket, backpack or briefcase, the wires to the headphone or remote control must extend out to connect to the headphone or remote control. In addition, the wires tend to get tangled or snagged. 
     If the wire extending from the player is eliminated, the headphone/remote control then require their own power and one of the components that the battery will have to supply is the radio interface. Personal audio manufacturers have stated that wireless headphones and remote controls must be small, lightweight, and operate for 100 hours before the battery needs to be replaced. 100 hours of operation from a 450 mAHr 3v supply (2 CR2032 Li coin-cell batteries) requires the headphone and/or remote control to consume an average of no more than about 6 mA from a 2v supply, of which about 4 mA is available for the radio. Current radio technology consumes on the order of 20 mA or more so does not meet the standard suggested by manufacturers. 
     While power consumption is the main hurdle facing wireless (i.e. radio) solutions for personal audio applications, such solutions must also deliver high quality audio, deal with interference from a plethora of other radio sources, and be small and inexpensive. 
     SUMMARY OF THE INVENTION 
     In order to overcome the deficiencies of the prior art there is provided a high quality, low power personal wireless audio system which incorporates a variety of enhancements which serve to improve the overall audio experience for the user. Such features as acknowledged packet transmission with retransmission, dynamic adjustment of the transmission interval between the audio source and sink, improved audio synchronization, lossless compression, dynamic channel selection and switching, and dynamic adjustment of the transmit power allow the wireless audio system to quickly overcome identified radio interference and transmit a signal whose strength is adjusted according to the surrounding transmission medium. 
     In accordance with one aspect of the invention there is provided a wireless audio system comprising: (a) an audio source for receiving audio signals and audio status information from a first external device and transmitting the audio signals and the audio status information over a wireless connection; and (b) at least one audio sink for receiving the audio signals and the audio status information from the audio source and communicating the audio signals and the audio status information to a second external device, wherein a specified one of the at least one audio sink receives audio control information from the second external device and transmits the audio control information to the audio source via the wireless connection. 
     In accordance with a second aspect of the invention, there is provided an audio source comprising: (a) a source packet formatter and buffer communicating with a first external device, wherein the source packet formatter creates a plurality of source transmit packets containing audio signals and audio status information, and wherein the source buffer stores the plurality of source transmit packets prior to transmission to an audio sink; (b) a source transmitter communicating with the source packet formatter and buffer for receiving the plurality of source transmit packets from the source buffer and transmitting the plurality of source transmit packets to the audio sink every defined unit of time; (c) a source audio synchronizer communicating with the source transmitter for defining the unit of time for the source transmitter; (d) a source receiver for receiving audio control information from the audio sink, wherein the audio control information is in the form of a plurality of sink transmit packets, and wherein the source receiver communicates with the source transmitter to coordinate the receiving and transmitting within respective ones of said defined units of time; (e) a source packet de-formatter communicating with the source receiver for receiving the plurality of sink transmit packets and extracting the audio control information; and (f) a source channel quality monitor communicating with the source packet de-formatter and source transmitter for monitoring a specified one of a plurality of available channels. 
     Preferably, if an analog audio signal is delivered from the first external device the wireless audio system further comprises an analogue-to-digital converter (ADC) communicating with the first external device, and wherein a source audio sampling clock signal generated by the source audio synchronizer is fed to the ADC. 
     More preferably, the wireless audio system of further comprises a compression module communicating with the ADC for compressing digital audio signals outputted by the ADC. 
     In accordance with a third aspect of the invention, there is provided an audio sink comprising: (a) a sink receiver for receiving a plurality of source transmit packets from an audio source every defined unit of time, wherein said plurality of source transmit packets comprise audio signals and audio status information; (b) a sink packet de-formatter and buffer communicating with the sink receiver, wherein the sink packet de-formatter extracts the audio signals and the audio status information from the source transmit packets, and wherein the sink buffer stores the extracted audio signals and the audio status information; (c) a sink audio synchronizer communicating with the sink receiver for defining the unit of time for the sink receiver; (d) a sink packet formatter communicating with a second external device for creating a plurality of sink transmit packets containing audio control information; (e) a sink transmitter communicating with the sink packet formatter for transmitting the plurality of sink transmit packets, wherein the sink receiver communicates with the sink transmitter to coordinate the receiving and transmitting within specified ones of the defined units of time; and (f) a sink channel quality monitor communicating with the sink packet de-formatter and buffer and the sink receiver for monitoring a specified one of a plurality of available channels. 
     Preferably, if an analog audio signal is required by the second external device, the wireless audio system further comprises a digital-to-analogue converter (DAC), and wherein a sink audio sampling clock signal generated by the sink audio synchronizer is fed to said DAC. 
     More preferably, the wireless audio system further comprises a decompression module communicating with the sink packet de-formatter and buffer for decompressing the digital audio signals received from the sink buffer. 
     The advantages of the invention are now readily apparent. The enhanced wireless audio system provides uninterrupted audio play using the lowest possible power through a variety of integrated features which operate seamlessly to provide a user with a superior wireless audio experience. 
     Further features and advantages of the invention will be apparent from the detailed description which follows together with the accompanying drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       A better understanding of the invention will be obtained by considering the detailed description below, with reference to the following drawings in which: 
         FIG. 1  depicts a graphical representation of a typical wireless digital audio system; 
         FIG. 2  depicts a block diagram of wireless digital audio system; 
         FIG. 3  depicts a block diagram of the audio source of  FIG. 2 ; 
         FIG. 4  depicts a block diagram of the audio sink of  FIG. 2 ; 
         FIG. 5  depicts a wireless media duty cycle in accordance with the present invention; 
         FIG. 6  depicts a packet acknowledgement scheme in accordance with the present invention; 
         FIG. 7  depicts the audio buffer operation in accordance with the present invention; 
         FIG. 8  depicts a graph showing the relationship between retransmission bandwidth availability and transport super frame size; 
         FIG. 9  depicts a graph showing the relationship between overhead capacity and transport super frame size; 
         FIG. 10  depicts a graph showing the relationship between interference robustness and transport super frame size; 
         FIG. 11  depicts examples of varying TSF lengths used to accommodate interference in the wireless shared media; 
         FIG. 12  depicts a chart depicting the thresholds at which dynamic channel switching will be initiated; 
         FIG. 13  depicts a decision matrix used by the Channel Quality Monitor while transmission is active to determine the appropriate interference avoidance action; 
         FIG. 14  depicts graphically the application of the dynamic transmit power feature of the present invention; and 
         FIG. 15  depicts an alternate embodiment of the audio source which supports bi-directional audio transmission. 
     
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENT 
     Referring to  FIG. 2 , the wireless audio system of the present system generally comprises an Audio Source  200 , Audio Sink  210  and wireless connection (shown generally at  220 ):
     (a) Audio Source  200  receives analog or digital audio and audio status information from an external device (e.g. audio player—not shown) and transmits it to audio sink  210  over wireless connection  220 , and receives audio control information from audio sink  210  over wireless connection  220 , and transmits it to an external device. Audio status includes information about the audio being transmitted, e.g. song title, artist, etc.. Audio status may also include information about the audio player, e.g. playing, stopped, rewinding, etc.. Audio control includes information that controls Audio Source  100 , e.g. play, stop, rewind, fast forward, skip, pause, etc.;   (b) Audio Sink  210  receives audio data and audio status information from audio source  200  over wireless connection  220  and transmits to an external device (e.g. headphone), and receives audio control information from an external device and transmits it to Audio Source  200  over wireless connection  220 ; and   (c) wireless connection  220  between Audio Source  226  and Audio Sink  210 . Wireless connection  220  is assumed to use one channel of a wireless shared media such as radio and the wireless shared media contains multiple channels.   

     Referring to  FIG. 3 , Audio Source  200  is comprised of:
     (a) an Audio Analog-to-Digital Converter (ADC)  300  that converts the analog data received from an external device such as audio player to digital data using the sampling clock supplied by Audio Synchronizer  310 . Audio ADC  300  can be omitted if digital data is supplied directly (e.g. from a compact disc player with digital output);   (b) an Audio Synchronizer  310  that generates the audio sampling clock and the Transport Superframe Interval (TSF_Interval) such that one can be derived from the other by a known relationship;   (c) a Lossless Compression Module  320  that reduces the average bit rate of the digital audio data by detecting and eliminating redundant information in such a way that the original digital signal can be completely recovered under normal conditions;   (d) a Packet Formatter &amp; Buffer  330  that creates packets containing compressed audio data, audio status data, and any overhead necessary to support packet delineation, error detection, wireless link management, etc. The packets are buffered awaiting transmission by the radio. Packet Formatter &amp; Buffer  330  will only present a new packet to a Transmitter  340  once the previous packet has been Acknowledged by Audio Sink  210 ;   (e) a Transmitter  340  that transmits the packet presented by Packet Formatter &amp; Buffer  330  every TSF_interval unit of time. Note that since Audio Source  200  is never transmitting and receiving on the wireless shared media at the same time, there may be shared circuitry between Transmitter  340  and Receiver  350 ;   (f) a Receiver  350  that receives a packet from Audio Sink  210  after Transmitter  340  transmits its packet to Audio Sink  210 . Receiver  350  performs error detection on the packet and if it is not corrupted it is passed on to Packet De-formatter  360 . Since Audio Source  200  is never transmitting and receiving on the wireless shared media at the same time, there may be shared circuitry between Transmitter  340  and Receiver  350 ;   (g) a Packet De-formatter  360  that extracts the audio control data and acknowledgement from the received packet. Acknowledgements are forwarded to Packet Formatter &amp; Buffer  330  which uses the information to determine whether to present a new packet to Transmitter  340 ; and   (h) a Channel Quality Monitor  370  that continuously monitors the quality of the current channel, controls Dynamic TSF Interval switching, maintains a Preferred Channel Sequence, and controls Dynamic Channel Switching.   

     Referring to  FIG. 4 , Audio Sink  210  is comprised of:
     (a) a Receiver  400  that receives a packet every TSF_Interval. Receiver  400  performs error detection on the packet and if it is not corrupted it is passed on to a Packet Buffer &amp; De-formatter  410 . Since Audio Sink  210  is never transmitting and receiving on the wireless shared media at the same time, there may be shared circuitry between Transmitter  420  and Receiver  400 ;   (b) a Packet De-formatter &amp; Buffer  410  that extracts the audio data and audio status data and buffers the audio data to maintain smooth audio playback;   (c) a Transmitter  420  that transmits the packets to Audio Source  200  immediately after the packet from Audio Source  200  is received every TSF_Interval. Since Audio Sink  210  is never transmitting and receiving on the wireless shared media at the same time, there may be shared circuitry between Transmitter  420  and Receiver  400 ;   (d) a Lossless Decompression Module  430  that reconstitutes the original digital audio data from the compressed data received in the packet;   (e) an Audio Digital-to-Analog Converter (DAC)  440  that converts the digital audio data to analog audio using the sampling clock supplied by an Audio Synchronizer  450 ;   (f) an Audio Synchronizer  450  that generates the TSF_Interval and the Audio Sampling Clock from the Packet Detected signal received from Receiver  400 ;   (g) a Packet Formatter  460  that creates packets containing Audio Control Data and Acknowledgements; and   (h) a Channel Quality Monitor  470  that continuously monitors the quality of the current channel, controls Dynamic TSF Interval switching, maintains a Preferred Channel Sequence, and controls Dynamic Channel Switching.
 
Wireless Shared Media Connection
   

       FIG. 5  depicts the wireless shared media duty cycle. In the solution of the present invention a time division duplex transmission scheme is used to control access to the wireless shared media. This means that the same wireless shared media is used to send packets from Audio Source  200  to Audio Sink  210  and from Audio Sink  210  to Audio Source  200 , but not at the same time. Thus packets ‘ping-pong’ back and forth between the two ends. 
     Transport SuperFrame Interval (TSF_Interval) (hereinafter “TSF  500 ”) is a period of time of defined length that repeats continuously while Audio Source  100  is connected to Audio Sink  210 . Within that period of time, there is time allocated for Audio Source  200  to access the wireless shared media to send an audio source packet  510  to Audio Sink  210 , and for Audio Sink  210  to access the wireless shared media to send an audio sink packet  520  to Audio Source  200 . Since the direction of transmission changes between these two periods, there is time allocated to allow the radios to switch between transmit mode and receive mode and vice-versa. Also, since TSF  500  may contain more time than is required for the transmission of all data, there may also be an idle period  530  allocated where there is no radio transmission. 
     The start of audio source packet  510  is triggered by the start of TSF  500 . This packet is always transmitted, regardless of whether there is audio data in it or not. It is also a variable length packet with a defined maximum size. Audio Sink  210  transmits its audio sink packet  520  beginning immediately after the end of audio source packet  510  (after allowing time for the radios to switch direction). Audio sink packet  520  is always transmitted and is also variable length with a defined maximum, but is typically much smaller than audio source packet  510 . 
     An audio synchronization function performed by Audio Synchronizer  310  in Audio Source  200  controls the length of TSF  500 . This information is communicated to Audio Sink  210  in audio source packet  510  overhead. The length of TSF  500  must take in account many competing factors as listed in the following table. The objective is to maximize the capacity available to audio, while minimizing the audio rate by using lossless compression: 
     
       
         
               
             
               
               
               
               
             
               
               
               
               
             
           
               
                 TABLE 1 
               
             
             
               
                   
               
               
                 System Parameters 
               
             
          
           
               
                 Parameter 
                 Description 
                 Value 
                 Units 
               
               
                   
               
             
          
           
               
                 Peak_Radio_Bit_Rate 
                 The rate that bits are 
                 2.4 
                 Mb/s 
               
               
                   
                 transmitted over the radio. 
               
               
                 Max_Packet_Size 
                 The maximum allowable 
                 1000.0 
                 bytes 
               
               
                   
                 packet size. 
               
               
                 Audio_Sampling_Clock 
                 The sampling clock used in 
                 44.1 
                 KHz 
               
               
                   
                 the analog to digital 
               
               
                   
                 conversion of audio. 
               
               
                 Ratio 
                 The ratio of the audio 
                 176 
               
               
                   
                 sampling clock to the TSF 
               
               
                   
                 frequency. 
               
               
                 TSF_Clock 
                 The TSF frequency 
                 250.6 
                 Hz 
               
               
                   
                 (1/TSF_Interval) 
               
               
                 TSF_Interval 
                 The length of the TSF in 
                 4.0 
                 msec 
               
               
                   
                 time. 
               
               
                 TSF_Bytes 
                 The length of the TSF in 
                 1182.3 
                 bytes 
               
               
                   
                 bytes. 
               
               
                 TSF_OH 
                 The amount of time during a 
                 200.0 
                 usec 
               
               
                   
                 TSF where data cannot be 
               
               
                   
                 transmitted. 
               
               
                 TSF_OH_Bytes 
                 TSF_OH in bytes. 
                 59.3 
                 bytes 
               
               
                 Packet_OH_Bytes 
                 The average amount of 
                 27.0 
                 bytes 
               
               
                   
                 packet overhead. 
               
               
                 Audio_Rate 
                 The audio data rate. 
                 1.4 
                 Mb/s 
               
               
                 Compression_Ratio 
                 The average compression 
                 0.7 
               
               
                   
                 ratio. 
               
               
                 Audio_Sync_Packet_Size 
                 The average size of the 
                 27.0 
                 bytes 
               
               
                   
                 packet transmitted from 
               
               
                   
                 audio sink to audio source. 
               
               
                 Max_Audio_Source_Packet_Size 
                 The maximum size of audio 
                 1000.0 
                 bytes 
               
               
                   
                 source packet taking into 
               
               
                   
                 account overhead and audio 
               
               
                   
                 sink packets. 
               
               
                 Compressed_Audio_Capacity_Bytes 
                 The capacity available to 
                 973.0 
                 bytes 
               
               
                   
                 transmit compressed audio 
               
               
                   
                 data in bytes per TSF. 
               
               
                 Compressed_Audio_Capacity 
                 The capacity available to 
                 2.0 
                 Mb/s 
               
               
                   
                 transmit compressed audio 
               
               
                   
                 data. 
               
               
                 Compressed_Audio_Rate 
                 The capacity required to 
                 1.0 
                 Mb/s 
               
               
                   
                 transmit the compressed 
               
               
                   
                 audio. 
               
               
                 Spare_Capacity 
                 The capacity available for 
                 1.0 
                 Mb/s 
               
               
                   
                 retransmission of corrupted 
               
               
                   
                 packets. 
               
               
                   
               
             
          
         
       
     
     In the solution of the present invention, a radio interface that transmits and receives at a bit rate of approximately 2.4 Mb/s is used. TSF  500  is 4 msec. This is derived from the audio sampling clock (44.1 KHz) by dividing by 176. At 2.4 Mb/s, approximately 1183 bytes can be transmitted in 4 msec. However, of this 4 msec, 100 usec is reserved to allow for the radio to turn-on at tie start of TSF  500 , and 100 usec is reserved to allow for the radio to switch directions (Rx&lt;−&gt;Tx) This reduces the number of bytes that can be transmitted during a specified TSF  500  by about 59 bytes to 1124 bytes. Audio sink packet  520  consumes about 27 bytes leaving 1097 for audio source packet  510 . The maximum packet size that is transmitted is 1000 bytes, therefore there will be idle time in every TSF  500 . A 1000 byte packet transmitted every TSF represents about 2 Mb/s of compressed audio capacity. A 1.4 Mb/s audio stream compressed at an average ratio of 0.7 only requires about 1 Mb/s. Therefore, there is twice as much capacity available for compressed audio than is required. The surplus 1 Mb/s is available for the retransmission of corrupted packets. Essentially, there is sufficient capacity to transmit every audio source packet twice. 
     There are other factors driving the selection of these parameter values as will be discussed later. The proposed method ensures that Audio Source  200  and Audio Sink  210  never try to transmit on the radio at the same time, resulting in a conflict and loss of data. It also provides the timing coordination that ensures that one end is in receive mode while the other end is in transmit mode. The proposed method also allows a real-time trade-off between interference robustness and power consumption by adjusting the length of TSF  500  depending on how much interference is present. Finally, the proposed method allows the start of TSF  500  to be used as a frequency reference sent from Audio Source  200  to Audio Sink  210 . This can be used to synchronize timing at Audio Sink  210 . 
     Acknowledged Packet Transmission with Retransmission 
     As those skilled in the art will appreciate, acknowledged packet transmission methods allow a sender to get explicit acknowledgement that each packet it sent was correctly received by the receiver. If an acknowledgement is not received (either the original packet or the acknowledgement was lost or damaged), then the sender retransmits the packet. Thus the lost or damaged information gets replaced. This method is sometimes referred to as Sender-Based Repair. In conventional acknowledged packet transmission methods, the receiver of a good packet responds to the sender with an acknowledgement packet that contains an identifier of the received packet. The identifier would typically be a data sequence number (DSN) that allows the sender to have several packets in transit without waiting for the acknowledgement of each one before transmitting the next one. This is useful in multi-mode networking solutions where the ability to have several packets in transit at the same time is important for achieving reasonable throughput. 
     In the solution of the present invention there is no explicit acknowledgement packet type. The fact that packets are ‘ping-ponging’ back and forth is exploited by making each packet serve as both a vehicle for sending data as well as an acknowledgement for the last good packet received. Referring to  FIG. 6 , each packet contains a DSN in the packet overhead. When one end transmits a packet with DSN=x, it expects to sees a DSN=x+1 in the next packet received from the other end. If it does, then it will transmit a new packet with DSN=x+2. If the original transmitted packet got lost or corrupted, then the next packet received from the other end, if there is one, will have DSN=x−1, in which case the DSN=x packet will be re-transmitted. If the packet from the other end gets lost or corrupted, then, again the DSN=x packet will be re-transmitted. This will continue until the DSN=x+1 packet is received, or a time-out occurs that declares the radio link to be bad. This may trigger a switch to another radio channel. 
     The implementation of acknowledged packet transmission with retransmission requires a buffer to hold the packet that was transmitted so that it can be retransmitted if it is not acknowledged. In addition, a wireless audio application where there is a continuous stream of audio must buffer the audio while the current packet is being transmitted. When the wireless shared media is poor such that frequent retransmissions are required, audio data can build up in the buffer awaiting the time when the wireless shared media clears up so that it can be transmitted. 
     Referring to  FIG. 7 , compressed digital audio is stored in an Audio Buffer  330 A in Audio Source  200 . When Audio Source  200  prepares a packet to be transmitted in the next TSF  500 , it extracts audio data from the Audio Buffer  330 A, combines it with audio status data and other packet overhead in Packet Formatter  330 B and presents it to Transmitter  340 . Audio Source  200  will extract as much audio data as it can from Audio Buffer  330 A without exceeding the maximum allowable packet size. 
     As Audio Sink  210  receives packets from the Audio Source  200 , it extracts the audio data at Packet De-Formatter  410 A and stores it in Audio Buffer  410 B. The compressed audio data is extracted from the Audio Buffer  410 B, and decompressed. When audio data first starts to flow through this system, Audio Sink  210  will not extract any data from Audio Buffer  410 B until it is almost full. Once it starts extracting data from Audio Buffer  410 B, it cannot stop or else it will interrupt the smooth flow of audio. 
     When the wireless shared media is good, Audio Buffer  330 A will stay relatively empty and Audio Buffer  410 B will stay relatively full. When the wireless shared media is bad, Audio Buffer  330 A at the transmitter will start to fill up and Audio Buffer  410 B at the receiver will start to empty out as it continues the audio flow to an external device (e.g., headphone). 
     Dynamic TSF Interval 
     TSF  500  essentially controls the radio duty cycle (i.e. the amount of time the radio is transmitting versus receiving versus off) and therefore also affects the average power consumption of Audio Source  200  and Audio Sink  210 . The instantaneous power consumption is defined by the peak consumption of the radio during receive, transmit and idle. The average power consumption is defined by the amount of time the radio is in receive mode, transmit mode and idle mode over the period of TSF  500 . Average power consumption is important because it is the main factor affecting battery life. 
     Since audio source packet  510  is always transmitted at the beginning of TSF  500 , a shorter TSF  500  means that audio source packets  510  are transmitted more frequently but the packet length may be constrained by the size of TSF  500 . Conversely, a longer TSF  500  means that audio source packets  510  are transmitted less frequently but they can be longer packets up to the defined maximum. The total capacity available to carry audio therefore increases as the size of TSF  500  increases until the maximum packet size is reached, at which point the total capacity begins to decrease as the size of TSF  500  continues to increase. As shown in  FIG. 8 , since the compressed audio data rate is relatively fixed, the spare capacity available for retransmissions increases as the size of TSF  500  increases until the maximum packet size is reached, at which point it begins to decrease. 
     Since each packet (audio source packet  510  or audio sink packet  520 ) contains a relatively fixed amount of overhead regardless of packet length, longer packets result in less capacity being used up by overhead transmission, and power consumption is reduced. Therefore, power consumption is reduced by adjusting to a longer TSF  500 . This is used to reduce power consumption when retransmission bandwidth is not required because the wireless shared media is good. It is also used when audio playback is inactive (since no capacity is required for audio) to reduce power consumption to an absolute minimum while maintaining the radio link. 
     Wireless shared media is imperfect, and data errors can occur due to a variety of causes. In radio communications, if the distance between transmitter and receiver is too long, there will be insufficient power arriving at the receiver for error-free reception. Multi-path effects may cause multiple copies of the signal arriving at the receiver with slightly different propagation delays, resulting in destructive interference and bit or burst errors. If other radios (e.g. WLAN, Bluetooth, cordless phones, microwave ovens) operating in the same frequency spectrum are within range, interference from them will also cause bit and burst errors. 
     Although a longer TSF results in longer packets and lower power consumption, the longer packets can also be more sensitive to wireless shared media errors. Arguably, the probability of interference hitting a small packet transmitted more frequently is about the same as the probability of hitting a large packet transmitted less frequently (this is arguable because more overhead is transmitted with the small packet and the impact of hitting overhead is the same as the impact of hitting the data payload). However, if the errors are predominantly bit errors, or small bursts (small relative to the packet size), then the retransmission of long packets, results in the retransmission of more ‘good’ information, whereas the retransmission of short packets has less ‘good’ information retransmitted and therefore requires less time and bandwidth to perform the retransmission. Therefore, a shorter TSF  500  that results in shorter packets sizes means less retransmission bandwidth is required. However, a longer TSF  500  results in more retransmission bandwidth being available, up to a point. Therefore, there is an optimum length of TSF  500  that balances the need for and availability of retransmission bandwidth. 
       FIG. 10  shows how interference robustness (the difference between the need for retransmission bandwidth and the availability of retransmission bandwidth) varies with the size of TSF  500 . At lower error rates, there are a wide range of sizes of TSF  500  that will satisfy the need for retransmission bandwidth. However, as the error rate increases, only smaller sizes of TSF  500  will satisfy the need. It is also important to note that the power consumption is also higher when using short packets. Therefore, controlling TSF  500  allows the radio to adjust to present conditions, using a longer TSF  500  when the wireless shared media is good (or audio bandwidth is not required) to optimize power consumption, and using a shorter TSF when the wireless shared media is poor to optimize error-correction. 
     In short, dynamic TSF control is used to address the trade-off between power consumption and interference robustness. When Channel Quality Monitor  370  associated with Audio Source  200  determines that the channel is deteriorating, it will instruct Audio Synchronizer  310  to switch to a shorter TSF  500 . The new TSF  500  will be communicated to Audio Sink  210  in the packet overhead. Upon reception of a packet with the new TSF  500 , Audio Sink  210  will also switch to the shorter TSF  500 . 
     Audio Synchronization Using TSF Interval 
     Digital audio data must be converted back to an analog signal before the amplification required to drive speakers. The digital to analog conversion requires a clock that must have low jitter and be exactly synchronous with respect to the audio sampling clock that was originally used to convert the analog audio to digital. In the solution of the present invention, the Audio Source synchronizes the TSF to the local audio sampling clock. Thus, at the audio sink the arrival of the first bit of each audio source packet is a frequency reference that is synchronous to that audio sampling clock. The audio sink uses this reference in a phase or frequency locked loop to recreate the sampling clock. 
     In the solution of the present invention, Audio Source  200  synchronizes TSF  500  to the local audio sampling clock generated by Audio Synchronizer  310 . Thus, at Audio Sink  210  the arrival of the first bit of each audio source packet  510  is a frequency reference that is synchronous to that audio sampling clock. Audio Sink  210  uses this reference in a phase or frequency locked loop to recreate the sampling clock. 
     Lossless Compression 
     In general, digital audio data that is received at Audio Sink  210  with no data errors or loss will have the same audio quality as it had at Audio Source  200 . In other words, the audio quality will not be limited by the radio transmission. A single bit error is audible. An error of one of the Least Significant Bits (LSBs) of an audio sample is less audible than an error of one of the Most Significant Bits (MSBs). An LSB error may sound like a quiet ‘click’ to a user whereas an MSB error may sound like a loud ‘pop’. Burst errors will also sound like loud ‘pop’s. Frequent bit errors may sound like radio ‘static’. 
     Some audio transmission methods use compression to reduce the amount of audio data that needs to be transmitted and thereby reduces the capacity required by the transmission medium. The impact of data transmission errors is affected by the use of compression. As those skilled in the art will appreciate, compression methods can be divided into two groups (lossy compression and lossless compression):
     (a) Lossy Compression is based on a psychoacoustic model of the human hearing mechanism. The audio signal is broken down into frequency components and those components that are less audible to the human ear are eliminated. Such methods can result in very high compression ratios: as much as 20:1. However, audio quality is affected because audio information is lost and in general, the greater the compression ratio, the greater the impact. Such compression methods also tend to be computationally complex and require significant power consumption to perform in real-time. MP3 compression is an example of a Lossy Compression method; and   (b) Lossless Compression exploits inherent redundancy in audio information, transmitting base samples (redundant information) infrequently, and for the remaining samples transmitting only the difference from the base. Such methods can achieve compression ratios of up to 3:1 but the compression ratio is dependant on the type of audio (some types of audio have more redundancy than others). However, no information is lost; the original audio data can be completely reconstructed such that audio quality is not affected.   

     In general, the higher the compression ratio, the greater the impact of errors since more information is contained in each bit. With lossy compression, errors cause frequency distortion resulting in ‘echoes’ and ‘warbles’. With lossless compression, base samples comprise about 20% of the transmitted capacity and the remaining 80% essentially represents the LSBs of the audio samples. Thus bit errors on losslessly compressed audio sound mostly like quiet ‘clicks’. 
     The solution of the present invention uses lossless compression to maximize audio quality while reducing the required audio capacity. This has two benefits. It reduces the radio duty cycle when the wireless shared media is good, thus reducing power consumption. And it makes more capacity available for retransmission when the wireless shared media is poor. 
     Dynamic Channel Selection and Switching 
     In order to establish a wireless connection across the multi-channel shared media, Audio Source  200  and Audio Sink  210  must be using the same channel. Ideally, this channel is also the best available channel, in terms of its quality, or ability to support error-free audio transmission. 
     The Channel Quality Monitor  370  associated with Audio Source  200  maintains a Preferred Channel Sequence (PCS), which is a list of the channels in the shared media in order of their desirability. The list is biased by knowledge of channels that are more likely to experience poor quality in the future. For example, radio channels that overlap spectrum used by known interferers would be biased low on the PCS. Prior to establishing a connection, Audio Source  200  scans the available channels looking for signal energy. High energy is interpreted to be an occupied channel and therefore undesirable and again the channel will end up low on the PCS. After Audio Source  200  has derived the PCS, it will start transmitting on the most preferred channel. Audio Source  200  will periodically re-scan the available channels after starting transmission in order to keep the PCS current. 
     Once a connection has been established with Audio Sink  210 , Audio Source  200  will send the PCS to Audio Sink  210 . Channel Quality Monitor  370  associated with Audio Source  200  and Audio Sink  210  will continuously monitor signal energy, rate of missing acknowledgements, and Audio Buffer  330 A fill to derive an assessment of the quality of the current channel. 
     If the quality drops below a pre-defined threshold for a pre-defined amount of time, the Channel Quality Monitor  370  triggers a Dynamic Channel Switch (DCS). Referring to  FIG. 12 , if Audio Source  200  is operating in channel X of 16 possible channels, and if channel X encounters interference from, for example a wireless local area network (WLAN), then Audio Source  200  will move to the next preferred channel (shown as “Y”) in the PCS and if it is still a good channel, will start transmitting there and wait for Audio Sink  210  to find it. Audio Sink  210  will also move to the next channel in the PCS and look for Audio Source  200 . While this is taking place, audio continues to accumulate in Audio Buffer  330 A associated with Audio Source  200 , and audio continues to play out of the Audio Buffer  330 A. The use of missing acknowledgements ensures that Audio Source  200  and Audio Sink  210  will decide to abandon the current channel at approximately the same time. An additional delay is applied before Audio Sink  210  switches in order to give Audio Source  200  time to start transmitting on the new channel. 
       FIG. 13  provides a decision matrix used by Channel Quality Monitor  370  while transmission is active to determine the appropriate interference avoidance action—specifically, whether to invoke dynamic TSF interval or dynamic channel selection. The channel energy on the selected channel (e.g. Y) and Audio Buffer  330 A fill are continuously monitored (Note: the rate of missing acknowledgements may also be used in place of Audio Buffer  330 A fill). If the channel energy is high but Audio Buffer  330 A fill is deteriorating (i.e. the buffer is filling up due to frequent retransmissions), it is deduced that the high channel energy is a result of interference. Initially, a shorter TSF interval is chosen to see if the deteriorating buffer condition is resolved. If not, then dynamic channel switching is triggered. If the channel energy is low but Audio Buffer  330 A fill is deteriorating, then it is deduced that interference is not the cause. A shorter TSF interval is then chosen. If this does not resolve the problem, then Audio Sink  210  is considered to be out of range of Audio Source  200  and the user is given an out of range signal. Finally, if the channel energy is low but Audio Buffer  330 A fill is stable, it is deduced that Audio Sink  210  is approaching out of range of Audio Source  200  and the user is given an out of range signal. 
     Dynamic Transmit Power 
     Depending on the link budget of the radio, there may be significant signal to noise ratio (SNR) margin available to combat interference. However, if the wireless shared media is good (i.e. based on quality measurement of the in use channel), the output power of transmitter  340  can be reduced. As shown in  FIG. 14 , as Audio Sink  210  gets closer to Audio Source  200  less output power is required. This has the effect of reducing power consumption at transmitter  340 . It also has the effect of reducing the interference caused on other radios. Thus the distance required between personal audio devices before a channel can be reused is reduced, freeing up other channels to be used by Dynamic Channel Switching to combat other sources of interference. 
     Although various exemplary embodiments of the invention have been disclosed, it should be apparent to those skilled in the art that various changes and modifications can be made which will achieve some of the advantages of the invention without departing from the true scope of the invention. For example, the following modifications are meant to be included within the scope of the invention:
         (a) Analog or digital audio data can be supplied to Audio Source  200 . If digital audio data is supplied, Audio ADC  300  is not required;   (b) Analog or digital audio data can be produced by Audio Sink  210 . If digital audio data is produced, the Audio DAC  440  is not required;   (c) The method of audio compression can be lossless, as described above, or lossy. Lossy compression allows for much greater compression ratios thereby reducing the audio bandwidth requirement. However, lossy compression reduces audio quality and it is much more complex and therefore consumes more power than lossless compression. The potential benefit of using lossy compression depends on the peak power consumption of the radio since there is a trade-off between the higher power consumption of the compression/decompression and the lower average power consumption of the radio because of the lower data rate;   (d) If the audio data rate is sufficiently low relative to the peak bit rate of the wireless connection, compression can be eliminated altogether;   (e) The present invention can be adapted to support multiple audio sinks all listening to the same audio source at the same time. However, only one audio sink can send audio control data. The additional audio sinks can only receive audio data and audio status data;   (f) The shared media could be wired;   (g) The wireless shared media could be radio, infra-red, or something equivalent; and   (h) The present invention could be adapted to support bi-directional audio transmission such as would be used between a cellular phone and wireless headset. The system configuration for this application is shown in  FIG. 15 . In this configuration, the transmit path of Audio Source  200  (i.e. components  300 ,  320 ,  330  and  340 ) is combined with the receive path of Audio Sink  210  (i.e. components  400 ,  410 ,  430  and  440 ) to provide bi-directional audio communication. Each end of the connection would have the identical configuration, however, one end would have to be assigned to be Master and the other end to be Slave with respect to the Audio Synchronization, and Dynamic Channel Selection and Switching functions.
 
In general, the present invention can be applied to any application that requires point-to-point wireless communication of streaming isochronous (i.e. transmissions that require timing coordination to be successful) digital data, including voice.
       

     As will be understood by those skilled in the art, the functionality described in the specification may be obtained using hardware or a combination of hardware and software. The software may be implemented as a series of computer readable instructions stored in a microprocessor. The computer readable instructions may be programmed in a procedural programming language (e.g. “C”) or an object oriented language (e.g. “C++”). Preferably, the components comprising Audio Source  200  and Audio Sink  210  are formed within respective integrated circuits which may be used in combination with other on-chip or off-chip components to perform the function described herein. 
     Persons skilled in the field of radio frequency and integrated circuit may now conceive of alternative structures and embodiments or variations of the above all of which are intended to fall within the scope of the invention as defined in the claims that follow.

Technology Classification (CPC): 6