• 专利附录
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    • 专利摘要:
    In-band signaling modems communicate digital data over voice channels of wireless communication networks. The encoder converts digital data into audio tones that synthesize the frequency characteristics of a human voice. Digital data is also encoded to prevent speech encoding circuitry in the communication network from degenerating synthesized audio tones representing the digital data. The output then outputs the synthesized audio tones to a voice channel of a digital wireless communication network.
    公开号:KR20030010757A
    申请号:KR1020027017575
    申请日:2001-06-22
    公开日:2003-02-05
    发明作者:단 에이. 프레스톤；조셉 디. 프레스톤；로버트 리옌덱켈；웨인 이테리；로드 엘. 프록톨；필립 알. 스미스
    申请人:에어비퀴티 인코포레이티드.；
    IPC主号:
    专利说明:

    Enhanced in-band signaling for data communications over digital wireless telecommunications networks
    [2] Cellular telephones allow one user to talk to another user without being tied to a "land line." The cell phone includes circuitry for sampling audio signals from a user's voice. These voice signals are converted into digital form using an A-D converter. These digitized voice signals are encoded by a voice coder (vocoder) and modulated on a carrier frequency that transmits voice signals over a cell network. Voice signals are sent over the wireless cellular network to other phones in the wireless cell network or to other phones in the landline phone network.
    [3] Other coders / decoders (codecs), modulators, vocoders, automatic gain controllers (AGCs), analog-to-digital converters (A / D), noise reduction Circuits, and digital-to-analog converters (D / A), are used in the cellular and landline phone networks. Such telephony components may implement other coding schemes to encode and decode speech signals.
    [4] These communication components are designed to efficiently transmit voice signals over wireless and landline voice communication channels. For example, a digital vocoder uses predictive coding techniques to represent sound beam signals. These predictive coders filter out noise (non-voice signals) while compressing and evaluating the frequency components of the voice signals before being transmitted over the voice channel.
    [5] Problems arise when a voice communication facility such as a vocoder is used to transmit digital data. Vocoders can interpret signals representing digital data, such as non-voice signals. The vocoder may completely filter or degrade such digital data signals. Therefore, digital data cannot be reliably transmitted over the same digital audio channel used to transmit voice signals.
    [6] Sometimes, a user needs to transmit both audio signals and digital data to different locations simultaneously. For example, when a cellular telephone user makes a "911" emergency help call, the user may need to send digital location data to the call center and at the same time orally explain the emergency to the attendant. It would be desirable to transmit this digital data through the cell phone without using a separate analog wireless modem.
    [7] Accordingly, there is a need to transmit digital data over voice channels of digital wireless communication networks.
    [1] TECHNICAL FIELD The present invention relates to wireless communications, and more particularly to a system for transmitting digital data over an audio channel in a digital wireless network "in-band".
    [11] 1 illustrates a wireless communication network providing in-band signaling (IBS) in accordance with the present invention.
    [12] 2 is a detailed view of a cellular telephone coupled to an IBS modem in accordance with an embodiment of the present invention.
    [13] 3 illustrates another embodiment of an IBS modem according to the present invention.
    [14] 4 is a detailed view of an IBS modem encoder.
    [15] 5 is a schematic diagram of an IBS packet.
    [16] 6 is a schematic diagram of digital data tones output from an IBS modulator.
    [17] 7 shows how digital data is degraded by an automatic gain adjuster.
    [18] 8 shows how a digital wireless network filters digital data tones.
    [19] 9 is a detailed diagram of a receiving circuit coupled to an IBS modem decoder.
    [20] 10 is a state diagram for the IBS decoder shown in FIG.
    [21] 11 is a block diagram showing a search state in the IBS decoder.
    [22] 12 is a block diagram illustrating an active state in an IBS decoder.
    [23] 13 is a block diagram illustrating a clock recovery state in an IBS decoder.
    [24] 14 is a schematic diagram of a cellular phone with an IBS modem located in a detachable battery pack.
    [25] 15 is a schematic diagram illustrating other data sources coupled to a cellular telephone via an IBS modem.
    [26] Fig. 16 is a schematic diagram showing an implementation of an IBS modem using a sound card.
    [27] 17 and 18 are schematic diagrams showing how the sound card of FIG. 16 operates.
    [28] 19 is a block diagram of a synchronization circuit for an IBS modem.
    [29] 20 is a detailed view of the synchronous circuit of FIG. 19;
    [30] 21 is a timing diagram showing how the synchronization circuit of FIG. 19 operates.
    [31] 22 is a graph showing how the synchronization circuit identifies the optimal synchronization start time.
    [32] 23 illustrates an alternative implementation of a synchronization circuit.
    [33] 24 is an encoder diagram for a multichannel IBS modem.
    [34] 25 is a decoder diagram for a multichannel IBS modem.
    [35] 26 and 27 show other channel configurations for the multichannel IBS modem shown in FIGS. 24 and 25;
    [36] Fig. 28 is an encoder diagram for a multicarrier IBS modem.
    [37] 29 is a decoder diagram for a multicarrier IBS modem.
    [8] Summary of the Invention
    [9] In-band signaling modems communicate digital data over voice channels in digital wireless communication networks. The input receives digital data. The encoder converts digital data into audio tones that synthesize the frequency characteristics of human speech. Digital data is also encoded to prevent speech encoding circuitry in the communication network from degenerating synthesized audio tones representing the digital data. The output then outputs the synthesized audio tones to a voice channel of a digital wireless communication network.
    [10] The above and other features and advantages of the present invention will become more apparent from the following detailed description of the preferred embodiments of the present invention, which is described with reference to the accompanying drawings.
    [38] Referring to FIG. 1, a wireless communication network 12 includes a cell phone 14 that receives voice signals 22 from a user 23. Voice coder (vocoder) 18 in cell phone 14 encodes voice signals 22 into encoded digital voice signals 31 which are subsequently transmitted over wireless digital radio channel 34 (cell call). The call phone 14 transmits the encoded voice signals 31 to a cellular communication site (cell site) 36 which relays the cell call to a Cellular Telecommunications Switching System (CTSS) 38.
    [39] The CTSS 38 connects the cell call to a landline phone on a PSTN network 42, such as another cell phone in the wireless cellular network 12 or a switched circuit, or a packet as a Voice Over IP (VoIP) call. Routine the cell call via the switched Internet Protocol (IP) network 46. The cell call may also be routine from the PSTN network 42 to the cellular network 12 or from the PSTN network 42 to the IP network 46 or the like. The cell call finally arrives at the phone 44 corresponding to the destination phone number initially entered at the cell phone 14.
    [40] Additional data may be inserted at any point within the cellular network 12, such as within the IP network 46 and the PSTN network 42, which has been remodulated to transmit via wired or cellular networks. Such data may be a system related to routine information, toll or tariff information and the like.
    [41] In-band signaling (IBS) modem 28 may allow cell phone 14 to transmit digital data 29 from data source 30 via radio channel 34 of cellular network 12. IBS modem 28 modulates digital data 29 into synthesized digital data tones 26. Digital data tones 26 prevent encoding components in cellular network 12, such as vocoder 28, and landline network 42, from degrading digital data. The encoding and modulation technique used in the IBS modem 28 allows digital data 29 to be transmitted through the same voice coder 18 used within the cell phone 14 to encode the voice signals 22. Any application, such as a bending machine, can be enhanced by this technique.
    [42] The synthesized tones are defined as signals representing digital data having signaling features that allow signals to be encoded and decoded by a voice codecd without losing digital data information in the signal. In one example, frequency shift keying (FSK) signals are used to generate the synthesized tones at different frequencies within the audible range of human speech.
    [43] IBS modem 28 allows voice signals 22 and digital data 29 to be transmitted over the same digital audio channel using the same cell phone circuit. This prevents the user from transmitting digital data using a separate wireless modem and allows the cell phone user to transmit and talk data during the same digital wireless call.
    [44] The present invention modulates digital data 29 into synthesized audio tones. This prevents the cell phone vocoder 18 from filtering or degenerating binary values associated with the digital data 29. The same cell phone transceiver and encoding circuitry is used to transmit and receive both voice signals and digital data. This allows the IBS modem 28 to be less or less complex and more energy efficient than a standalone wireless modem. In the same embodiments, IBS modem 28 is implemented entirely in software using only existing hardware components in cell phone 14.
    [45] One or more servers 40 are located at any of a variety of locations within wireless network 12, PSTN network 42, or IP network 46. Each server 40 includes one or more IBS modems 28 that encode, detect and decode digital data 29 transmitted and received over digital radio channel 34. The decoded digital data is processed at server 40 or routinely processed at another computer, such as computer 50.
    [46] 2, a first transmitter of an IBS modem 28 includes an IBS encoder 52 and a digital-to-analog converter (D / A) 54. The IBS encoder 52 is typically realized using a digital signal processor (DSP). Data source 30 represents any device needed to wirelessly transmit and receive digital data. For example, data source 30 may be a laptop computer, a palm computer, or a Global Positioning System (GPS) (see FIG. 15).
    [47] Data source 30 outputs digital bit stream 29 to IBS encoder 52. IBS encoder 52 converts digital data 29 into specially shaped IBS packets for transmission over a digital wireless voice channel. The IBS encoder 52 then converts the bits from the IBS packets into digital data tones that are subsequently fed to the D / A converter 54.
    [48] IBS modem 28 outputs binary values representing the amplitude and phase components of the audio tones, respectively. The D / A converter 54 converts these digital values into analog audio tones 26 that are subsequently output to the auxiliary audio port 15 on the cell phone 14. Analog audio tones 26 are then processed by cell phone 14. Analog-to-digital (A / D) converter 16 in cell phone 14 encodes the synthesized analog audio tones 26 into digital values. Vocoder 18 encodes the digital representative values of synthesized tones 26 into encoded digital data 32 and transmits the encoded data over radio channel 34 to transmit encoded digital data 32. Output to (19).
    [49] The preferred voltage of the synthesized audio tones 26 output from the D / A converter 26 is about 25 millivolt peak to peak. This voltage level is intended to prevent the audio tones 26 from saturating the voice channel circuitry in the cell phone 14.
    [50] Since digital data 29 is supplied through an existing auxiliary hands free audio port 15 within the cell phone 14, the IBS modem 28 is connected to the cell phone 14 with any data source 30. It can be installed as an after market device that can be connected. Data source 30 may transmit digital data 29 in any digital form. For example, digital data 29 may be sent via a universal serial bus (USB) interface or any other serial or parallel interface.
    [51] 3 shows another embodiment of an IBS modem 28. The IBS modem 28 of FIG. 3 is located inside the cell phone 14 and is implemented in software using existing cell phone processors or any combination of components thereof and existing cell phone components. In this embodiment, cell phone 14 may include a data port 56 that receives digital data 29 from an external data source 30. In another embodiment, digital data source 30 is in cell phone 14. For example, data source 30 may be a GPS chip that includes a GPS receiver (not shown) for receiving global location data from a GPS satellite (FIG. 14).
    [52] The IBS encoder 52 of FIG. 3 as described above is typically realized in software using a DSP and may use the same DSP used to implement the vocoder 18. D / A converter 54 outputs synthesized audio tones representing digital data 29 to internal A / D converter 16 of cell phone 14. An alternative embodiment IBS encoder 52 not only synthesizes digital data 29 into audio tones, but also quantizes it into digital frequency values. The IBS encoder 52 then outputs the quantized data 55 directly to the vocoder 18. In another embodiment of the present invention, the IBS encoder 52 is implemented entirely in software of the same DSP that executes the vocoder 18.
    [53] Vocoder 18 uses a particular encoding scheme associated with wireless communication network 12 (FIG. 1). For example, the vocoder 18 may be a VCELP encoder that converts voice signals into digital CDMA signals. A / D converter 16, D / A converter 54, and transceiver 19 are conventional cell phone components known to those skilled in the art.
    [54] It is important for the IBS encoder 52 to recognize that the digital data 29 is to be transmitted using the same cell phone circuit that transmits voice signals. The IBS encoder 52 is an arbitrary signal approximation, quantization, encoding, modulation, etc., performed by the A / D converter 16, the vocoder 18, or the transceiver 19. Prevents degenerating or filtering the bits of.
    [55] 4 is a detailed view of the IBS encoder 52 shown in FIGS. 2 and 3. Data buffer 58 stores a binary bit stream 29 from data source 30. Packetizer 60 divides the bits in buffer 58 into bytes containing the IBS packet payload. The packet formatter 62 adds packet preambles and postambles to help prevent deterioration of the IBS packet payload. IBS modulator 64 then modulates the bits into an IBS packet with two or more other frequencies 66,68 to produce digital data tones 69.
    [56] Prevent degeneration of digital data in voice channels
    [57] Cell phone speech coders increase the bandwidth of speech channels by using predictive coding techniques that attempt to describe speech signals without transmitting all information related to human speech. If any strange frequencies or tones occur in the voice channel (ie, frequencies representing digital data), those frequencies may be discarded by the voice coder 18 (FIG. 2). For example, if the amplitude of the digital data bins is greater than the normal speech signals, or if the same digital data tone is generated too long for a period of time, the speech coder 18 may output its high amplitude or extended frequency signal. You can filter. Depending on how digital data tones are encoded, the digital bits represented by such strange audio tones may be partially or wholly removed from the voice channel.
    [58] IBS encoder 52 encodes digital data 29 in a manner that voice coders do not filter or degenerate tones representing digital data. IBS encoder 52 does this by controlling the patterns of synthesized audio tones used to represent amplitudes, time periods, and binary bit values.
    [59] Referring to FIG. 5, the packet formatter 62 (FIG. 4) adds a packet preamble 73 and a header 75 to the front of the IBS packet 70. The packet preamble 73 includes a preamble pattern 72 and a sync pattern 74. Checksum 78 and packet postamble 79 are attached to the backend of IBS packet 70.
    [60] 6 shows synthesized digital data tones 69 output from IBS modulator 64 (FIG. 4). IBS modulator 64 (FIG. 4) converts the digital bits in IBS packet 70 into one or two other tones. The first tone is generated at f1 frequency and represents a binary "1" value, and the second tone is generated at f2 frequency and represented by a binary "0" value. In one embodiment, the f1 frequency is 600 Hz and the f2 frequency is 500 Hz.
    [61] The most efficient frequency range for generating tones representing binary bit values is determined to be in the range between 400 Hz and 1600 Hz. IBS modulator 64 includes sine and cosine tables used to generate digital values representing different amplitude and phase values for the f1 and f2 frequencies.
    [62] In one embodiment of the invention, the digital data is output on radio channel 34 at a baud rate of 100 bits / second. This baud rate has been found to be effective in preventing deterioration of digital audio data by a wide variety of different cellular telephone voice coders. The sinusoidal waveforms for each f1 and f2 tone are continuous for a period of 10 milliseconds beginning and ending at the zero amplitude point. Eight samples are generated for each digital data tone.
    [63] Referring to FIG. 7, an automatic gain adjuster (AGC) 80 is one encoding function used for the cell phone 14. AGC 8 may be software located within the same DSP that executes voice coder 18. AGC 8 scales instantaneous energy changes in speech signals. There are situations where no voice signals are supplied to the AGC 80 for a period of time followed by a series of audio tones 82 including the start of the IBS packet 70. AGC 80 scales the first group of tones 82 at the beginning of IBS packet 70. The AGC 80 is shown forward at zero signal level 84 after the end of the EH IBS packet 70 and will scale the tones 83 at the end of the IBS packet 70 as part of its predictive scaling design. . This scaling prevents excessive amplification of the signal and noise when momentary energy changes occur in the voice channel.
    [64] As previously shown in FIG. 6, the "1" and "0" bits of the IBS packet 70 represent tones f1 and f2, respectively. If these tones are scaled by AGC 80, the digital bits represented by those frequencies may fall off during encoding. For example, the vocoder 18 can recognize scaled tones as noise and can filter them out of the audio channel. To prevent unintentional filtering of the tones representing digital data, the IBS packet 70 of FIG. 5 includes preamble bits 72 and postamble bits 79. Preamble bits 72 and postamble bits 79 do not include any digital data bits 29 from the data source, but a certain number of bits that are not needed to detect or encode the IBS packet 70. Sacrificial bit (s). The tones generated for these victim bits of the preamble and postamble may be scaled or filtered by the AGC 80 without affecting any digital data contained in the IBS packet payload 76.
    [65] The bit pattern and sync pattern 74 in the preamble 72 are specifically shaped to further prevent degradation of the packet payload 76. A random sequence and / or an alternating "1"-"0" sequence of bits is used in the preamble 72 and / or in the sync pattern 74. These alternating or random bit patterns prevent the adaptive filters of the cell phone vocoder 18 (see FIG. 2) from filtering the tones representing the remaining bits of the IBS packet 70.
    [66] Referring to FIG. 8, adaptive filters adapt around frequencies currently being transmitted over the wireless network. For example, if a long period of a f1 tone is currently being transmitted, the adaptive filter used in the cell phone may be adapted around that f1 frequency spectrum as shown by filter 86.
    [67] Another short tone at another frequency f2 can immediately follow the long period of the F1 tones. If filter 86 is too slow to adapt, the first f2 tones can hardly be filtered from the voice channel. If the filtered f2 tone represents bits in the IBS bit stream, those bits are lost.
    [68] To prevent adaptive filters in the cell phone from dropping bits, part of the preamble 73 includes a random or alternating "1"-"0" bit pattern. This pre-adjusts the adaptive filter as shown by filter 88. Preamble 73 (FIG. 5) attempts to include some of the bit sequences that occur or are likely to occur in packet payload 76. For example, IBS encoder 52 may appear to precede the bit pattern in payload 76. Encoder 52 may then place a subset of the bits of the portion of the preamble to indicate a sequence of bits in the packet payload.
    [69] This pre-adjusts the adaptive filter for the same f2 and f2 frequencies in a similar sequence and at the same time period to follow with the IBS packet payload 76. Thus, the adaptive filter substantially represents the digital data being transmitted. Try to filter fewer tones.
    [70] 9 is a block diagram of a receiving circuit 91 for receiving voice and data signals of a radio channel 34. IBS modem 28 also includes an IBS decoder 98 that detects and decodes digital data tones transmitted on radio channel 34. Receive circuit 91 is disposed in CTSS 38 (FIG. 1) that receives wireless transmissions from call sites 36 (FIG. 1). The same receiving circuit 910 is also disposed within the cell phone 14.
    [71] As described above in FIGS. 2 and 3, the decoder portion of the IBS modem 28 may be outside the cell phone 14 or may be inside the cell phone 14. Dotted line 104 shows IBS modem 28 outside the cell phone, and dotted line 106 shows IBS modem 29 inside the cell phone. IBS modems 14 may be located at any telephone location within PSTN network 42 or IP network 46 (FIG. 1). The receiving circuit 910 may be different when the IBS modem 28 is coupled to the ground line. However, IBS modem 28 operates on the same principle by transmitting and receiving synthesized tones over the voice channel of the phone line.
    [72] Signals in the radio channel 34 are received by the transceiver 90. Vocoder 92 decodes the received signals. For example, the vocoder 92 may decode signals transmitted in TDMA, CDMA, AMPS, and the like. The D / A converter 94 then converts digital voice signals into analog signals. Analog voice signals are then output from the audio speaker 17.
    [73] When the IBS modem 28 is outside of the receiving circuit 91, the A / D converter 96 converts analog signals back to digital signals. IBS decoder 98 demodulates any tones represented in digital data back into digital IBS packets. The packet disassembler 100 disassembles the packet payload from the IBS packets 70 and stores the decoded digital data in the data buffer 102.
    [74] FIG. 10 is a state diagram illustrating how the IBS decoder 98 of FIG. 9 operates. The IBS decoder 98 repeatedly samples and decodes the audio signals received from the radio channel 34. State 110 searches for tones of an audio signal representing digital data. For tones within the frequency range of digital data tones, if the Signal to Noise Ratio (SNR) is greater than a preselected value, the IBS decoder 98 goes to an active state 112. Active state 112 collects tone samples. In any time period of the active state 112, if the SNR falls below the active threshold or reaches the timeout before sufficient tone samples are collected, the IBS decoder 98 returns to the search state 110. And restart to search for digital data tones.
    [75] After many samples have been collected, the IBS decoder 98 waits for bits identifying the preamble 73 in the IBS packet 70 (FIG. 5). Once the preamble 73 is found, the IBS decoder 98 moves to the clock recovery state 114. The clock recovery state 114 is synchronized with the synchronization pattern 74 of the IBS packet 70 (FIG. 5). The IBS decoder 98 then demodulates the packet payload 76 in state 116. If the preamble 73 is not found, the IBS decoder 98 returns to the search state 110 and resumes the search for the start of the IBS packet 70.
    [76] The IBS decoder 98 demodulates all packet payloads 76 and then executes a checksum 78 with the final confirmation that the valid IBS packet 70 has been successfully demodulated. Control then returns to retrieval state 110 and the retrieval of the IBS packet 70 then begins.
    [77] 11 is a detailed view of the search state 110 of the IBS decoder 98. The search state 110 uses in band and out of band filtering. "In-band" is used in the discussion below to represent tones within the frequency range of the two tones representing the digital data binary "1" value (500 Hz) and the digital data binary "0" value (600 Hz).
    [78] The first band pass filter 110 (in band) measures the energy for the signals in the audio channel in the frequency range of about 400 Hz to about 700 Hz. The second band pass filter 120 (out of band) measures the energy of the audio channel for signals outside the 400 Hz to 700 Hz range. The signal-to-noise ratio (SNR) is calculated at block 122 between in-band energy and out-of-band energy. If tones representing digital data are present in the audio channel, the energy measured by the in-band filter 118 will be much greater than the energy measured by the out-of-band filter 120.
    [79] If the SNR is at the selected threshold wife in comparator box 124, the signals of the audio channel are determined to be real of speech signals or noise. If the SNR is higher than the threshold, the IBS decoder 98 determines tones that represent in-band digital data. When digital data is selected, IBS decoder 98 moves to active state 112 (FIG. 10) to initiate a search for the beginning of IBS packet 70.
    [80] 12 shows an active state 1120 for the IBS decoder 98. Block 130 is windowed by the search state 110 when in-band tones are searched for the audio channel. Samples for audio tones are shown at block 133 with many samples associated with a single binary bit. In one embodiment, eighty samples of the digital data tone are taken, filled with zeros, and then correlated to discrete Fourier transforms (DFTs).
    [81] The first DFT has coefficients representing 500 Hz and is adapted to the windowed data in block 134. The first DFT produces a high correlation value if the samples contain 500 Hz tone (“0” binary bit value). The second DFT represents a 600 Hz tone and is applied to the samples windowed in block 136. The second DFT produces a high correlation value if the windowed samples contain 600 Hz tones (" 1 " binary bit values). Block 138 selects a binary "0" or binary "1" bit value for the windowed data as the 500 Hz DFT or 600 Hz DFT yields the largest correlation value.
    [82] The IBS decoder 98 in decision block 140 continues to demodulate the tones until the preamble of the IBS packet 70 is detected. The IBS decoder 98 then moves to the block recovery state 114 (FIG. 13) to synchronize with the sync pattern 74 of the IBS packet 70 (FIG. 5). If more bits are needed to be demodulated before the preamble 73 can be verified, decision block 140 returns to block 132 and the next 80 samples of digital data tones are windowed and demodulated.
    [83] 13 shows a block recovery state 114 for the IBS decoder 98. After preamble 73 in IBS packet 70 is detected in active state 1120, clock recovery state 114 demodulates the next row of bits associated with sync pattern 74 (FIG. 5). The clock recovery state 114 arranges tone samples into the center of the correlation filters described in the active state 112. This improves the accuracy of the decoder when demodulating the IBS packet payload 76.
    [84] The decision block 142 looks for the sync pattern 74 in the IBS packet 70. After demodulating the next tone, if sync pattern 74 is not found, decision block 142 offsets the window used to sample sync pattern 74 by one sample in block 148. )do. The decision block 150 then rechecks the sync pattern 74. Once the sync pattern 74 is found, the decision block 144 determines the power ratio for the detected sync pattern. This power ratio represents a confidence factor of how well the demodulator is synchronized with the sync pattern. The power ratio is compared with the power ratio derived for the other window in which the sampling positions are shifted. If the power ratio is greater than the previous sampling position, the power ratio is stored as the new maximum power ratio in block 146.
    [85] If the power ratio for sync pattern 74 is less than the previously measured power ratio, the decoder in block 148 offsets the sampling window by one sample position. The power ratio is then determined for the shifted window and compared to the current maximum power ratio in decision block 144. The window is shifted until the maximum power ratio is found for the sync pattern 74. The window offset value at the maximum power ratio is used to arrange the demodulator correlation filters into the center sample of the first bit 77 (FIG. 5) in the IBS packet header 75.
    [86] IBS decoder 89 is used to demodulate the remaining 500 and 600 Hz tones where the identified window offset represents packet payload bits 76 and checksum bits 78. Demodulation state 116 correlates f1 and f2 tones with the DFT in the same manner as the active state (FIG. 12). Checksum bits 78 are then used as a final check to verify that a valid IBS packet is received and correctly decoded.
    [87] 14 is a diagram of an IBS modem 28 located in a battery pack connected to a cellular telephone 14. Handsfree audio channel pin 200 couples IBS modem 28 to voice channel 202 in cell phone 14. The switch 204 couples voice signals from the microphone 17 or digital data tones from the IBS modem 28 to the voice channel 202.
    [88] The switch 204 is controlled via a menu on the screen (not shown) of the cell phone 14 or by a button 206 outside the rear end of the battery pack 208. Switching 204 may also be controlled by one of the keys on the keyboard of cell phone 14.
    [89] The button 206 may also be used to initiate other functions provided via the IBS modem 28. For example, the GPS includes a GPS receiver 210 disposed in the battery pack 208. The GPS receiver 210 receives GPS data from the GPS satellites 212. The cell phone operator simply presses button 206 during an emergency. Pressing button 206 automatically causes GPS receiver 210 to collect GPS data from GPS satellites 212. At the same time, the switching 204 connects to the IBS modem 28 on the voice channel 202 of the cell phone 14. Thereafter, the IBS modem 28 is activated. As soon as the GPS data is collected by the IBS modem 28, the data is formatted, encoded by the IBS modem 28 in the voice channel 202 of the cell phone 14, and output.
    [90] The user 23 may press the phone number at any time after pressing the phone number by hand. After the audio channel has been established with the other endpoint, the user 23 presses the button 206. Switching 204 is connected to IBS modem 28 and IBS modem 28 is activated. GPS data (or other digital source) then sends digital data tones via the IBS modem 28 to the endpoint via the established audio channel. After the data has been successfully transmitted, the user presses an array of buttons 2060 that reconnect the switching 204 to the audio receiver 17.
    [91] 15 shows other forms of data sources that may be connected to the IBS modem 28. Any one of the palm computers 212, a GPS receiver 214, a computer 216, or the like may be coupled to the IBS modem 28. IBS modem 28 converts the bits output from the device into digital data tones that are subsequently output via radio channel 34 of the wireless network. Since data can be sent to the other endpoint via cell phone 14, no devices 212,214,216 need a separate wireless modem.
    [92] Implementation of in-band signaling modem in sound card
    [93] IBS modems can be implemented on standard computer sound cards. Referring to FIG. 16, a sound card 252 such as a Sound Blaster card, 1523 Cimarron Plaza, Stillwater, OK 74075 manufactured by Creative Labs, Inc. Included within 250. Speaker output 253 of sound card 252 outputs audio tones to a hands free port 257 on cell phone 258. The microphone input 259 on the sound card 252 is connected to the speaker output of the cell phone 258.
    [94] The computer includes a processor 254 that converts digital data into the form of an audio channel used by the sound card 252 to output the synthesized audio tones. Cell phone 258 encodes and transmits such audio tones over a voice channel of a wireless communication network. Cell site 261 receives the transmitted audio tones and advances the audio tones through PSTN network 263. Computer 262 is connected to telephone line 260 at the destination location of the phone call. Another sound card 264 and processor 266 in computer 262 demodulate the audio tones back into digital data. Digital data represented by audio tones is displayed on computer 262. Sound cards can be used for encoding, decoding or for both encoding and decoding data. Sound cards may be used in computer 250, computer 262, or both.
    [95] 16 and 17, data files, GPS data, data entered by a user by a keyboard, or any other data is shaped by the computer 250 into IBS packets in block 270. Packetization Packet shaping is disclosed in FIGS. 4 and 5. Binary bit values in IBS packets are converted in block 272 to the digital format used by sound card 252 (FIG. 16) to produce synthesized audio tones. For example, binary "1" bit values in an IBS packet are converted to a digital format representing the first f1 frequency, and binary "0" bit values are converted to a second f2 frequency tone. f1 and f2 tones are generated similar to the scheme disclosed in FIG.
    [96] The sound card in block 274 outputs analog tones representing binary bit values in a manner similar to the digital-to-analog converter 54 and IBS encoder 52 disclosed in FIG. 3. The cell phone in block 276 encodes the audio tones and transmits the encoded audio tones over a voice channel in the wireless communication network in block 278.
    [97] 16 and 18, the cellular tone call is set to the destination phone number. At block 280, the user picks up the ringing phone line or the computer 262 (FIG. 16) at the destination end of the cellular phone call is programmed to detect a ringing signal from the phone line 260. If a ring signal is detected, the user or computer 262 in block 282 generates a "hook-off" signal on telephone line 260. Sound card 264 in block 284 acts like an analog-to-digital converter by converting audio tones on telephone line 260 into digital data. Sound card 264 coupled with processor 266 (FIG. 16) decodes IBS audio tones similar to IBS decoder 98 disclosed in FIGS. 9-13. The digital representation of the detected IBS tones is displayed on the computer 262 screen in block 290.
    [98] As an example, a user wants to find a location for cell phone 258. The user instructs the computer 262 (FIG. 16) to dial the phone number for the cell phone 258. Computer 262 uses sound card 264 to transmit IBS tones that directly connect cell phone 258 to retransmit GPS location data. Computer 250 may have a GPS receiver and cell phone 258 may have a standalone GPS receiver. If the GPS receiver and IBS modem are inside cell phone 258 as shown in FIGS. 2-9, computer 250 need not be connected to cell phone 258.
    [99] GPS data is converted into IBS tones by the sound card 252 as disclosed in FIG. 17 or via an internal IBS modem as shown in FIGS. IBS tones representing GPS data are sent back to the telephone line 260 via the PSTN network 263 and the wireless communication channel. Sound card 264 in computer 262 monitors phone line 260 for IBS audio tones. When detected, the IBS tones are converted back to digital GPS data and displayed by the processor 266 to the user on the screen of the computer 262. The mapping process in computer 262 then converts GPS longitude and latitude values into state, city, and street addresses.
    [100] synchronization
    [101] 19 shows an alternative technique for demodulating and synchronizing the IBS modem in the IBS decoder 300. IBS audio tones are received at the interface 301 via a voice channel of the wireless communication network. The received tones are converted from analog to digital form by the A / D converter 302. IBS signal detector 304 detects the presence of IBS audio tones in the manner as disclosed in FIG.
    [102] An alternative synchronization technique begins with a decoder 300 that tunes IBS signals to synthesize basebands with multipliers 306 and 308. Multiplier 306 efficiently moves any IBS tones at the first and second IBS frequencies f1, f2 to DC. This first baseband signal is referred to as S A 'and the second baseband signal is referred to as S B '. Matched filter bank 310 applies matched filters to baseband signals with expected pulse shapes for two audio tones representing binary " 1 " and binary " 0 " values. The S A signal output from the matched filter bank 310 represents a binary one value and S B represents a binary zero value. The matched filter bank may also add a filtering step to calculate for known characteristics of the wireless communication channel that may be present in the S A or S B signals.
    [103] The matched filter is selected to match the pulse shape applied to the modulator. The pulse shape is selected for the best exchange between signaling bandwidth, bit rate and intersymbol interference. The pulse shape filter is applied to the integrated pulses of the numerical oscillator of the modulator.
    [104] The IBS synchronizer 312 arranges the modulators in a sync pattern attached to the front of the IBS packet. Segments 316 of samples from S A and S B are input to synchronous demodulator 314 along a sample start time T B. Demodulator 314 outputs power value 320 to IBS synchronizer 312 indicating how closely the demodulator is synchronized with the start bit of the sync pattern. The IBS synchronizer 312 uses the power values 320 for each sample start time T B to determine the optimal synchronization start time * T B to demodulate the remaining bits in the IBS packet. IBS packet modulator 322 then uses the optimal start time (* T B ) to demodulate binary bit values from the S A and S B signals.
    [105] 20 shows a more detailed description of the IBS packet demodulator 322 and synchronous demodulator 314 of FIG. First integrator 324 integrates a first segment of samples for the S A signal. The integrator integrates N samples representing a period T of one IBS bit (board time) starting at sample start time T B. Rectifier 326 supplies the magnitude of the integral value to adder 322. In the same way, integrator 328 integrates segments of samples for signal S B starting at sample start time TB. Rectifier 330 supplies the size of the integrated segment of the S B signal to adder 322. The output of adder 322 is power signal 320 which is fed back to synchronizer 312. IBS packet demodulator 322 (FIG. 19) also includes a comparator 334 that generates a binary one or binary zero value depending on the magnitudes of the S A and S B signals.
    [106] For further explanation, FIG. 21 shows an indication of the signals S A and S B output from the matched filter bank 310. Many samples 336 of the S A and S B signals represent the bit period T of one IBS tone. In the example shown in Fig. 21, five samples are taken for each bit period T. Sample start time T B is shifted to one sample for each integration. The start sample for the first integral starts at sample start time T b1 . As shown in FIG. 21, the sample start time T b1 is not arranged with an S A signal representing a binary “1” value or an S B representing a binary “0” value. The synchronous demodulator 314 of FIG. 20 generates a power output value of 0.0 for T b1 .
    [107] When sample start time T B2 is used, demodulator 314 generates an output value of -2.0. The sample start time T B3 represents the sample with the best synchronization with the start of the "0" tone of the signal S B. At the synchronous start time TB3, the output power is -3. As the sample start times T B4 and T B5 move further away from the best position, the magnitude of the output power is reduced. 22 shows the magnitude of the power distribution for different sample start times. The maximum power magnitude is identified at the sample start time T B3 . Thus, the optimal sample start time T b3 is used by the IBS synchronizer 312 (FIG. 19).
    [108] 20 and 21, the first sampling segment 338 starting at sample time Tb3 produces an output value of −3 from the adder 332 of FIG. 20. Comparator 334 of FIG. 20 generates a binary " 0 " for any adder value less than zero. The output of adder 332 for the second segment of sample values 340 produces an output value of +3. Because the output value for the second sample segment is greater than zero, the comparator 3340 generates a binary "1" value. IBS packet demodulator 322 (FIG. 19) continues to decode tones within S A and S B signals for the remainder of the IBS bit stream.
    [109] FIG. 23 illustrates changes to the synchronous design disclosed in FIGS. 19 to 22. IBS tones are detected at block 341. IBS tones are shifted to baseband by multipliers 342 for both audio tone frequency f A representing binary bit " 1 " and audio tone f B representing binary bit " 0 ". Baseband shift is performed for each individual sample T (x) of the f A and f B signals.
    [110] Instead of summing the entire baud of samples, a running running sum of the final board value is taken using the new sample T (x) of block 344. For example, at a sample rate of 20 samples per bit, the 21st sample T (N + 1) is deleted from the running sum, and then the sample Tx is added to the running sum. The magnitudes of the two running summations for tone A and tone B are each taken at blocks 345 and compared by comparator 346. The binary "1" or binary "0" value is output from comparator 346 as the A tone or B tone samples have the largest magnitude value. Binary bit values output from comparator 346 are correlated with a known sync pattern in correlation block 347. The selected sample start time (* T B ) is identified as the last sample that produces the largest correlation value with the sync pattern. The remaining bits in the IBS packet are then demodulated according to the selected sample start time (* T B ).
    [111] Multichannel In-band Signaling Modem
    [112] FIG. 24 illustrates an encoder 350 of a multichannel inband signaling (MIBS) modem. Data source 351 generates a binary bitstream. MIBS encoder 350 creates multiple inband signaling channels within the same voice channel. Data buffer 352 stores the binary bit stream from data source 351. The packet assembler 353 combines the bits in the buffer 352 into a packet payload and adds preambles and postambles to the packet payload to form IBS packets, as described above in FIG. 4.
    [113] Encoder 350 includes two modulators 356 and 362, each generating different audio tones representing bits in the IBS packets. Modulator 356 modulates binary "1" values using f1 frequency 360 and modulates binary "0" values using f2 frequency 358. Modulator 362 uses f3 frequency 364 to modulate other bits in IBS packets having binary " 1 " values, and modulates binary " 0 " values using f4 frequency 366. The f1 and f2 tones output from the modulator 356 are referred to as a first inband signaling channel and the f3 and f4 tones second IBS channel output from the modulator 362. The tones output from the two modulators 356 and 362 are combined together by the adder 368 and then output to the D / A converter 370 and the other cell phone circuit 14 (FIG. 2). The cell phone circuit 14 encodes and transmits the tones in the two IBS channels over the audio channel of the cellular telephone network.
    [114] Each of the individual modulators 356, 366 is similar in operation to the IBS modulator 64 shown in FIG. Any number of IBS channels may be created at the IBS modem 24. For example, a third IBS channel may be provided by adding a third IBS modulator that modulates the bits for the third portion of the IBS packets into tones using frequencies f5, f6. The output of the third IBS modulator will be supplied to the adder 368. However, for simplicity, only two channel IBS modems with two corresponding IBS modulators 356, 362 are shown in FIG.
    [115] IBS channel controller 354 controls how composite IBS channels are utilized by transmitting and receiving IBS modems. For example, the first IBS channel may be used only by the first IBS modem for transmitting IBS packets, and the second IBS channel may be used only by the first IBS modem for receiving IBS packets. The second IBS modem on the other end of the transmission then uses the second IBS channel for transmission and the first IBS channel for reception. IBS channel controller 354 adds control bits to the IBS packets that negotiate the use of composite IBS channels between two communication IBS modems. Other configurations for the IBS modems are described in more detail below in FIGS. 26 and 27. The controller 354 also controls which portions of the IBS packets are modulated by the modulators 356 and 362. For example, modulators can modulate all other IBS packets, or each modulator can modulate different portions of the same IBS packets.
    [116] 25 shows a decoder 375 of a MIBS modem. Audio tones from the audio channel are decoded by the receiving circuit 372 and supplied to the A / D converter 374. The first filter 376 filters signals outside the frequency range of the two tones in the first IBS channel, and the second filter 378 filters the signals outside the frequency range of the two tones in the second IBS channel. The frequency range of the filter 376 is from f1-Δf to f2 + Δf, and the frequency range of the filter 378 is from f3-Δf to f4-Δf. Filters 376 and 378 are shown before decoders 380 and 382, respectively. However, the filters 376 and 378 may be implemented with the same DSP anywhere in the decoding process.
    [117] The first IBS channel decoder 380 detects the two tones in the first IBS channel and demodulates them into binary bit values, and the second IBS channel decoder 382 detects the two tones in the second IBS channel to binary numbers Demodulate by Decoders 380 and 382 detect, synchronize and demodulate IBS tones in the manner as already disclosed in decoder 300 of FIG. 19 or decoder 98 of FIG. The packet assembler 386 combines the bits output from the two decoders 380 and 382 into IBS packets which are subsequently output to the data buffer 388.
    [118] The IBS channel controller 384 in the receiving IBS modem synchronizes the two decoders 380 and 382 and determines which parts or which IBS packets to demodulate. The controller 384 also implements a communication protocol with a transmitting IBS modem that negotiates which IBS modems to transmit over which IBS channels and which IBS modems will receive IBS packets.
    [119] Decoder 380 and filter 376 for the first IBS channel and decoder 382 and filter 378 for the second IBS channel may be implemented in software with the same DSP. Alternatively, one DSP may be used for each channel encoder and decoder in each MIBS modem.
    [120] It is desirable for the "MIBS" modem for frequencies f1 and f2 to be spaced away from frequencies f2 and f3. One advantage of MIBS is interference mitigation and the ability to adapt to changes in call phone performance beyond manufacturers by dynamically changing frequencies when performance is poor. A robust low baud rate control signal can be sent to select a new frequency when one modem detects errors.
    [121] FIG. 26 shows one possible configuration for two multi-channel in-band signaling (MIBS) modems 390 and 396. Two IBS channels 398 and 400 are transmitted from the MIBS modem 390 over the voice channel of the wireless communication network and possibly later over the landline telephone network to the MIBS modem 396. The two MIBS modems shown in FIG. 26 are in half duplex mode where one of the IBS modems simultaneously transmits IBS packets on both the first IBS channel 398 and the second IBS channel 400. Operate.
    [122] After the first IBS modem 390 completes the transmission of IBS packets 392 on two IBS channels, the second IBS modem 396 reconnects to the modem 390 via two IBS channels 398,400. Allow start of transmission 394. MIBS modem 390 sends information in one of the IBS packets starting at MIBS modem 396 where transmission 392 is completed.
    [123] FIG. 27 shows an alternative configuration in which the first IBS channel 398 is dedicated to transmitting IBS packets from the MIBS modem 390 and the second IBS channel 400 is dedicated to transmitting packets from the MIBS modem 396. . Thus, both MIBS modems 390 and 396 can transmit and receive packets at the same time. This full duplex configuration can provide faster communication for certain types of IBS transmissions.
    [124] MIBS modem 390 may transmit different portions of the same IBS packets over two IBS channels 398,400, or may alternatively transmit other IBS packets over two IBS channels. In other configurations, one IBS channel may be used to transmit IBS packets, and a second IBS channel may be used exclusively for signaling and protocol communications between two MIBS modems. In other alternative structures, some bits from the same IBS packets are inserted into two IBS channels, or the same IBS packets are transmitted on both IBS channels for redundancy. The information in the two IBS channels can be reconstructed depending on the application associated with the IBS packet data.
    [125] Requests to reconfigure IBS channels may be encoded in the IBS packet header. For example, IBS channel controller 354 (FIG. 24) in IMBS modem 390 may send an IBS packet to MIBS modem 396 that includes a reconfiguration request in IBS packet preamble 73 (FIG. 5). . The reconfiguration request from the modem 390 may request both the first IBS channel 398 and the second IBS channel 400, and then acknowledge messages to the modem 390 at a lower baud rate. Request the MIBS modem 396 to assign a third IBS channel 401. The MIBS modem 390 then waits for the approval of the configuration request from the modem 396.
    [126] IBS channel controller 384 (FIG. 25) in MIBS modem 396 reads the reconfiguration request in the IBS packet preamble. The controller 384 then outputs the acknowledgment back through the encoder of the MIBS modem 396. The encoder formats the acknowledgment in the preamble of the response IBS packet that is subsequently modulated and transmitted back to the MIBS modem 390 over one or more currently allocated IBS channels. The controller in modem 396 then receives the IBS packets on the first and second IBS channels 398,400 and transmits the packets on a third low baud rate channel 401. Reconfigure the encoder.
    [127] When an acknowledgment from the modem 396 is received at the modem 390, the controller transmits on the first and second IBS channels and the encoder and decoder in the modem 390 to receive from the third channel at the lower baud rate. Is connected to. The two modems 390, 396 then transmit and receive IBS packets according to the new channel configuration.
    [128] Multicarrier In-band Signaling Modem
    [129] FIG. 28 illustrates a multicarrier inband signaling modem according to another aspect of the present invention. The multi-channel IBS modem disclosed in FIGS. 24-27 generates two different audio tones, where one tone represents a binary "1" value and the second tone represents a binary "0" value. The two tones are generated in a sequential tone stream over time to represent a binary bit stream.
    [130] The multicarrier IBS modem of FIG. 28 generates composite audio tones at the same time, where each tone represents a different bit position in four bit positions of the IBS packet. The particular audio tone associated with one of the four bit positions represents a binary "1" value (or alternatively a binary "0" value). If no audio tone is generated for a particular bit time (board), the IBS decoder assumes a binary value associated with the bit position being "0".
    [131] Referring to FIG. 28, a stream of bits is input to a data buffer 402 for transmission over an audio channel of a wireless communication network. The packet formatter 404 formats these bits into an IBS packet. The first portion of one of the IBS packets includes bits "1010". The packet formatter 404 outputs each one of the four bits to the other of the four modulators 406, 408, 410, and 412. The first bit "1" of the four bit sequences is called bit B1, the second bit "0" is called bit B2, and the third bit "1" of the four bit sequences is called B3, and the fourth bit "0" is referred to as bit B4.
    [132] Modulator 406 receives bit B1, modulator 408 receives bit B2, modulator 410 receives bit B3, and modulator 412 receives bit B4. Because bit B1 is a binary " 1 " value, modulator 406 generates a tone at frequency f1 during the first board period. The modulator 408 does not generate f2 tones during the first board period because bit B2 is a binary " 0 " value. Thus, modulator 410 generates f3 tones during the first board period and modulator 412 does not generate f4 tones during the first board period. The modulators operate in essentially the same manner as the IBS modulator 64 of FIG. 4 except that the frequency tone is generated for binary " 1 " values and generates no tone for the binary " 0 " value. do.
    [133] The f1 and f3 tones are combined with each other by summer 414. The digital-to-analog converter 416 converts the digital signal into an analog signal and supplies it to the cell phone transmission circuit 418. The transmitting circuit 418 transmits audio tones over the voice channel of the cellular telephone network.
    [134] 29 shows a decoder for a multicarrier IBS modem. Receive circuit 420 receives IBS tones from a voice channel of a cellular communication network. A / D converter 422 converts the audio tones into a digital signal. Each of the four bandpass filters 424, 426, 428, 430 is centered at the frequency for each of the tones f1, f2, f3, f4. Tone representing binary bit B1 is passed to bandpass filter 424 while other tones, such as tone f3, are filtered by bandpass filter f1. Decoder 432 identifies tone f1 in a manner similar to the IBS decoder disclosed in FIGS. 11-13 only for a single tone. Since the f1 tone is detected by the decoder 432, a binary " 1 " value is generated to represent bit B1 in the four bit sequence.
    [135] Since the f2 tone is not detected by the decoder 434, a binary "0" value is generated for bit B2 in the four bit sequence. The decoder 436 generates the binary " 1 " value of bit B3 by detecting the f3 tone. Decoder 438 generates a binary " 0 " value for bit B4 because the f4 tone was not generated by the multicarrier encoder. Packet assembly 440 receives four bits B1 through B4 and places them in the appropriate IBS packet location in data buffer 442.
    [136] Once the principles disclosed and described in the present invention in the preferred embodiment are understood, it is clear that the present invention can be modified without departing from such principles in detail. The inventors claim all modifications and variations that fall within the spirit and scope of the appended claims.
    权利要求:
    Claims (33)
    [1" claim-type="Currently amended] A system for communicating digital data over a voice channel of a digital communications network, the system comprising:
    An input for receiving digital data,
    A processor for converting the digital data into audio tones,
    Encoding the audio tones in the same manner used to encode voice signals and transmitting the encoded audio tones over the same voice channel of the digital communication network used to transmit the voice signals. And an output for outputting audio tones.
    [2" claim-type="Currently amended] The method of claim 1,
    A computer for converting the digital data into a format used by a sound card for generating audio tones.
    [3" claim-type="Currently amended] The method of claim 2,
    The computer formats the digital data into packets with preambles that precondition the digital transmission circuitry to prevent degeneration of the tones representing the digital data.
    [4" claim-type="Currently amended] The method of claim 3, wherein
    The output couples the sound card to a microphone input of a cellular telephone, the cellular telephone providing digital transmission circuitry for encoding and transmitting the audio tones over the digital communications network.
    [5" claim-type="Currently amended] The method of claim 4, wherein
    The audio tones are generated by the sound card and fed to an analog-to-digital converter in a cell phone that also processes human voice signals.
    [6" claim-type="Currently amended] The method of claim 1,
    A sound card converts binary "1" bits in the digital data into a first tone having a first frequency in the human voice range, and the sound card converts binary "0" bits in the digital data into a second tone in the human voice range. Converting to a second tone having a frequency.
    [7" claim-type="Currently amended] The method of claim 6,
    Wherein the first and second frequencies are both between 400 and 1000 Hz.
    [8" claim-type="Currently amended] The method of claim 1,
    The input is coupled to a telephone line and the processor digitizes audio tones on the telephone line.
    [9" claim-type="Currently amended] The method of claim 8,
    And the computer to convert the digitized audio tones returns to binary bit values.
    [10" claim-type="Currently amended] The method of claim 1,
    A computer coupled and a sound card coupled to the computer;
    The computer sends a location request to a cellular phone, the sound card converts the location request into audio tones and transmits the audio tones over the digital communication network, after which the sound card responds to a response from the cellular phone. Monitor the phone line and digitize audio tones from a cellular telephone representing location data, and then the computer converts the digitized audio tones into digital data and displays the digital data on a computer screen.
    [11" claim-type="Currently amended] In the synchronizer,
    An input for sampling the first and second audio tones,
    A demodulator for generating a synchronization value by comparing the samples of the first audio tone and the samples of the second audio tone;
    And a synchronizer for synchronizing the demodulator by shifting a start time for samples of the first and second audio tones until the demodulator produces an optimal sync value.
    [12" claim-type="Currently amended] The method of claim 11,
    And a tuner coupled to an input that shifts the first and second audio tones to a baseband frequency.
    [13" claim-type="Currently amended] The method of claim 11,
    The demodulator,
    A first integrator for summing samples of the audio tone,
    A second integrator for summing samples of the second audio tone,
    And a summer for generating the synchronization value by comparing the output of the first integrator with the output of the second integrator.
    [14" claim-type="Currently amended] The method of claim 13,
    A first rectifier coupled between the first integrator and the summer,
    And a second rectifier coupled between the second integrator and the summer.
    [15" claim-type="Currently amended] The method of claim 13,
    And a comparator coupled to a summer for generating binary bit values.
    [16" claim-type="Currently amended] The method of claim 11,
    And the first and second audio tones are transmitted over a voice channel of a digital communication network.
    [17" claim-type="Currently amended] The method of claim 11,
    Wherein the plurality of samples of the first tone and the plurality of samples of the second tone compared by the demodulator represent an amount of time, wherein the first and second tones are generated for one bit of the digital data, Synchronizer.
    [18" claim-type="Currently amended] The method of claim 11,
    Wherein the demodulator takes a running sum of the samples and compares the running sum for the first and second tones to produce a binary " 1 " value or a binary " 0 " value.
    [19" claim-type="Currently amended] The method of claim 18,
    And a sync pattern correlator that identifies an optimal sync value by correlating the binary "1 " and binary " 0 " values with a sync pattern.
    [20" claim-type="Currently amended] In a multichannel inband signaling modem for communicating digital data through a voice channel of a communication network,
    An input for receiving digital data,
    Convert a first binary bit value in the first portion of the digital data to a first audio tone having a first frequency and convert a second binary bit value in the first portion of the digital data to a second audio tone having a second frequency A first modulator to convert,
    Convert a first binary bit value in the second portion of the digital data to a third audio tone having a third frequency and convert the second binary bit value in the second portion of the digital data to a fourth audio tone having a fourth frequency A second modulator to convert,
    And an output for outputting the audio tones over a voice channel of a digital wireless communication network.
    [21" claim-type="Currently amended] The method of claim 20,
    Visible to the first and second audio tones and reconvert any detected first audio tones to the first binary bit value and reconvert any detected second audio tones to the second binary bit value. With 1 decoder,
    A second monitoring for the third and fourth audio tones and reconverting any detected audio tones to the first binary bit value and reconverting any detected fourth audio tones to the second binary bit value A modem comprising a decoder.
    [22" claim-type="Currently amended] The method of claim 21,
    And a controller that controls when the first and second modulators generate audio tones and when the first and second decoders monitor the audio tones.
    [23" claim-type="Currently amended] The method of claim 22,
    And the controller is in a configuration session with another multichannel in-band signaling modem.
    [24" claim-type="Currently amended] The method of claim 22,
    Wherein the controller controls which bits in the digital data are changed to audio tones by the first and second modulators.
    [25" claim-type="Currently amended] The method of claim 21,
    A first filter coupled to a first decoder for filtering signals outside the frequency range of the first and second audio tones;
    And a second filter coupled to a second decoder for filtering signals outside the frequency range of the third and fourth audio tones.
    [26" claim-type="Currently amended] The method of claim 20,
    Wherein the audio tones are fed to the same analog-to-digital converter in a cell phone that processes human voice signals.
    [27" claim-type="Currently amended] The method of claim 21,
    And the first and second modulators and the first and second decoders are located within a cellular telephone.
    [28" claim-type="Currently amended] The method of claim 20,
    A digital-to-analog converter that converts the audio tones into analog signals that are input to a voice input of a telephone and then encoded and transmitted by voice processing circuitry within the telephone through a voice channel of the communication network, modem.
    [29" claim-type="Currently amended] In the multicarrier inband signaling modem for communicating digital data over a voice channel of a communication network,
    An input for receiving digital data,
    Complex modulators each converting binary values into different audio tones at associated bit positions in the digital data;
    And an output for outputting the audio tones over a voice channel of a digital communication network.
    [30" claim-type="Currently amended] The method of claim 29,
    Wherein the modulators each generate a different audio tone for the first binary value and no audio tone for the second binary value.
    [31" claim-type="Currently amended] The method of claim 29,
    And a summer coupled to the outputs for all complex modulators outputting a multitone signal representing all bit values for all the relevant bit positions at the same time.
    [32" claim-type="Currently amended] The method of claim 29,
    A composite decoder, each monitoring for an associated one of the composite audio tones, generating a first binary bit value when the associated audio tone is detected, and generating a second binary bit value when no associated audio tone is detected; Including a modem.
    [33" claim-type="Currently amended] The method of claim 32,
    And bandpass filters coupled to each of the composite decoders that filter signals outside the frequency range of an associated one of the audio tones.
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    EP1297648A1|2003-04-02|
    US8068792B2|2011-11-29|
    WO2001099295A3|2002-04-04|
    JP5414139B2|2014-02-12|
    CY1109185T1|2014-07-02|
    JP2003536116A|2003-12-02|
    KR100867885B1|2008-11-10|
    CN101505160B|2012-12-05|
    引用文献:
    公开号 | 申请日 | 公开日 | 申请人 | 专利标题
    法律状态:
    2000-06-22|Priority to US09/602,593
    2000-06-22|Priority to US09/602,593
    2001-06-22|Application filed by 에어비퀴티 인코포레이티드.
    2001-06-22|Priority to PCT/US2001/020021
    2003-02-05|Publication of KR20030010757A
    2008-11-10|Application granted
    2008-11-10|Publication of KR100867885B1
    优先权:
    申请号 | 申请日 | 专利标题
    US09/602,593|US6493338B1|1997-05-19|2000-06-22|Multichannel in-band signaling for data communications over digital wireless telecommunications networks|
    US09/602,593|2000-06-22|
    PCT/US2001/020021|WO2001099295A2|2000-06-22|2001-06-22|Inband signaling over digital wireless telecommunications networks|
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