TW295753B - - Google Patents

Description

Printed by the Central Standard Falcon Bureau Employee Consumer Cooperative of the Ministry of Economic Affairs A7 _________B7___ V. Description of the invention (1) Technical field The present invention generally relates to sound compression technology, and more particularly relates to a method of using effective frequency bandwidth usage and time compression technology And device 0 Background With current technology, it is not economical to broadcast voice messages for large-scale paging systems. The air time required for voice calling exceeds that required for timbre, digital or textual calling. With current technology, voice calling services will be very expensive compared to timbre, digital value, or textual calling with lower than ideal sound quality reproduction. Another limitation that restricts the calling of voice messages is the current method of bandwidth and the bandwidth of the calling channel. In comparison, the growth of alphanumeric paging has been limited by a limited approach to a keyboard input device. This device is used to send alphanumeric messages to the broadcast by means of a personal keyboard or call to the operator center Call the terminal. Since the caller only needs to pick up the phone, dial the number to be contacted and speak the message, the voice system can overcome these entry problems. In addition, none of the current voice calling systems use Motorola's new high-speed calling protocol architecture (also known as FLEXTM). Existing voice calling systems lack many established advantages: including high storage batteries Storage ratio, multi-channel scanning capability 'mode mix (such as voice and data), reply call (allowing the caller to send the recovered signal), location finding capability, and system and frequency reuse especially in the metropolitan area , And the scope of selective retransmission of missing message parts is expanded. About the characteristics and class of time-scaled calling with sound signal -4-This paper scale is applicable to China National Standard (CNS) A4 specification (210X297mm)---1 ^ 1 If In I ^ li n . ---- I— i I ——— i I 03., -ff--(please read the precautions on the back first and then fill out this page) B7 5. Description of the invention (2) The other system of sound material is mentioned, Current time scaling methods lack the ideal combination of providing sufficient voice quality and flexibility, which allows designers to optimize their applications within known constraints. Therefore, for the "Six" anorectic communication system, it is within the known configuration to allow the most economically feasible and flexible needs of optimization, and more particularly retains many of the advantages of the Motorola flex1 ^ protocol for the application of calling. SUMMARY OF THE INVENTION-A feature is that the present invention includes a method for compressing multiple voice signals in a voice communication resource with a known frequency bandwidth in a voice communication system. The method includes the steps of placing at least one of each of the plurality of sound signals on a primary channel and compressing the time sub-frequency voice communication resources of each voice signal in each secondary channel, wherein these steps provide compressed sound signal. Printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs. Another feature of the invention is that the communication system using sound compression has at least one transmitter base station and multiple selective beer receivers. The transmitter base station includes an input device for receiving audio signals; a processing device for compressing audio signals using time-scale reduction technology and sideband modulation technology to provide signals for disposal; and a processing device for processing The quadrature amplitude modulator used for the subsequent transmission of the signal. Each of the multiple selective call receivers includes a selective call receiver module for receiving the transmitted processed signal; a single sideband demodulation technique is used to demodulate the received processed signal and the time scale expansion technique is used to Provide a processing device for the reconstructed signal; and an amplifier for amplifying the reconstructed signal into a reconstructed audio signal. Another feature of the clothing invention is: the choice for receiving compressed sound signals -5- This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (21〇; < 297 mm) Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs Printed A7 B7 V. Description of the invention (3) The sexual call receiver includes a selective call receiver module for receiving the transmitted processed signal; a single sideband demodulation technique to demodulate the received processing And time scale expansion technology to provide a processing device for the reconstructed signal, and an amplifier for amplifying the reconstructed signal into a reconstructed audio signal. Another feature of the present invention is: a broadcasting base station for transmitting selective call signals on a communication resource having a predetermined frequency bandwidth includes an input device for receiving multi-inverted audio signals; A device for sub-channels into a predetermined number of sub-channels;-Amplify the amplitude of individual audio signals and filter the amplitude compression and filter module for individual audio signals for each sub-channel; time compression for a pair of individual audio signals for each sub-channel Time compression module and a quadrature amplitude modulator for subsequent transmission of processed signals. BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram of a voice communication system according to the present invention. 2 is a block diagram of a base station transmitter according to the present invention. The stop 3 is an enlarged electrical block diagram of the base station transmitter according to the present invention. Circle 4 is an enlarged electrical block diagram of another base station transmitter according to the present invention. 5 is a block diagram of the speech processing 'coding and modulation part of the base station transmitter according to the present invention. Β FIG. 6 is a spectrum analyzer of the 6 single sideband signal transmitter according to the present invention. FIG. 7 is a diagram according to the present invention. "Expansion box of sex call receiver i paper size appropriate financial clearance (please read the precautions on the back before filling this page).

, 1T -6-A7 A7 Printed by the Employee Consumer Cooperative of the Central Falcon Bureau of the Ministry of Economy V. Description of the invention (4) Eun 0 Figure 8 shows the expansion of another selective call receiver according to the invention. 1 < £ > ^ 9 Electrical block diagram. 9 is an enlarged electrical diagram of another selective call receiver according to the present invention,

Fig. 10 is a timing diagram showing the transmission format of the outward signal protocol according to the present invention. Fig. 11 is another timing diagram showing the transmission format of the outgoing signal agreement containing details of the sound frame according to the present invention. FIG. 12 is another timing diagram showing a control block and two analog blocks according to the present invention. Figures 13-17tf show the timing diagram of several repetitions of your time scale (compression) method according to the present invention. Figures 18-22π show the timing diagram of several repetitions of the time scale (compression) method from the 8〇1 ^ 8-8 ° according to the present invention. 23-24 show repetitive timing diagrams of the wsola_sD time scaling (extended) method according to the present invention. Figure 25 shows a block diagram of the overall WSOLA-SD time scaling method according to the present invention. DETAILED DESCRIPTION OF THE PREFERRED SPECIFIC EXAMPLE Referring to FIG. 1, a communication system showing the sound compression and expansion technology of the present invention is shown in a block diagram of a selective call system 100. This system includes a class for receiving audio signals such as telephone 114 (or The input device is similar to other input devices of computers. This input device triggers a selective call based on the sound to transmit the mask. The paper standard is applicable to the Chinese National Standard (CNS) A4 specification (210X 297 mm). Order * I (please read the precautions on the back before filling in this page) A7 A7 Central Government Bureau of Economic Affairs Employee Consumer Cooperation Du Printed 5. Description of Invention (5 to selective call receiver in system 100. Via telephone 114 Each incoming selective call typically includes (a) the receiver address of at least one of the selective call receivers in the system and (b) an audio message. The selective call initiated is typically provided for formatting and queuing Transmitter base station or selective call terminal U3. The voice compression circuit 101 of the terminal 113 is used to compress the length of time of the provided voice message (discussed in the description of Figures 2, 3 and 4 below this f Detailed operation of the compression circuit 101). The sound compression circuit 101 preferably includes a processing device that uses a time scaling technique to compress the audio signal and the single sideband modulation technology street to provide a processed signal. Then selectivity The call input selectively calls the transmitter 102, where the transmitter modulates the selective call into an RF signal sent to the air via the antenna 103. This transmitter is preferably a quadrature amplitude modulation transmitter for transmitting the processed signal. The antenna 104 in the selective call receiver i 12 receives the modulated and transmitted radio frequency signal and inputs it to the received processed signal or radio frequency signal for the selective call receiver module or radio frequency receiver Module 丨 〇5, the radio frequency signal is demodulated in this module and the receiver address and compressed voice message are modulated back. Then, then this reputational voice message is provided to the analog / digital converter (A / D) 1 15. Selective call receiver [12 preferably includes a processing device that uses a single sideband demodulation technique to demodulate the received processed signal and time scale expansion techniques to provide a reconstructed signal .then The crimped voice message is provided to the voice expansion circuit 106, where the voice message is of a length that is preferably expanded to the desired value (the description of Figures 7 and 8 below will discuss this gui sound expansion circuit for the present invention 106 detailed operation). Then this audio message is supplied to an amplifier such as audio amplifier i 08, so as to amplify it to reconstruct-8-M's Zhang scale is applicable to the Chinese National Standard (CNS) A4 specification (210X297 mm (please Read the precautions on the back before filling this page) 衮., 1Τ 2 ^ 〇3 ο

3. Description of the invention (6 Cephalogram signal printed by the Consumer Cooperative of the Central Standard Falcon Bureau of the Ministry of Economic Affairs. The bitter receiver ... supplied to the decoder ~ this receiver is matched with the decoder and stored in the decoder. Any-in the receiver address user provides a simple call receiver 112 that has received a selective call-7 early sensing indications. This simple sensing refers to π that can contain acoustic signals, such as vibration -OT ^ ^ ^ Tactile bluff, or a handle like a light ..., and then a large-scale audio message is supplied from the audio amplifier 108 to the audio frequency < piano frequency speaker in the warning 1 for notification and inspection by the user. Decoding It can be included in 07-memory, in which the received voice message can be stored and can be recalled and checked by driving—or more control 1 10. In another feature of the present invention, the 'Figure k component can be explained in the same way Be part of a dictation device, voice mail system, answering machine, or editing device such as voice tracking. Remove the wireless features of system 100, including the removal of selective call transmitter 102 and radio frequency receiver 105. The reduction circuit 1 0 1 is hardwired to the sound expansion circuit via A / D 1 1 5 (as shown by the dotted line) 06. Therefore, in the voice mail, answering machine, sound tracking or dictation system, the input device 1 14 will A voice input signal such as a voice signal is supplied to the terminal 1 i 3 with a voice compression circuit 101. The voice expansion circuit 1 () 6 and the control 110 provide the listening and operating devices to voice mail, answering machine, dictation, The output π sound signal in sound tracking editing or other applicable systems. The invention clearly anticipates that the time scale technique of the invention of the application has many applications other than calling. The examples of calling disclosed here are only those applications. An example of one of them. The recharge is -'------ 丁 '~ -β (please read the precautions before filling in this page) This paper standard is applicable to China National Standard (CNS) Α4 specifications (210Χ297mm) A7 B7 printed by the Consumer Cooperative of the Central Falcon Bureau of the Ministry of Economic Affairs 5. Description of the invention (7) Now referring to FIG. 2, the block diagram of the calling transmitter 102 and the terminal 113 is shown. The terminal 113 Includes coupling to time compression module The amplitude I and ill wave module 1 6 0 of the 6〇, and the time compression module 16 0 are transferred to the selective call transmitter 10 2 which uses overhead lines or antennas 103 to transmit messages. Refer to FIG. 3 and 4. The block shown is the lower-level block diagram of the block diagram of Figure 2. Please keep in mind that the frequency bandwidth of this compressed sound paging system is highly effective and uses the quadrature amplitude (QAM) or single sideband of the voice signal ( SSB) Modulation and time scale are based on the basic concept, trying to support 6 voice 30 voice messages per 25 kHz channel. In a first specific example and referring also to FIG. 6, compressed voice channels or voice communication resources It is best to include 3 sub-channels separated by 62 50 Hz. Each secondary channel includes two sidebands and a pilot carrier. Each sideband of these two sidebands may have the same message in the first method or in the second method, the voice information may be separated on each sideband or a single message may be split between the upper and lower sidebands. The single secondary channel has a bandwidth of approximately 6250 Hz, and each sideband occupies a bandwidth of approximately 3 125 Hz. The frequency band of the sound voice is roughly 300-2 800 Hz. Alternatively, quadrature amplitude modulation may be used, where the two independent signals are transmitted directly through the I and Q components of the signal to form the sub-channel signals. The bandwidth required for transmission is the same as in QAM and SSB. Note that modules 1 50 and 1 60 in Fig. 2 can be used repeatedly with different sound signals (up to 6 times on a 25 kHz wide channel and up to M times on a 50 kHz wide channel), in order to allow Effective and simultaneous transmission (up to 6 in the example shown). They can then be summed in a summation device (not shown, but see Figure 5), and are preferably processed as a composite signal at 102. A separate signal (not shown) contains FM modulation of FLEXTM agreement (will be explained later) -10-This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (21〇X297mm)-I --- -nn--I--II-. II n 1-1- I-^ eJJ ^ i (Please read the precautions on the back before filling out this page) A7 B7 Printed by the Employee Consumer Cooperative of the Central Standard Falcon Bureau of the Ministry of Economic Affairs 5. Description of the invention (8. This may be generated in software or as a hardware FM signal exciter. In the example shown here, the incoming voice message is preferably received by the terminal Π3. This system is best Use time-scale methods or techniques to obtain the desired compression. The preferred compression technique used in the present invention requires certain special parameters of incoming messages to provide the best quality. Time-scale compression techniques preferably process speech signals to have A signal with the same bandwidth characteristics as uncompressed speech. (Once these parameters are calculated, the speech is compressed using the desired time-scale calendar-shrinking technique.) Then use a digital encoder to encode the time-scaled speech To reduce the need to allocate The number of bits of the machine. In the case of a calling system, the coded speech assigned to multiple simultaneous broadcasting station transmitters in the simultaneous broadcasting system will need to be decoded again for further processing such as amplitude compression. The amplitude compression of the voice signal (preferably byte. Compression expander) is used in the transmitter to provide protection against channel degradation. The known waveform similar basic superposition technology or WSOLA time scaling technology encodes the voice into Uncompressed voice analog signal with the same bandwidth characteristics. This WSOLA feature makes it possible to combine SSB or QAM modulation, so the overall compression obtained is the bandwidth compression ratio of multiple QAM or SSB sub-channels (in my case, there are 6 sound channels) multiplied by the time compression ratio of WSOLA (typically between 1 and 5). In the present invention, the modified version of WSOLA is used-it will be described later and treated as "WSOLA-SD". WSOLA-SD retains the compatible characteristics of WSOLA that can be combined with SSB or QAM modulation. It is best to use an ADPCM (Adaptive Differential Pulse Code Modulation Encoder) 11-the paper size is suitable China National Standard (CNS) Λ4 specification (210X297mm) (please read the precautions on the back before filling in this page) Pack · A7 A7 Printed by the Consumer Cooperative of the Central Bureau of Economics of the Ministry of Economy V. Invention description (9 = second code The data is then distributed to the transmitter. In the transmitter, the digital bits are decoded to obtain the WS0LA_SD compressed voice, and then the vibration of the voice is compressed and expanded to provide protection against channel noise. This signal is expected Hilbert conversion to obtain a single sideband signal. Alternatively, this signal is subjected to = intermodulation to obtain a QAM signal. Then a pilot carrier is added to this signal and the flawed signal is preferably inserted at a 16 Hz sampling rate and converted into an analog. This is then modulated and transmitted. The present invention can be used as a hybrid (voice or digital) single or dual communication system to send analog analog and / or digital messages to a selective call receiver unit on a forward channel (outside the base transmitter μ) and use Recognition is received with the same selective call receiver unit with the selected transmitter from the selected reverse channel (inward to the base receiver). The system of the present invention preferably uses synchronization similar to flextm (a high-speed paging agreement developed by Motorola Corporation and US Patent No. 5,282,205 and referenced here) on both the forward channel and the address channel. Frame construction. Two frame types are used: technical frame and sound frame. The control box is preferably used for addressing and to transmit digital data to the selective call receiver in the form of portable voice units (PVU's). The voice header is used to send analog voice messages to PVU, S. The length of the two frames is the same as that of the standard FLEXTM frame and the two frames start in synchronization with the standard flEXtm. These two frames are time multiplexed on a single forward channel. The frame structure of the present invention will be discussed in more detail later when referring to Figs. 10, 11 and 12. Regarding modulation, two types of modulation are most commonly used on the forward channel of the present invention: digital FM (2-level and 4-level FSK) and AM (SSB or with guideline -12. This paper standard is applicable to China Standard (CNS) A4 specification (210X 297mm) Buttons for clothes (please read the notes on the back before filling in this page) Employee's consumer cooperation of the Central Bureau of Standards of the Ministry of Economic Affairs A7 -------- _ B7 2. Description of the invention (τοί '~ One ~ wave of QAM). Digital fm modulation is used for the synchronization part of the two frame types, as well as the address and data area of the L frame. AM modulation (each sideband can be Used alone or combined in a single message) is the voice message area of the sound box. The digital FM part of the transmission supports 6400 bits per second (32⑽ baud (Baud) symbol). The signal of the AM part of the transmission supports the frequency band Limited sound (28,000 Hz) and a pair of sound signals requires 6.25 kHz. As will be shown later, this agreement subdivides a full channel into 6.25 Hz channels to take advantage of the reduced bandwidth , And each channel and Am sideband are used for independent messages. The sound system of the present invention It is better to design to operate on a 25 kHz or 50 kHz forward channel. However, other scales of spectrum are of course within the scope of the present invention. 1: 2:> The 1000 Hz forward channel supports a single control signal during the control frame, And during the message part of the sound box, it supports up to 3 am sub-channels (6 independent signals). A 50 Hz forward channel supports the operation of time-locked two FM control signals during the control box, and in the sound box Supports up to 7 AM sub-channels (14 independent signals) during the message part of course ^ Of course, the use of different scales of bandwidth and number of sub-channels and other configurations of signals are within the scope of the present invention. The examples disclosed here are only For illustrative purposes and to express the potentially broad scope of the patent scope filed here. In addition to achieving spectrum efficiency through spectrum modulation and subchannelization, in another specific example, the present invention can use a sound compression related to the narrator Technology, this technology scales speech by a factor of 1 to 5 times. Use two AM sidebands of the primary channel for different parts of the same message or different messages (alternatively, 2 Q AM component), the total compression factor of each channel is 2 to 10%. The sound quality will typically decrease with increasing time. The reduction factor is the most. -13- This paper scale applies to the Chinese national falcon ( CNS) A4 specification U10X297 mm) -------- Gen ------ Ding (, 1 '-1 (please read the precautions on the back before filling this page) A7 A7 Central Bureau of Standards, Ministry of Economic Affairs Employee consumption cooperation du printing drawing 5. Description of the invention (11) The compression technology redundantly used in the sound system of the present invention is a modified form of the known time scaling technique, as mentioned previously, it is a similar superposition technique for waveforms ( WSOLA) »WSOLA's modified form is related to the specific narrator or language used. The name" WSOLA-SD "is" WSOLA-narrator related ", which will be discussed later. When the field reverse (inward to base receiver) channel is available, the operation of the present invention is strong. The frequency division simplex mode of operation is supported by the inward operation mode. (Both of them give way to the present invention-Motorola's U.S. Patent Nos. 4,875, 〇38 and 4.882,579 describe the use of multiple poles on the inward channel and are incorporated herein by reference). In frequency division simplex, a separate ^ dedicated ^ channel (usually paired with an outward channel) is used for internal transmission. The inward rate of _to_bit / # > 'is considered in the appropriate width of 12 5 Hz. , The system of Yi invention can be operated in one of several modes depending on the availability of the reverse channel. When the reverse channel is available, this system for both addressing and voice messages =: in simultaneous playback mode. When a reverse channel is provided, the system: operation: target message mode ’so the message is only broadcast on a single transmitter or transmitter located at the earliest possible sound band 々—44 W, Oh Bian. The target message mode q ~ is characterized by simultaneous broadcast addressing to find a portable sound stand. The response to the sound unit on the channel provides the second heart: the local message transmission of this portable sound unit. Since providing = two reuse: opportunity, the operation of the target message mode is very advantageous ❼I: This: This security mode can lead to an increase in system capacity in many large systems. 1 Figure k shows the first-specific example block of the transmitter 300 according to the present invention * 14-The size of the paper is thorny) Doing clothes --- ir-(please read the precautions on the back before filling this page) Ministry of Economic Affairs A7 B7 printed by the Central Standard Falcon Bureau Employee Consumer Cooperative V. Invention description (Ί2). An analog voice signal input is nicknamed a low-pass filter 301, which strongly attenuates all frequencies that are higher than half the sampling rate of the analog-to-digital converter (80 (:) 303, which is still coupled to the filter 30i. Eight (: 3〇3 is best to convert the analog voice signal into a digital signal 'so the use of digital processing technology can complete further signal processing. Digital processing is the preferred method, but using analog technology or analog and digital technology The combination can also perform the same function. The bandpass filter 305 integrated into the ADC 303 strongly attenuates frequencies below and above its cut-off frequency. The low cut-off frequency is preferably 300 Hz to allow for important Voice frequency passes, but attenuation will interfere with the lower frequency of the carrier. The high cut-off frequency is preferably 2800 Hz to allow important voice frequencies to pass, but attenuation will interfere with higher frequencies of adjacent transmission channels. It is best to couple to the wave The automatic gain control (agC) block 307 of the device 305 equalizes the volume level of different sounds. It is best to couple the time compression block 309 to the AGC block 307 to shorten the transmission of the voice signal It takes time and essentially maintains the same signal spectrum as the output of the bandpass filter 3 〇5. This time compression method is preferably ws〇la_sd (will be explained later), but other methods can be used. Amplitude | The reduction block 3 ji and the corresponding amplitude expansion block 720 in the receiver 700 (FIG. 7) form a compression-expansion expansion device. This device is well known for increasing the apparent signal / noise ratio of the received speech. Compression expansion The ratio is preferably 2 to 1 decibel, but other ratios can be used according to the present invention. In the special case of a communication system such as a paging system, the device 301-309 may be included in a paging terminal (113 in FIG. 1) In addition, the remaining components in Figure 3 can constitute a broadcast transmitter (Figure 1 No. 2). In this case -15-This paper standard is applicable to the Chinese National Standard (CNS) A4 specification (210X 297 public advantage)- ~~ — ~~-• ^ 1 1 nn nd Kn m. M · sn ^ i 1 ^ 11- Kn 1, i side-swallow (please read the notes on back ^ before filling out this page) Central Bureau of Standards, Ministry of Economic Affairs Employee consumption cooperation Du Printed A7 __ B7 V. Description of the invention (13), typically here the calling terminal and calling transmitter There will be a digital link between them. For example, using pulse code modulation (PC Μ) technology can encode the signal after block 309, and then use PCM decoding to reduce the transmission between the calling terminal and the calling transmitter In any case, the second bandpass wave 308, which is turned to the amplitude compression block 3 1 1, strongly attenuates below 殳 above its cut-off frequency to remove the agc 307, time compression block 309, or amplitude compression Any false frequency components generated by Box 3 11. The low-stop frequency is preferably 300 Hz to allow important speech frequencies to pass, but attenuation will interfere with the lower frequencies of the carrier. The high cutoff frequency is preferably 2800 Hz to allow important voice frequencies to pass, but attenuation will interfere with higher frequencies of adjacent transmission channels. Time-compressed speech samples are preferably stored in the buffer 313 until the complete speech message has been processed. Then this allows time-compressed voice messages to be transmitted as a whole. This buffering method is best used for paging services (typically non-immediate services). Other buffering methods may be more appropriate for other applications. For example, for applications involving two-way instant conversation, the delay caused by this type of buffering may not be tolerable. ◊ If that is the case, it is more appropriate to interpolate several small segments of the conversation. For example, if the time reputation ratio is 3: 1, three real-time voice signals can be transmitted via a single channel. Three transmissions can be interleaved on this channel at 150 millisecond bursts, and the final delay will not be unsatisfactory. The time-compressed speech signal from the buffer 3 1 3 is applied to both the Hilbert conversion waver 323 and the time delay block 3 1 5, this delay block has the same delay as the Hilbert conversion filter, However, this signal is not affected. Time delay box 3 1 5 (via summing circuit 3 1 7) and Hibbert conversion waver _ -16-This paper scale is applicable to the Chinese National Standard (CNS) Μ specification (21Qx 297mm 1 I-. — ^ ϋ — ^ n 1 ^ 1 f In — _i ·-1 I-m ^^^ 1-One OJ ~ (please read the notes on the back before filling this page) A7 -------- B7_ V. DESCRIPTION OF THE INVENTION (14) ~ — The output of 323 respectively forms the in-phase (1) and quadrature (Q) components of the upper sideband (USB) single sideband (ssb) signal. This time delay and the negative (325) of the Hilbert conversion filter ) The outputs are formed separately-the in-phase (1) and stop (Q) components of the lower sideband (LSB) single sideband signal. Therefore, they can be transmitted on the upper or lower sideband as shown by the dotted line connection. When the upper sideband is used to transmit a time When compressing the voice signal, using another similar transmitter operation on the lower sideband, the lower sideband can be used to simultaneously transmit a second time compressed voice signal. Due to the use of transmission bandwidth and resistance to crosstalk, , SSB is the preferred modulation method. Double-band amplitude modulation (AM) or frequency modulation (FM) can be used, but at least it will require at least It is also possible to transmit the frequency bandwidth of the device. It is also possible to directly transmit a time-reduced voice signal directly through the I component and a second time-compressed voice signal directly through the Q component, but in this specific example when the receiver has multipath When reception occurs, this method will be lifted between the two signals. — A direct current (DC) signal is added to the I component of the signal to generate a pilot carrier. This carrier is transmitted with this signal and is used by the receiver (700) to approximate It eliminates the need for gain and phase change or fading in the transmission channel. The I and Q components of the signal are converted into analog form by digital-to-analog converters (DAC) 3 19 and 327. Printed by the Staff Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs (Please read the precautions on the back before filling in this page) Then these two signals are filtered by low-pass reconstruction filters 321 and 329 to remove the false frequency components caused by the digital / analog conversion process. Quadrature amplitude modulation The QAM modulator 333 modulates the I and Q signals on a low-power radio frequency (RF) carrier. Other modulation methods: for example, direct digital synthesis of the modulated signal will be achieved with DACs ( 319 and 327), reconstruction filters (321 and 329) and QΑλί modulator 333 for the same purpose. Finally, the linear RF power amplifier -17-the size of this paper is applicable to the Chinese National Standard (CNS) Α4 specification (210X297 mm) economy A7 '-----— B7_____ printed by the Employees ’Consumer Cooperative of the Central Standards Bureau of the Ministry of Education. 5. Description of the invention (15) ——-— 1 33 5. Amplify the modulated RF signal to the desired power level (typically% W) Or higher). The output of the RF power amplifier 335 is then coupled to the transmitter, and its artistic changes can essentially produce the same result. For example, amplitude zooming can be performed before time compression, or omitted altogether and the device will essentially perform the same function. A block diagram of a second specific example of a transmitter 400 according to the present invention is shown. In Figure 4, both the upper and lower sidebands are used to simultaneously transmit different parts of the same-time compressed signal. Transmitter 400 preferably includes coupling and configuration such as castration antialias filter 4〇4, ADC 4〇3, band-pass filter; AGC 407, time compression block 4〇9, amplitude compression block 4ιι and band-pass filtering Yi 408. Before the entire voice message has been processed and stored in the buffer 4 1 3, the operation of the transmitter of FIG. 4 is the same as that of FIG. 3. Then the time-compressed speech samples stored in the buffer 413 are divided into upper or lower sideband uploaders. The first half of the time-compressed voice message is transmitted via one sideband, and the second half of the time-compressed voice message is transmitted via the other sideband (or alternatively directly on the I and q components). The first part of the time-compressed speech signal from the buffer 413 is added to the first Hilbert transform filter 423 and the first time delay block 415. This time delay block has the same delay as the Hilbert transform filter 423 This signal will not be heard. The output of the first time delay (via summing circuit 4-7) and the first Hilbert transform filter 423 (via summing circuit 465) are in-phase (1) and positive-phase (Q) signal components, when these components are coupled to (^ Eight to Modulator and ρ input eight, will produce the upper sideband signal with only the first part of the information from the time-compressed speech sample, and the second time-compressed speech signal from the buffer 413-18-( Please read the precautions on the back first and then fill in this page j,-& this paper-like degree thorn (CNS) Α4 ^ Τ ^ Τ97 ·; ^ 7 added to the second Hilbert conversion filter 46 1 and the second time Delay block 457 'This time delay block has the same delay as the Hibbert filter 461 ~ but does not affect this signal ^ the output of the second time delay (via summing circuit 459 and = 7) and the second Hebrew The negative (463) output of the special conversion filter 461 (again via the sum% path 465) is output as in-phase (1) and quadrature-phase (1) signal components, when these components are coupled to (^ 8 from the r ^ Q input of the modulator At this time, the upper sideband signal of the second part of the information with only $ self-compressed speech samples will be generated. The carrier component (via the summing circuit 459) is added to the T components of the upper and lower sideband signals to form the composite component for transmission. The ◎ points of the upper and lower sideband signals are accumulated by 1 (via the summing circuit 465) to form the composite for transmission Q component. Will be happy to see tl pieces 415, 423, 45 7, 461 '417, 459, 463, 465, 419, 427, 421 and 429 formed to produce pre-processed! And signal component foot pre-processor, When these components are coupled to the QAM modulator 453, low-level secondary channel signals and sub-carriers with two single-sideband signals will be generated. These ~ r 边 命 ## have independent information on each sideband. The machine 400 still includes the configuration and configuration as illustrated in FIG. 3 (^ 419 and 427, reconstruction wavers 421 and 429, QAM modulator 433, and RF power amplifier 455. The remaining operations and diagrams of the transmitter in FIG. 4 The three are the same. Printed in Figures 3 and 4 of the individual consumer cooperatives of the Central Standard Falcon Bureau of the Ministry of Economic Affairs. The individual transmitters 300 and 400 are preferably only anti-alias filters, reconstruction filters, RF power amplifiers, and analog / digital conversion as appropriate. The converter and the digital / analog converter are individual hardware components. This The rest of the device should preferably be able to run on a processor (preferably a digital signal processor). Figure 7 shows a block diagram of the receiver 700, this receiver is best Issue -19-This paper standard is used in "Home CNS> A4 specifications (21 ();) < 297 meals> ~ ~ --- A7 B7 V. Description of the invention (17) Transmitter 300 of Figure 3 Simultaneous operation. A receiving antenna is coupled to the receiver module 702. The receiver module 702 includes traditional receiver components such as amplifiers, mixers, band-pass filters and intermediate frequency (IF) amplifiers (not Shows). The QAM demodulator 704 detects the I and q components of the received signal. The analog / digital converter (ADC) 706 converts the I and Q components into digital form for further processing. Digital processing is the preferred method ', but the same function can be performed by analog technology or a combination of analog and digital technology. Other demodulation methods such as sigma-dalta converter or direct digital demodulation will achieve the same purpose as QAM demodulator 704 and ADC 706. The pre-granted automatic gain control (AGC) block 708 uses the pilot carrier transmitted with the time-compressed speech signal as a phase and amplitude reference signal to substantially cancel the amplitude and phase miss effects that occur in the transmission channel. The output of the pre-authorized automatic gain control is the corrected and received components of the received signal. The corrected Q component is added to the Hibbert conversion filter 712, and the corrected Bu component is added to the time delay block 710, which has the same delay as the Hibbert conversion filter 712 but does not affect the signal . The voice signal of the second time (iv) is transmitted on the upper police, the output of the Hibbert conversion wave filter 712 is added (via the summing circuit 714) to the time delay block, the consumer consumption cooperation of the Central Falcon Bureau of the Ministry of Economic Affairs · Printed by Du. ——-(Please read the precautions on the back before filling in this page) ^ Output to generate a voice signal when replying. If the time-compressed language: ί: number is transmitted on the lower sideband, then the output of the time-delayed square is subtracted from the output L16). The output of the wave filter 712 is desirably converted to generate a time-compressed speech signal. The time-reduced quotation mark of the reply is best to be shot in the buffer 7 1 8 until the full message of-+ + 丨 70 has been received. Other buffer methods d are possible, (see discussion in Figure 3) This paper is detailed ^ -20-

Printed by the Employees ’Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs I A7 _ B7 V. Description of the invention (18) The amplitude expansion block 720 works simultaneously with the amplitude compression block 3 1 1 of FIG. 3 to perform the poor line compression expansion function. The time expansion block 722 works simultaneously with the time compression block 309 of FIG. 3 and preferably reconstructs the speech into a natural time frame for audio output via the converter 724 or other time frames suggested by other applications. A variety of applications may include the transfer of digitized sound to the computing device 726, where the receiver-to-computer interface may be a PCMCIA or RS-232 interface or any number of interfaces well known in the art. The time compression method is preferably WSOLA-SD, but as long as the complementary method is used for the transmitter and receiver, other methods can be used. Other changes in fabric can essentially produce the same result. For example, the amplitude compression can be performed after the intermittent compression, or it can be omitted altogether and the device will still perform the same function essentially. FIG. 8 shows a block diagram of a receiver 7 50 operating simultaneously with the transmitter of FIG. 400 according to the present invention. The receiver of FIG. 8 includes the antenna configured and configured as illustrated in FIG. 7, the receiver module 752 ′ QAM modulator 754, the ADC 756, the pre-authorized AGC /;> 8, time delay block 760 and Hibbert conversion Filter 762. Until the time delay block 760 and the output of the Hilbert transform filter 762, the receiver operation of FIG. 8 is the same as that of FIG. The output of the Hilbert transform filter 762 is added to the output of the time delay block 760 (via the summing circuit 764) to generate a time-honored speech signal corresponding to the first half of the voice message transmitted on the upper sideband . The output of the Hilbert transform filter 762 is subtracted (766) from the output of the time delay block 760 to produce a time-compressed speech signal corresponding to the reply of the second half of the header message transmitted on the lower sideband. These two time-compressed voice signals are stored in the upper and lower sideband buffers 768 or 769, respectively, until the complete message has been received. Then to __ -21-this paper CNS) Μ specifications (-~~-~~~~ (please read the precautions on the back before filling in this page) Binding-Binding A7 ----------- B7 _____ V. Description of the invention (a) The signal corresponding to the first half of the message and the signal corresponding to the second half of the message are sequentially added to the amplitude expansion block 77. The amplitude expansion block 770 and the amplitude plot of FIG. 4 Block 4 1 1 works at the same time to implement the expansion and contraction function. The operation of the rest of the receiver in FIG. 8 is the same as that in FIG. 7. The time expansion block 772 works simultaneously with the time compression method in FIG. 4 and it is best to reconstruct the voice into The natural time frame or other time frames suggested or required by other applications. The time compression method is preferably WSOLA-SD, but as long as the complementary method is used in the transmitter and receiver, other methods can also be used. Other configurations are essentially Can produce the same result. For example, amplitude compression can be performed after time compression, or all can be ignored and the device will still essentially perform the same function. As in the transmitter entity of FIGS. 3 and 4, many of FIGS. 7 and 8 Components can be implemented in software, including (but not limited to): AGCs , Single-sideband or QAM demodulator, summation circuit, amplitude expansion block and time expansion block. All other components are preferably implemented in hardware. If the speech processing of the present invention, the coding and modulation part should be implemented in hardware , Printed by the Employees ’Cooperative of the Central Standard Falcon Bureau of the Ministry of Economic Affairs------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------Suffix 5, the transmitter 500 of FIG. 5 will include a series of frequency-sideband exciter (571-576) groups that correspond to the frequency of individual pilot carriers (581_583). Exciter 571-576 and pilot carrier 58 1-583 correspond In the individual sound processing path. All these signals from the FM signal exciter 577 (used in the previously described synchronization 'address and digital FM modulation of the data area) will be integrated into the summing amplifier 5 7 0' The linear amplifier 580 is amplified and then transmitted. The low level output of the FM exciter 577 is also linearly combined in the summing amplifier 57. The combined output signal of the summing amplifier 57 is amplified by the linear RF power amplifier 5SO to Required power level (usually 50 watts or more -22- This paper adopts the Chinese National Standard (CNS) A4 specification (210X 297mm) A7 B7 20 V. Description of invention (. Then the output of the linear RF power amplifier 580 is coupled to the transmitting antenna. Other devices can be used in combination Sub-channel signal. For example, several digital base bands and signals obtained from the output of 417 and 465 in FIG. 4 can be translated into individual sub-carrier offset frequencies (combined in digital form), etc. The analog is converted into an analog form for modulation on the carrier frequency. Shown with reference to Fig. 9 'is another receiver unit 900 according to the present invention. The receiver 900 also includes a device that detects and decodes FM modulation control signals for the flex ™ signal protocol. Block 902 is the receiver front end and? 1 ^ rear end. A digital automatic frequency controller (DAFC.) And automatic gain controller (AGC) merge block 902. Block 906 includes radio frequency processing β 'with support chip 95. Blocks 9 1 1, 9 14 and 9 16 include output devices. Block 9U4 is a battery saver or battery economic circuit operating under the control of the processor 906. Block 850 is a linear decoder, followed by an analog-to-digital converter tile random access memory (RAM) block 868. The receiver block 902 is preferably—including the modified FM reception of DAFC and an AGC described in US Patent No. 5,239,306 (which has been assigned to the assignee of the present invention and is hereby incorporated by reference) In this case, this modified FM receiver provides an intermediate frequency (IF) output at a point after most of the receiver gain but before the FM demodulator. The same processor that controls Motorola ’s FLEXTMi compatible callers will properly handle all of the protocol functions of the present invention. These functions include — FM demodulated signal address identification and message decoding. In addition, in response to the FM modulation address (and possibly the message index codeword), the processor 906 targets analog analog digital conversion and RAM block 86.8 operations. Block 868 samples the output of the linear decoder block 850! (In-phase) and quadrature) linear tone -23 This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210X297 public --------- installed ... 1-] (Read the notes on the back first (Fill in this page again) Printed by the Employees and Consumers Cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs A7 ------B7 Printed by the Employees and Consumers Cooperative of the Central Standard Falcon Bureau of the Ministry of Economic Affairs V. Invention ~ Variable signal two: # or one of them The sample is nested in RAM directly with the help of an address meter and responds to a control signal from the processor 906. A sound can be sent out on the channel according to the SSB signal with a single sound frequency bandwidth, or as explained earlier Equally transmitted on any channel. Each 5 [and Qi & No. occupies the same RF at the same time with the second type of sideband (sSB). The width of the band. The bandwidth of the b band is at the level of 2 · 8 kHz, so if the analogy ssb reverts from the j and Q channel information, then the analog / digital converter needs a typical signal sampling rate of about 6 to 4 Hz each. The analog / digital converter samples with 8-bit accuracy (although it is best up to 10 bits ). Direct memory access speed of analog / digital converters The power is not a processor that is a direct function of the channel data rate. That is, a microprocessor with direct memory access can be used, however, the analog-to-digital conversion data can be read into the memory via this microprocessor, which will obviously require Higher speed processor. Analog / digital converter (A / D) 'Dual port RAM and address counter form block 868. A second RAM input / output (I / O) port can be serial or parallel, And operate at a sampling rate of 6 or 1 2 K in the mother second. After providing the second ram I / O, the processor can take out the sampled sound or data, process the demodulation function, and expand the reputation or format of the sound. Data. The recovered sound is reproduced by the sound processor 9 ^ 4 and the converter 916, and the formatted data can be displayed on the display 91. Refer again to FIG. 9 '. An expanded electrical block diagram is used to explain the present invention in more detail. Receiver operation of dual-mode communication receiver. The transmitted data signal modulated in FM modulation format or linear modulation format (like SSB) is intercepted by antenna 8 0 2 'This antenna consumes the data signal To receiver section 9 〇2, especially coupled to the radio Frequency (RF) amplifier 806 input. This information is -24- This paper standard is applicable to China National Standard (CNS) A4 specification (210x 297 mm) '~~~ ---- (please read the back note first (Fill in this page again)

, 1T Ministry of Economic Affairs Central Bureau of Standards and Staff Cooperative Printing DuA A7 B7 V. Invention description (22) Uploaded on any appropriate RF channel (such as those in the VHF band and uhf band). The RF amplifier 806 amplifies the received information signal (like a signal received on the frequency of one million hertz calling channel), and then amplifies the amplified data signal to the input of the first-mixer 8〇8. In the preferred embodiment of the present invention, the first oscillator signal generated by the frequency synthesizer or local oscillator 80 is also coupled to the first mixer 80. The first mixer 8 〇8 mixes the amplified data No. 6 with the first oscillator signal to provide a first intermediate frequency (or scary) signal (such as a 45 MHz signal), which is then coupled to the first IF filter 8 1 2 input. Especially when other calling channel frequencies are used, it will be appreciated that other IF frequencies can also be used. The output of the IF filter 812 (which is the data signal on the channel) is coupled to the input of the second conversion section 84. This conversion section will be described in more detail below. The second conversion section 814 uses a second oscillator signal (also generated by the synthesizer S 10) to mix the data signal on this channel to a lower intermediate frequency (e.g. 45 5 kHz) signal. The second conversion section 8 14 amplifies the final intermediate frequency signal to provide a second IF signal, which is suitable for coupling to the demodulator section 90S or to the linear output section 824 c. The receiver section 840 is similar to a conventional FM The receiver operates in the same way, but unlike the traditional FM receiver: the receiver section 804 of the present invention also includes an automatic frequency control section 8 1 6 which is switched to the second conversion section 8 1 4 and the frequency control section appropriately samples The second IF signal is used to provide a frequency correction signal coupled to the frequency synthesizer 81 to maintain the receiver tuned to the designated channel. The maintenance of receiver modulation is particularly important for the reception of appropriate QAM (ie, I-Q components) and / or SSB information transmitted in linear modulation. Use a frequency synthesizer to generate the first and second oscillator frequencies so that the receiver operates at multiple operating frequencies -25- This paper standard is applicable to the Chinese National Standard Falcon (CNS) A4 specification (210X297mm) (please read the back Please fill in this page again for attention). Fifth, the description of the invention (23) To choose the operation, you can borrow the code memory / number (for example, in ⑽TM agreement) to choose ^ = / or the parameters received in the air are from the automatic frequency system Bite 816 "I will be happy to see that you can also use other oscillator circuits such as the F ¢ 2 system 816 frequency correction car vibrator circuit. The fixed frequency automatic gain control 820 of ^ ㈣ is also coupled $ 太 1 n -Rong Mi Xun. To the first-transition section 814 of the dual-mode receiver of the present invention. Automatic Gain Control> In, 'j 820 estimated the second IF signal sample to the amplifier 806 to maintain the RF amplifier. Gain correction signal for gain. This gain correction signal is also combined to the first conversion section 8 to 4 to maintain the second conversion section 814 with a predetermined gain. With the linear modulation ... , And can still distinguish the dual-mode receiver of the present invention from the traditional FM receiver. As will be explained in detail below, when the information or control information is based on ™ the Ministry of Economy Central Standards Bureau staff consumer cooperatives print modulation In format transmission, the second difficulty number is consumed by the ⑽ demodulator section. The FM demodulator section 908 demodulates the second ιρ signal in a manner familiar to those skilled in the art to provide a reply-to-data letter. (Iv) The data signal corresponding to the received address and the binary information string β of the information resource transmitted in the modulation format is coupled via the input of the input / output port (or 1 / 〇 port 828) for use The input of a decoder and controller microcomputer 9〇6. The microcomputer 9〇6 provides complete operation control of the communication receiver 900, providing functions such as decoding, message storage and retrieval, display control and warning, etc. 9〇6 is preferably a single-chip microcomputer like the MC68HC05 microcomputer manufactured by Motorola and includes a CPU 840 for operation control. The internal bus 830 is connected to each operating element of the device 906. The I / O port 828 (shown in FIG. 9 The separated part) For -26-This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210X 297 mm) Printed by the Staff Consumer Cooperative of the Central Standards Bureau of the Ministry of Economic Affairs A7 --- --____ B7 V. Invention description (24) Multiple controls and data Line to provide communication from external circuits (such as battery saver switch 904, audio processor 914 'display 911, and digital storage 868) to S 906. Timing devices such as timer 834 are used to generate communication receivers Timing signals required for operation, such as battery saver timing, alarm non-timing, and message storage and display timing. Oscillator 832 provides a clock for cPU 840 operation and a reference clock for timer 834. The RAM 838 is used to store information on the firmware routines used to perform various operations to control the communication receiver 900, and can also be used to store short messages (such as digital messages). The ROM 836 contains firmware routines for controlling the operation of the device 906, including data signals such as decoded recovery, battery saver control, message storage and retrieval in the digital storage section 868, and pager operation and message presentation. -General routines required for control. The alert generator 842 provides an alert signal in response to the decoded FM modulated signal information. The code memory 9 (not shown) is coupled to the microcomputer 906 via the IO port 828. This code memory is preferably an EEProm (electrically erasable and programmable read-only memory) storing one or more predetermined addresses to which the communication receiver 900 responds. When the FM modulated signal information is received, the device 906 acts as a decoder and decodes in a manner well known to those skilled in clothing. When the information in the returned data signal matches any wrongly stored predetermined address, the subsequent received information is decoded to determine whether the additional information modulated in the FM modulation format or the linear modulation format is introduced into the receiver 〇As explained further below, when the additional information is sent in the FM modulation format, the received message data can be received and stored in the microcomputer RAM 838 or digital storage section 868, and the alarm generator 842 generates A warning signal. This warning signal is suitable for driving -27 -'s Zhang scale is applicable to China National Standards (CNS) A4 specifications (21GX 297 mm) ~ '-~ n In 1 ii-n I--I-.---J -II D *-(please read the precautions on the back before filling in this page) A7 A7 Printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economy V. Invention Instructions (25) The frequency processing circuit 914 of the converter 916 to transmit— Audible warning. Other types of perceptual warnings such as tactile or vibration warnings can also be used to warn users. σ When the additional information is to be transmitted in a linear modulation format (such as SSB or " L ^ Q "), the computer 906 decodes the index information, which includes indicating to the receiver that the channel bandwidth of the additional information will be transmitted Is the name of the sideband group, < or the combination of I and q components). The device 906 maintains the operation of monitoring and decoding the information transmitted in the modulation format until the current batch end is i. At this time, the power supplied to the receiver is suspended until the next batch specified by the specified batch or index Only when the batch arrives, the high-speed data is transmitted during this period. The device 906 generates a battery saving control signal via the I / O port 828. This signal is coupled to the battery saver switch 9〇4 to stop the supply to the FM demodulator 908. As described below, the power is supplied to the linear output section 824, the linear demodulator 850, and the digital storage section 868. The second IF output signal now carrying SSB (or "1 and Q") information is coupled to the linear output section 824. The output of the linear output section 824 is coupled to the quadrature detector, in particular to the input of the third mixer 852. A third local oscillator is also incorporated into the third mixer 852, and its frequency range is preferably from 50 to 80 Hz, but it will be appreciated that other frequencies can also be used. The signal from the linear output section S.24 is mixed with the third local oscillator signal 854. And in the second; the output of the kneader 852 generates a third IF signal, which is coupled to the second IF amplifier 856. The third IF amplifier is a low-gain amplifier that buffers the input signal and the output signal. The third output signal is coupled to [channel mixer 858 and Q channel mixer 860. I / Q Oscillator 862 provides the third if frequency _-28-This paper standard is applicable to the Chinese National Standard (CNS) A4 specification (210X 297 public) (please read the precautions on the back before filling this page) _ 定 1 ..... ί-I ...... 1 A7 printed by the Staff Consumer Cooperative of the Central Standards Bureau of the Ministry of Economic Affairs --------- B7 V. Description of invention (26) Quadrature oscillator Signals, these signals are mixed with the second output signal in the channel mixer 858 and the Q channel mixer 860 to provide the baseband 1 channel signal and the Q channel signal at the mixer output. The baseband I channel signal is coupled to the low pass filter 864, and the baseband Q channel signal is coupled to the low pass filter 866 'to provide a pair of baseband audio signals representing the compressed and deflated expansion of the sound signal. These audio signals are coupled to the digital storage section 868, and in particular to the input of an analog 'digital converter 87. The A / D converter 870 samples the signal at a rate at least twice the highest frequency component output at 864 and 866. The sampling rate of each I and Q channel is preferably 6.4 kHz. It will be appreciated that the data sampling rate indicated is only an example, and other sampling rates may be used depending on the width of the received audio message channel. Field | During the batch of dedicated speed data, the microprocessor 906 provides a count enable signal coupled to the address counter 872. The A / D converter 870 is also caused to allow the information symbol to sample it. The A / D converter 87〇 generates a high-speed sample clock signal for clocking the U address counter 872, and this counter sequentially generates the sampled sound signal into the dual-port random access memory 874 (via the other converter 87〇 to RΑΜ 8? 4 data line) address. After receiving all the sounds ... the sound signal loaded into the dual-port RAM 874 at high speed in real time is processed by the Xuan computer 9〇6. Since the microcomputer 9〇6 is not required to process the real-time information, it is eliminated and said Significant reduction in volume. The microcomputer 9〇6 takes η. Stored numbers through data and address lines, and in a preferred embodiment of the present invention, the microcomputer 9〇 & processes information symbol pairs to produce ASCII-encoded information (transmitted data resources) < case) or digitized sampling data (case of transmitting sound). Digitalized sound ____ -29- The size of this paper 1 -———— m ^^^ 1--.-III * £ {. I ii 1 ^ 1 I--,-° (please read the precautions on back 1 first (Fill in this page again) Π ^ clothing can alternatively be stored in other formats: for example ^ 匚 〇, 匚 乂 8〇 or uc based forms and other types on request. In the case of time-compressed audio signals, the L ^ Q component sampled by ADC converter 870 is further processed by cpu84〇 (via dual-port RAM 874 and WO 828) with similar operations as illustrated in the receivers of FIGS. 7 and 8 (1) audio signal Amplitude expansion and (2) time expansion of the signal. Then the sound is stored in RAM 874 again. The ASCII code or audio data is stored in the dual-port RAM until the user of the communication receiver requests to present the information. The user selects and reads the stored message using a switch (not shown) to reply to the stored ASC "coded data. When the stored eight 8 ^ 1 code Λ information is to be read out, the user selects the message to be read out and drives Unison can read the switch of the computer 9〇6 to reply the data, and present the recovered information on the display 9Π (like a liquid crystal display). When the voice message is to be read, the corpse selects the message to be read and Drive the read switch of the microcomputer 9〇6 to restore the data from the dual-port RAM, and present the recovered data to the audio unit = device 9 ~ 4. This audio processor converts the digital sound information into an analog sound signal. The signal is coupled to the speaker 916 to present the audio message to the user. The computer 906 can also generate a frequency selection is number coupled to the frequency synthesizer 80 to enable selection of different frequencies as previously described. Central Standard of the Ministry of Economy The bureau employee consumer cooperative printed the timing chart shown in FIG. 10, which shows the outward signalization code used in the radio communication system 100 of FIG. 1 according to a preferred embodiment of the present invention. The characteristics of the format, and this figure includes the details of the control box 330. The control box is also categorized as a digital box. This signaling protocol is subdivided into-hour 3i〇, one period 320, box 330, 430 '-field 340 and a word 35Γ) + μ ίί 邵 'in each hour 310 has up to 15 4 minute unique recognition cycles passed -30-This paper scale is suitable for the National Standards (CNS) Α4 specifications (21 () < 297 public Author>

,. Normally all 15 cycles of 32 ° are transmitted every hour. In each cycle, ^! Still has 128 1.875 seconds including the digital frame 330 and the analog frame 430. The unique frame is transmitted. Usually all 128 frames are transmitted. At each control frame, there is a synchronization and frame data signal 331 lasting 115 milliseconds and n unique identification fields 340 of 160 milliseconds are transmitted. During each control block 33, a bit rate of 3200 bit / s (1) 1) 8) or 6400 bit / s is preferably used. The bit rate during each control block 330 is communicated to the selective call radio at the time of the synchronization signal 33. When the bit rate is 3200 bits / sec, as shown in FIG. 10, each field 340 contains 16 unique Recognized 32-bit word 3 When the bit rate is 64⑽ bits / sec, each field 340 contains 32 unique-recognized 32-bit words (not shown in each word, as is familiar to those skilled in the art) At least Η bits TL is used for error detection and correction, and 2 bits or less is used for data: the bits and words 350 in each field 340 are utilized by techniques familiar to those skilled in the art. Interleaved transmission, to improve the error correction capability of the agreement. Printed by the Ministry of Economic Affairs Central Standard Falcon Bureau Employee Consumer Cooperative II-(Please read the precautions on the back before filling this page) The data is included in each control frame 330 Area, these data areas include the frame structure data in the field data area (BΙ) 332, one or more selective call addresses in the address area (AF) 3 ^, and the vector area (VF) 334 — Or more vectors. Vector area 334 starts at vector boundary 334. Each direction in vector area 334 Corresponds to one of the addresses in the address area 333. The data area μ] The boundaries of 333 and 334 are defined by the field data area 332. Depending on the type of system data in the synchronization and frame data area 33, the The number of addresses in the address area 333 and the number and type of vectors contained in the vector area 334 depend on the factors, and the data areas 332, 333, and 334 are variable. 31-This paper scale is subject to the Chinese National Standard (CNS ) Α4 specification (210 'x 297 mm) A7 ----- B7 V. Description of the invention (29) Employee consumption cooperation of the Central Bureau of Standards of the Ministry of Economic Affairs is printed with reference to the timing chart shown in Figure 11, which is based on The preferred embodiment of the present invention is featured in the transmission format of the outbound signaling protocol used in the radio communication system of (1), and this figure includes details of the sound box 43. The sound box here is an analog box. The hour of the agreement department is 31 〇, periodic coffee and frame coffee, 430 and period are the same as those described for the control box of FIG. 10. The various ratio blocks 43〇 have a letterhead portion 435 and an analog portion 44. Synchronization and frame data signal 331 The data is the same as the synchronization signal 331 in the control box 33. As explained above, the letterhead portion 435 is frequency modulated, while the analog portion 440 of block 43〇 is amplitude modulated. There is a transition portion 444 between the letterhead portion 435 and the analog portion 44. According to the present invention A preferred specific example. This conversion section includes a pilot 'human carrier suitable for amplitude modulation of the three secondary channels 441, 442, and 443. The analog section 440 shows the three secondary channels 44 1, 442, and 443 transmitted simultaneously. Each sub-channel includes an upper sideband signal 401 and a lower sideband signal 402 (or alternatively, together with a quadrature signal). In the working example shown in FIG. 11, the 'upper sideband signal 401 includes a message segment 415, which is the first segment of the first analog message. Included in the lower sideband 402 are four quality evaluation calls 420, 422, 424, and 426; four message segments 410, 412, 416, 41 8, and segment 4 1 4 (not used in this example). The two segments 4 1 0, 4 1 2 are segments of the second segment of the first analog message. The two segments 416, 4 1 8 are segments of the first segment of the second analog message. The first and second analogue messages are compressed audio signals, which have been segmented so as to be included in the first channel 441 of block 1 (43 0) of cycle 2 of 320. The second segment of the first message and the first segment of the second message are each divided to contain quality evaluation signals 420, 426, which are below each of the three sub-channels 44 1, 442, 443 -32- Standards Standards (CNS) A4 specifications (210 X 297 mm) (please read the notes on the back before filling in this page) m · Pack. Order A7 A7 Printed and printed by the Employee Consumer Cooperative of the Central Bureau of Standards of the Ministry of Economy V. Description of invention ( 30) Repeat at the predetermined position in * 402. One phase) · Bu iKk people, the smallest segment of the message contained in the analog box is defined as the sound increment 450, in the various analog parts 44〇 of the analog box 43〇 There is a unique-recognized 8 sound increment. The quality assessment signal is preferably transmitted as an unmodulated sub-carrier pilot signal, preferably as one of the audio quantities in the period, and preferably has a 3-separation of no more than milliseconds in the analog part of the frame. I will be happy to see that more than one message segment occurs between the two quality-estimated signals, and the message segment is typically a sound increment of variable overall length. Referring to FIG. 12, shown is a timing chart illustrating the control block and the second analog block of the out-signaling protocol used in the radio communication system of L1 according to a preferred embodiment of the present invention. The timing chart of FIG. 12 shows an example of the frame 0 (FIG. 10) of the control frame 33. The illustrated ones have four addresses 5 i 〇, 5 J i, 5〗 2, 5 13 and four vectors 520, 521, 522, 523. The two addresses 510 and 511 include a selective calling radio 106 address, and the other two addresses 5 12, 5 and 13 are the addresses used for the second and third selective calling radio 106. Include an index indicating the agreed position of the associated vector (that is, where the vector starts and its length) in each address. Each address at addresses 5 10, 5 1 1, 5 12, 5 13 uniquely associates the vector 520 , 52 1 '522 and 523 one. In the example shown in Fig. 12, the vectors 520, 521 '522, and 523 are also uniquely associated with the message part of the primary channel. In particular, the vector 5 2 0 can point to the upper sideband of the secondary channel 441 (see Figure i 丨), while the vector 522 can point to the lower sideband of the secondary channel 441. Similarly, the vector 5 2 1 can point to the second sideband of the secondary channel 442. That is, in the case of sub-channel 441, this example can show that two different message parts are carried by the upper and lower sidebands. In the case of the secondary track 442, the two halves of a message part are carried by the upper and lower sidebands, respectively. Therefore, it is best to pack the vector in it -33-This paper size is applicable to China National Standards (CNS) A4 (210X297mm) (please read the precautions on the back before filling this page). I »11—- -A7 B7 V. Description of the invention (31 includes information indicating the secondary channel (ie, radio frequency) where the receiver should find the message, and indicating whether a separate message is to be recovered from this channel or the first and first of a single message Whether the two halves will reply data. Two specific examples of N messages being transmitted simultaneously on the upper and lower sidebands (or〗 and Q channels) are that one message is a direct voice broadcast message; and the other message is about to be The voice mail message stored in the pager. The Ministry of Economic Affairs Central Standards Bureau employee consumption cooperation du printing office clothes-•-(please read the precautions on the back before filling out this page) Order a better specific example according to the present invention , By identifying the number of words 350 and the vector length (in words) after the vector boundary 335 at the beginning of the vector, the vector position can be provided. It will be appreciated that the relative position of the address and the vector are independent of each other. This It is illustrated by an arrow. An index indicating the agreed position of the associated vector (that is, where the segment starts and its length) is included in each vector, and each vector 52U ′ 52 1, 522, 523 uniquely associates the message segment 55. , 55 i '552, 553. According to a preferred embodiment of the present invention, the number of identification frames 43〇 (from 1 to 127), the number of secondary channels 441, 442, the number of materials 3 (from β 3)' sidebands 401, 4 〇2 (or I or Q) and the sound increment of the beginning of the message segment 45〇, and the length of the message segment expressed by the sound b-day volume 4 5 0, can provide the position of the message segment. For example, vector 3 (522) includes the indication Message 2 fragment 1 (552) data, which is required by the selective call transceiver 106 with the selective call address 512, whose position starts at the sound increment 46 (450) of box 1 (560) (in the figure The chime sound increment 450 is not marked in 12); and the vector 13 (523) includes the segment 9 (i) of the message 9 (553). ≪ • Bei material 'This is a selective call transceiver with selective call address 5 1 3 106 wants the sound increment 〇 (45〇) whose position starts at box 5 (561) (the sound increment 450 is not shown in the figure). The best specific example of the Ming is when the monthly sound signal is -34- A7 ㈣ / 4) 3 ________B7________ V. Description of the invention (32) Other analogue signals such as modulation and demodulation signals or dual tone multi-frequency (DTMF) signals It is alternatively included by the present invention. It should also be appreciated that the field data used in the frame structure described earlier can be used to achieve further enhancements that will allow the communication system to have a greater overall output and allow for additional Teba. "Such as" a message sent to a portable sound unit may require the approval signal sent back to the system to include information that will identify the transmitter from which the transmitter is receiving information. Therefore, the method of transmitting the information to the known portable / sound unit by using the transmitter required to terminate the portable sound unit can achieve frequency reuse in the simultaneous broadcast system. In addition, once the system knows that j can engage in the location of the audio unit, then logically implement the target information D. In another feature of the present invention, the time scaling technique such as WSOLA was previously used in the present invention. There are several shortcomings. Therefore, a technique to modify WSOLA to become related to the narrative and be freely named " WSOLA-SD · ’was developed. In order to further understand the modification of WSOLA to WSOLA-SD, a brief description of WSOLA is as follows. A technique called Waveform Similarity Basic Addition Technology (WSOLA) can achieve high-quality time scale correction compared with other techniques, and is simpler than other methods. When used to accelerate or decelerate speech, even with WSOLA technology, the speech quality is not very good. The reconstructed speech contains many processed products, such as echoes, metallic sounds, and reverberations in effects. This feature of the present invention illustrates the ability to overcome this problem and minimize the number of enhancements present in the processed product. Many parameters in the WSOLA interpretation method must be optimized to obtain the best possible quality for the known narrator and the required compression / expansion or time scaling factor. This feature of the invention can handle determining which parameters and how With -35-This paper scale applies the Chinese National Standard (CMS) A4 specification (210X297mm) -------- Installed-please read the text of the year if you want to save the Qi Cunqi () -5 ° Printed A7 B7 by the Ministry of Economic Affairs, Central Standard Staff and Consumers Cooperative V. Invention description (combining them with the compression / expansion or time scale of the speech or sound signal on the 33th with improved quality. &Quot; WSOLA performance Translation method: Let \ ⑷ be the input voice signal to be corrected 'y (n) is the time purlin correction signal and α is the time scale parameter. If α is less than', the voice M is expanded in time. If ^ 丨, the voice The signal is compressed on ^. Refer to Figure 13-17. The 'shown is the timing of several repetitions of the scaling (compression) method between ws〇L / U #, which is compared with the better Ws〇LA_sc ^ of the present invention. Assuming that the input voice signal has been properly digitized and stored, Figure 13 shows WS The OLA method repeats for the first time on an uncompressed speech input signal. The WSOLA method requires an α time scale factor (in this example, it is assumed to be equal to 2, if it is greater than 1, then it will shrink; and if α < 1 , Then expand) and randomly analyze the segment scale (Ss), which has nothing to do with the characteristics of the input speech and especially has nothing to do with the spacing. ^ The R degree of the stacked segment So is calculated by 0.5 * Ss and is fixed at Ws〇LA ^ As shown in Fig. 4, the first Ss sample is directly copied to the output. Let the index of the last sample of the output be Ifl. Determine the overlap index 〇1 as the last Ss " 2 sample from the output of the last sample 4. The sample to be superimposed is now between 〇 丨 and &: the search index (S 〖) is determined by β * 〇ι. In the initial part of the input signal, copy the employee consumer cooperative of the Central Bureau of Standards of the Ministry of Economic Affairs to print to the output &; ίfoot input; the f multi-action window of the sample is determined: this window is determined near the search index s 1. Set the start of the window to SrL offset and the end to S1 + H offset • Repeat for the first time, i = 1. In this window, use the following normalized intersection and related equations to determine the best Relevant So sample: 36 The paper wave scale applies the Chinese National Standard (CNS) A4 specification (210X 297 mm) A7 'Invention description (34 R (k) ^ ~ SΣ-^ (S + k + j) y (〇 + j ) = ο 1 = S〇) j = Σ (S. + A- + y) £ y2 (O. + j) r = 〇 / = 0; 1/2 where k = srL_ 'VH · Central Bureau of Standards, Ministry of Economic Affairs The printing of employee consumer cooperatives determines that the normalized R (k) is the maximum lag k = m. The best index is ~ 1 Π1. Note that other methods like the average size difference function (AMDF) and other correlation functions can be used to find the best matching waveform. Then the 8 samples starting at ι are multiplied by an increasing ramp function (although other weighting functions can be used) and added to the last So sample 3 in the output. Before accumulation, the s samples in the output are multiplied by a decreasing ramp function (though Other weighting functions can also be used here). The cumulative final sample will replace the last S 〇 sample in the input. Finally, the So sample next to the best matching So sample is copied to the end of the output and reused next time. This is the end of the first repetition in WSOLA. For the next 'entry repeat' referring to Figures 15 and 16, we need to calculate a new overlap index 〇2 similarly to 〇 丨. Similarly, a new search index s and the corresponding search window can be determined as previously repeated by the author. Again, in the search window, the cross-correlation equations previously described are used to determine the optimal correlation S 〇 sample, where the determined optimal sample starts with b 2. Then the S 〇 samples starting at b 1 are multiplied by an increasing ramp function and added to the end of the output; § 〇 samples. Before the accumulation, the S 0 samples in the output are multiplied by a decreasing ramp function, and the final samples of the accumulation will replace the last So samples in the input. In the end, the tightest one takes the best match S 〇 samples and the second S 〇 samples are copied to the end of the output to add the first ^ 1 ·-m — ^ n -1: 1-I--I-I. In- ------V **** 0¾, ve-* (Please read the precautions on the back before filling in this page) 37 This paper size is applicable to China National Standard (CNS) A4 specification (210X297mm) A7 B7 Fifth, the invention description (35 Ministry of Economic Affairs Central Standard Falcon Bureau Employee Consumer Cooperative printed the rafters for the next reuse, which will have the i-th repetition will have-overlap index 〇1, a search index Si, output Ifi last The sample, and the best finger drum B 丨. Figure 17 shows the final output of the two repetitions described previously with reference to Figures 13 to 16. It should be noted that there is no overlap in the final output signal between these two repetitions. The quasi-method continues this method. The WSOLA method scales (compresses) the entire sound signal, but there will never be any overlap between the repeated results. The WOLA time scale expansion is done in a similar way. ^ Relative In the preferred method of the present invention (ws〇LA_SD), several blocks or missing poles of Ws〇la Obviously. When tracing Ws〇lA-Sd & ~ Xia Xuan shown in Figure 18_23, these shortcomings should be kept in mind. The main shortcomings of ws〇LA package f can not get a good quality time scale voice, this is because The fixed analysis η τ and the scale (Ss) are used for all input speech without considering the spacing characteristics. For example, if Ss is too large for the input speech signal, the expanded final speech will be echoed and reverberated. If the input voice signal is ~ too ', the expanded final voice will make a harsh sound. When the shrinkage rate (〇〇 is greater than 2, the second significant drawback of Ws〇LA arises. In this case, repeat The separation of the moving windows between them will cause this party to miss important input speech components, thus seriously affecting the comprehensibility of the final input + sound. During the repetition, increase the moving window [degrees to compensate: the stacked search window will When the result of the cross-correlation function leads to more rounds of linguistic loss, and it still has a significant impact on the final output of the variable voice ^ Please read the back * first, and then fill in the I page. The degree of WSOLA method A shortcoming includes the inability to provide the designer or Lu with 38 copies of the paper standard suitable for the financial standard rate A4 specification (A7 ------ B7 V. Description of the invention (%) due to the known sound quality and known limitations The elasticity of the computational complexity of the system is 1 and the known time latitude factor)). Because the overlap (f) in the WSOLA method is also 0_5, it is especially obvious. Therefore, in an application that requires high-quality speech reproduction If there is enough processing power and memory, the WSOLA-SD method of the present invention can utilize higher overlap to provide higher quality speech reproduction at the cost of increased computational complexity. On the other hand, in an application limited by processing power, memory, or other limitations, ▲ WSOLajd can reduce the degree of sophistication, so that considering the existing special application limitations, the voice quality is only sacrificed to the level of hearing. Du Printed by the Consumer Consumption Cooperation of the Central Standard Falcon Bureau of the Ministry of Economic Affairs d. The overall block diagram of the WSOLA-SD method cannot be shown. In this block diagram, Ss, f, and α are calculated depending on whether or not the voice will be compressed or expanded. This Ws〇la-SD interpretation method has provided a huge improvement to the WS〇1A method in terms of reconstructed speech quality: The WSOLA-SD method is related to the narrator, especially related to the distance between specific narrators. Therefore, the pitch decision 2 is completed before determining an analysis segment scale U4). For known 丨 'and π (this visual distance determines 12 corrections to provide a corrected α (16)), WSOLA-SD time scale (18) speech. This time scaling can be the expansion or compression of the input signal. Alternatively, if α > 丨, a time scale signal with an α factor is inserted; or if α < 1, a time scale signal with a factor of 1 / 〇 is selected by ten to obtain a frequency scaled signal. As described in Oppenheim & Schaefer, separate time signal processing, interpolation and f are selected as well-known techniques for digital signal processing. For example, suppose that the 2-second interval of language 3 is sampled at 8 Hz, and the signal has a significant frequency component between 0 and 00 Hz. This system assumes that the input voice signal is a time-scale signal scaled by a factor of 2. The final signal will have a length of 1 second -39-This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (21 Ο 297 mm) A7 A7 37 V. Description of invention (degrees between 0 and 4000 Hz will remain It has a significant frequency component. This signal is inserted with a factor of α = 2 (see pppenheim & Schaefer said month). This will result in a 2 second long signal, but with a frequency between 0 and 2000 Hz The Feng component β borrows a frequency constriction signal with a factor of α = 2 from ten, and can return to the inter-scale field to obtain the original time-scaled speech without any loss of capital connotation (frequency component between 0-4000 Hz) ). With reference to FIGS. 18-22, shown is a timing diagram of several repetitions of the ws〇la-SD time scale (compression) method according to the present invention. If the input voice signal has been properly digitized and stored, FIG. 1 8 shows the first repetition of the SOLA-SD method on an uncompressed speech input signal. The WSOLA-SD method also needs to determine the appropriate spacing period of the voiced portion of the input speech signal. The spacing determines and how to obtain the segment size from it Is briefly explained in the next 3 1) Enter the voice box into a 20 millisecond field, 2) Count the energy in each field. 3) Calculate the average energy of the parent field. 4) Determine the energy threshold to detect the voiced speech as a function of the average energy of each field. 5) Use this energy threshold to determine the continuous fields of voiced speech of at least 5 field lengths. 6) Perform a pitch analysis on each field of the continuous voiced speech found in step 5 'This can be accomplished using various methods including modified automatic correlation method, AM D F, or truncated automatic correlation method. 7) Use an intermediate filter to smooth the pitch value to eliminate the estimation error. 8) Take the average of all smoothing spacing values to get the proper narrator room @ -40 This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210XM7 mm) n In — ^ ϋ n 1— mu— m · I ^. *-(Please read the precautions on the back before filling in this page) A7 B7 _--- the consumer cooperation cooperation of the Central Bureau of Standards of the Ministry of Economic Affairs A5 B7 _--- V. Description of invention (estimation of 38. / Meal-(please first Read back ^ 7 ''s notes #fill in this page) 9) Therefore, the fragment size S s is calculated as follows. If the interval P is greater than 60 samples, Ss = 2 * The interval P is between 40 and 60 samples, Ss = 1 20, and the service P is less than 40 samples, S s = 10. In all cases above It is assumed that the sampling rate is 8 kHz. The important factor in providing WSOLA-SD with the advantage of overcoming some of the disadvantages of Ws〇LA explained earlier is the degree of overlap f. If the degree of overlap f in WSOLA-SD is greater than 0.5, this provides higher quality at a more complex cost. If the degree of overlap f in Wsolla-SD is less than 0.5, this reduces the complexity of the rendering method at the cost of quality. Therefore, users have more flexibility and control over the design and use of special applications. For the printed printing of the employee consumer cooperative of the Central Bureau of Standards of the Ministry of Economics, refer to Figure 18-23 again. The WSOLA-SD method requires a time scale factor of Shen (in this example, it is assumed to be equal to 2, if 〇 / > 1, it is compressed; and if With < 丨, it is an extension) and a pair of input voice characteristics optimized for the analysis segment size (Ss) (called Ma Narrator Spacing). The overlapping segment scale so is calculated as f * Ss, and So is fixed in WSOLA-SD for a known interval period and f '. In the example shown, Γ is greater than 0.5 to show higher quality final output speech. The first Ss sample is copied directly to the output 3 and the index of the most drunk sample is set to " I. The undecided overlap index of the last available sample from the output can be the same as the s〇 sample. As shown in Figure 19, the sample to be superimposed now is between 〇 1 and〗 ^ From the figure]. 8 Hearers, the first search index (s 〖) is determined by Shen. After the initial part of the round eight signal is copied to the output, the position of the sample moving window from the input voice signal can be determined. This window is determined near the search index. A7 _-__________B7 printed by the employee consumer cooperative V. Description of invention (39). In this window, the best correlated 8 samples are determined using the cross-correlation equation previously described above, where the determined best sample starts with B, D, and So samples starting with B, multiplied by-gradually The increased ramp function (although other weight functions can be used) is added to the last S〇 sample in the output. Before accumulation; the So samples in the output are multiplied by a decreasing ramp function. The accumulated last sample will replace the last sample in the input. Finally, the next sS-So sample that follows the previous best Western match So sample is copied to the end of the output for the next iteration. This will be at the end of the first repetition of WSOLA-SD. For the next repetition, refer to Figures 20 and 21. We need to calculate the new overlap index like 〇 〖2. Similarly, as repeated in the previous time to determine the new search index S 2 and the corresponding search window. Again, in the search window, the cross-correlation equation previously described above can be used to determine the best correlated So samples' where the best sample determined is B 2 ^ and then the So samples starting at B 2 are multiplied by a gradual The increased ramp function is added to the last so sample in the output. Prior to accumulation, the S 0 samples in the output are multiplied by a decreasing ramp function. The last sample accumulated will replace the last So sample in the input. Finally, the next Ss-So sample that closely follows the previous best matching So sample is complex to the end of detection, so that it can be used for the next iteration. Figure 22 shows the final output signal after two repetitions using the WSOLA-SD method. Note that there is an overlapping area (Ss-So) in the final output signal. Compared with the WSOL.A method, this area ensures an improved understanding and prevents this method from missing important input speech components.

Referring to FIGS. 23 and 24, the display is based on the use of WSOLA-SD-42 according to the present invention-this paper scale is applicable to the Chinese National Standard (CNS) A4g (210X297mm) --------: installed ---- --Order • (Please read the precautions on the back and then fill out this page) A7 printed by the Employee Consumer Cooperative of the Central Standardization Bureau of the Ministry of Economic Affairs ---------_____ B7__ Gram, Invention Description (40) " '一"-The input timing diagram and output timing diagram of the time scale expansion of the first repeating example of the method. Except that (overlap index) moves faster than \ (search index), the function of the extension method is basically similar to the example shown in Figure 18_22. In order to be correct, during the expansion period, the movement is π times faster. The analysis segment scale s »is related to the interval between input languages. The degree of overlap can range from 0 to ^ 'but 0.7 is used in Figures 23 and 24. In this case, the time scale factor or will be the inverse of the expansion rate. Assuming an expansion rate of 2, the time scale factor α -0.5. The overlapping segment sibling degree so will be equal to f * Ss * overlapping degree multiplied by the analysis segment R degree. Therefore, after several repeated overlapping accumulations and using a gradual ramp function on each optimal input segment and a decreasing ramp function on each output overlapping segment, the input speech signal is expanded before the accumulation To output voice signals while maintaining all the advantages of WSOLA-SD as explained previously. Further improvement can be obtained by dynamically adjusting the clip scale S s in the WSOLA-SD rendering method with clip spacing at that instant. This can be done by modifying the way explained earlier. If we use a short segment scale of Ss = 100 (assuming a sampling rate of 8 Hz) for silent speech with improved quality, the segment & degree of speech will be Ss = 2 * spacing. In order to decide whether the audio clip is voiced or unvoiced, some changes need to be made at the same time. The methods of these changes are described as follows 3 1) Put the scorpion into the speech frame into a 20 millisecond field. 2) Calculate the energy in each field. 3) Calculate the number of zero crossings in each field. 4) Calculate the average energy of each field. 5) The energy threshold is determined as a function of the average energy of each field -43- This paper scale is applicable to the Chinese National Standard (CNS) A4 specification (210 X 297 mm) n II-,--11 !-II! 1 1 II 1---^ aJ (Please read the precautions on the back before filling in this book) A7 B7 L, invention description 'ΉΊ / voice voice. 5) Use this energy limit and the continuous field limit of zero voiced speech ^ at least 5 field lengths 6) Perform a flat 3 Ψ distance knife analysis of all voiced segments and parse them in each of these voiced segments * Dai ^ = I This can be accomplished using various methods including amended automatic correlation method, Α ~~ 鞔 excluding automatic method. "Segments that are not labeled as voiced speech are now marked as experimental silent segments. ^ Take the continuous field of the 5 boxes in" Experimental Silent Segments "and extend the distance + analysis. Determine the ratio of the largest to the smallest correlation coefficient. If it is a ratio, then the segment is classified as silent; or if it is a small ratio, the segments are marked as voice, and determine the start and end of the second segment of those segments. 9) The segment size w of each segment of these classified voice segments is determined as follows (1 and large-scale filming (please read the precautions on the back before filling out this page). Installed. Va Printed by the employee consumer cooperative of the Central Standards Bureau of the Ministry of Economic Affairs If the length of the system is reduced by sound, Ss = 2 * If the spacing is unbounded, Ss = 100 (assuming a sampling rate of 8 kHz) See the WS0LA_SD method on the time scale, but with a variable segment scale. Here It can be determined by the position of the input speech piece used for processing at the moment. Depending on its position, the determined fragment size. It is used for processing m technology will result in-a higher quality time scale speech signal if it is in our The situation of the communication system of Ws〇la_sd is used for the compression term, but the wave will be used continuously for the same voice input signal. For the known average time scale factor using several techniques, it can be further improved Reconstruction -44 All paper wave scales apply the Chinese national standard (CNS> A4 specification (210x1 ^^ · V. Invention description (42 A7 B7 The quality of the voice signal printed by the Central Bureau of Standards of the Ministry of Economic Affairs-Industrial and Consumer Cooperatives. It can be seen from the perceptual test For a given voice quality ', a voice signal with a higher fundamental frequency (lower pitch period) can be compressed more than a voice signal with a lower fundamental frequency (higher pitch period). For example, on average children and Female narrators have a higher basic frequency. Therefore, their ^ sounds can be compressed / expanded by more than 10%, without significantly affecting their equivalent speech quality. On the contrary, on average, those with lower basic frequency speech Male narrators can reduce their equivalent speech by 10% compression / expansion. Therefore, in a typical communication system of a narrator with roughly the same number of souther and lower basic frequencies, the same compression / expansion (time scale) Degree) factor can obtain the overall quality improvement of voice reproduction. Another feature of the expansion and compression of this technology is to make it stronger: for example, note that most of the processed products in the voice are on the scale of the voice signal Produced during the expansion. The more the voice signal is expanded, the more processed products. It is also observed that if the voice signal is regenerated slightly faster (10% lower) than the original voice, the change in rate It is difficult to detect, but the processed product is significantly reduced. This feature helps to expand the voice signal with a lower expansion factor, because the processed product is reduced by 7 and the quality is improved. For example, if the input voice is shortened by a time scale of 3, It will be expanded by a factor of 2 to 7 during the expansion period.% Anthony said that the voice will be performed 10% faster. Because the change of this voice rate will not be obvious and reduce processed products, it should be used in applications where the accuracy of the voice is not absolutely important. It is implemented by the method of clothing invention. It takes less time and therefore I--I--— ^ ϋ — ^ 1 .....-_ m ^^ 1 ml IT * 4¾ > -0 »» (please first Read the precautions on the back and then fill out this page] -45 This paper scale is applicable to China National Standard (CNS) Α4 specifications (210Χ: 97mm)

Claims (1)

A8 B8 C8 ____ ^ ______ VI. Scope of patent application " ~~ 一-1. A strider with less than one transmitter base station and multiple selective call receiving. The machine uses a sound compression communication system, including: At the transmitter base Taiwan: an input device to receive audio signals; a processing device to compress the audio signal using time-scale compression and a single band modulation technique to provide a processed signal; and a transmitter to transmit the processed signal Signals of each of the multiple selective call receivers:-selective call receivers, used to receive transmitted processed signals; a processing device, using single-sideband demodulation and time scaling Expansion and demodulation of the received processed signal to provide a reconstructed signal; and an amplifier to amplify the reconstructed signal into a reconstructed audio signal Modulation technology provides the transmission of a single message split between an upper sideband and a lower sideband. 3. The communication system according to item 1 of the patent application scope, in which the single-sideband modulation technology provides a single message repeated on the upper and lower sidebands, printed by the Central Standard Falcon Bureau Employee Consumer Cooperative of the Ministry of Economy HI 111 fm ^^^^ 1 nn I nn In ml n ^ n nn — ^ n one eJ (please read the precautions on the back before filling this page) Wall Green c. I Μ 1 rj * 4 4-According to the 1st Item of the communication system, where the system also includes: At the transmitter: Indicate the carrier signal generator, used as a reference for the amplitude and phase of the missed amplitude caused by the channel difference; -46-This paper scale is applicable to the Chinese National Standard (CMS) A4 Specifications (210X297mm) A8 B8 C8 D8 printed by the Consumer Cooperative of the Central Department of Economics of the Ministry of Economic Affairs 6. The scope of patent application is in the receiver: a receiver circuit for detecting, filtering and indicating the carrier signal generator The resulting amplitude and phase reference response. 5 · —Selective call receivers that receive compressed sound signals, including: —Selective call receivers to receive transmitted processed signals, which include compressed sound signals that have been compressed using time-scale compression A processing device 'uses single-sideband demodulation and time scale expansion and demodulation to process the received signal to provide a reconstructed signal; and an amplifier to amplify a reconstructed signal into a reconstructed audio signal 6. According to the application The selective call receiver of the patent scope item 5, wherein the selective call receiver further includes: a receiver circuit for detecting, filtering and detecting the pilot carrier signal generator in the transmitter of the base station The generated amplitude and phase reference ring 7 — a selective call paging base station that transmits selective call signals on communication resources with a predetermined frequency bandwidth, including: an input device for receiving multiple audio signals; A device for sub-channelizing this communication resource into a predetermined number of sub-channels; amplitude compression and filtering modules for each sub-channel Reduce the amplitude of individual audio signals and filter individual audio signals; a time compression module to compress individual audio signals of each sub-channel -47-This paper wave standard is applicable to China National Standard (CNS) A4 specification (210X297 mm) ( Please read the precautions on the back and then fill out this page.} Install. 、 Applicable patent scope A8 B8 C8 D8 Employee's consumption cooperation of the Central Standards Bureau of the Ministry of Economic Affairs to print time; and-Quadrature amplitude modulation transmitter to transmit the processed signal. 8. The selective beer calling base station according to item 7 of the patent application scope, which is used to receive multiple audio signals > 私 i # 其 · '员 L 哬 夂 Round-in device includes one for receiving from one The voice of the computing device is from, or the sister 4 * The calling terminal of the heart's cravings. 9 _According to item 7 of the patent scope of application w_ <, Μ 不 / 月 selective calling the calling base Platform, where the amplitude compression and filtering module includes -4 ^ 3 drafts combined into an analog / digital converter, the heart is nicknamed the waver 'analogy / zhao ^ silk TO V_ β I digital converter is coupled to a band pass Filter, this bandpass filter Coupling stride-white " automatic gain controller and cut-off circuit. 10. According to the selective call broadcasting of item 7 of the patent application scope, the time compression module contains a # _ _ # σ υ. M time scale Compression technology uses audio signal processing devices. 11. I hate the selective call broadcasting base station according to item 7 of the patent application. The time scaling module in it includes-processing using one WSOLA time compression to compress audio signals Π.—Selective call receiver unit for receiving compressed sound selective call signals, including: a receiver with an analog / digital converter to provide digitized received signals; It is used to implement single-sideband demodulation and at least-wave pilot carrier; use a pre-feedback loop to implement automatic gain control; decompress and expand the digitized received signal to provide-processed signal capability; and Its name is called 遽 遽 or Gong ^^ 1 ^^ 1 mi ^ i 11 (n In HI I—.--I «In ^^^ 1 ^ 1«, ve {Please read the notes on the back before filling in this Page) -48-Clothing Paper Scale Use the Chinese National Standard (CNS) A4 specification (210X297mm 8 8 8 8 ABCD. Six, apply for a patent scope-a digital / analog converter and reconstruction filter, and convert it into a-digitized audio signal; and used to convert the processing to one An amplifier is used to amplify the digitized audio signal. 13. A communication base station, including: a terminal for receiving audio voice signals; an analog / digital converter for converting this audio voice signal into a digitized voice signal; a digital signal processor The digital speech signal is processed by the function of splitting the digitized speech signal and one of the functions of less band-pass filtering, automatic gain control, time scaling, enlarging expansion, or buffering; and a transmitter, at least With a Hibert transform filter coupled to a digital / analog converter, this digital / analog rod is combined into a reconstruction filter, which is coupled to a positive III binding (please read first Note on the back and then fill out this page) Converter amplitude modulator, this modulator is coupled to a radio frequency power amplifier Printed by the Ministry of Economic Affairs, Central Bureau of Precinct Employee Consumer Cooperatives -49-This paper size applies to China National Standards (CNS ) A4 specification (210X297mm)