CN101917375B - Method for adding audio frequency signaling in broadcast system - Google Patents

Abstract

The present invention provides a method for calculating and adjusting the time delay of synchronous audio in the FM synchronous broadcasting system, which is characterized in that the audio front-end server of the system performs the following steps: communicating with the man-machine interface module to obtain the digital FM synchronous broadcasting system User data at the application layer; polling is used to select the data frame type to be sent, synchronous frames and asynchronous frames are sent in turn, and only one synchronous frame is sent per second; data from the application layer to the data link layer is framed; the data The data in the link layer is modulated by FSK; the digital audio signal is received from the audio input terminal, and after low-pass filtering, it is added to the FSK waveform modulated by the accompanying audio signaling, and after CODEC D/A conversion, it is converted by digital or analog The audio interface outputs to the post-stage equipment.

Description

A method of adding audio signaling in a broadcasting system

technical field

The invention relates to FM synchronous broadcasting system technology, in particular to a system and method for solving the audio delay adjustment problem of digital FM synchronous broadcasting system. the

Background technique

FM simulcasting refers to a form of broadcasting in which two or more transmitters located at different locations use the same frequency to broadcast the same program content. The establishment of the FM synchronous broadcasting network can well solve the problem of using a single frequency for large-scale FM broadcasting. the

The advantages of FM synchronous broadcasting are manifested in the following two aspects. First of all, because FM synchronous broadcasting adopts the coverage method of low-power on-demand distribution, it can reduce electromagnetic pollution and save operating costs for the station. At the same time, its single-frequency coverage, compared with the traditional FM radio coverage method, not only can save a lot of spectrum resources, but also does not need to frequently adjust the frequency of the receiver, providing more and more mobile receivers (such as car drivers) , bringing great convenience. the

To realize FM synchronous broadcasting, it is not feasible to broadcast the same program simply by using multiple stations distributed in different locations and using the same carrier frequency. As shown in Figure 1, transmitter A and transmitter B use the same carrier frequency to broadcast the same program. In the area covered by transmitter A and transmitter B alone, due to the capture effect of the FM receiver, the listening effect of the FM program is very good. good. However, in the coherent area where the signal field strength difference from the two transmitting stations A and B is less than 15 dB, a serious interference sound will generally be heard due to co-channel interference. the

The key problem to be solved by FM synchronous broadcasting is to ensure the listening quality in this coherent area. The specific solution is the realization of "Three Ones Guarantee" (according to GY/T154-2000). "Three same" refers to the same frequency, same phase, and same modulation degree. "One guarantee" refers to guaranteeing the lowest usable field strength. The in-phase here is to ensure that the phases of the audio signals transmitted by two adjacent transmitting stations received by the receiver are the same, that is to say, the audio signals must have the same audio delay during transmission. The realization of "Three Sameness Guarantee" can reduce the scope of the coherent area from the overlapping area where the field strength difference is less than 15dB to the overlapping area where the field strength difference is less than 6dB. Second, it can reduce the distortion in the coherent area and significantly improve the Hearing effect in the coherent zone. the

In order to ensure the same audio delay in the "three-same", it is necessary to adjust the adaptive delay of the audio signal received by the exciter. Because in the transmission process of the audio signal from the broadcasting control center of the radio station to the exciters of each transmitting station (as the exciter A and exciter B shown in Figure 2), not only will A, B and the broadcasting The audio signals of each exciter have a fixed audio delay difference due to the different distances from the control center, and factors such as changes in the audio transmission link route, clock changes, and multiplexing and demultiplexing of signals during transmission will cause audio time delays. Delayed uncertain changes. To ensure the same audio delay, it must be able to automatically measure and automatically compensate for this uncertain change. The specific method is to transmit a time scale signal while transmitting the broadcast signal, and the two are simultaneously transmitted in the same channel with the same audio delay. In this way, the exciter can perform corresponding delay adjustment according to the measured time delay of the time scale signal. the

However, the process of adding, separating and synchronizing the audio time scale signal has always been a difficult problem to solve. It can neither affect the quality of the audio signal transmitted by the FM synchronous broadcasting system, but also ensure that the system can be accurately synchronized. the

Contents of the invention

In order to solve the same audio delay problem of the digital FM synchronous broadcasting system, the 1PPS second pulse signal provided by GPS is used as the time scale signal, and frequency division multiplexing is transmitted in the channel through FSK modulation. The present invention provides a channel audio signaling Add and extract methods. Described technical scheme is as follows:

A method for calculating and adjusting synchronous audio time delay in an FM synchronous broadcasting system, characterized in that the audio front-end server of the system performs the following steps:

Step A: Communicate with the man-machine interface module to obtain the user data of the application layer of the digital FM synchronous broadcasting system;

Step B: Select the data frame type to be sent by polling, the synchronous frame and the non-synchronous frame are sent in turn, and only one synchronous frame is sent per second;

Step C: Data framing from the application layer to the data link layer;

Step D: Carry out FSK modulation to the data of data link layer;

Step E: Receive the digital audio signal from the audio input terminal, add it to the FSK waveform modulated by the accompanying audio signaling after low-pass filtering, and output it to the rear stage through the digital or analog audio interface after CODEC D/A conversion equipment. the

Preferably, the data frame type sent by the system audio front-end server includes a synchronization frame, a site information frame and a command frame, the synchronization frame includes a time delay reference point as a time scale signal, and the site information frame provides parameter control for each exciter, and the command The frame can centrally set the parameters of all actuators in the whole network. the

Preferably, the frame header of the synchronization frame is aligned with the pulse-per-second signal provided by GPS. the

Preferably, a CRC check is performed on the frame header of the synchronization frame and the data frame respectively. the

Preferably, for the FSK modulation of the data link layer data, a table look-up plus keying method is adopted. the

A method for calculating and adjusting synchronous audio time delay in a FM synchronous broadcasting system is characterized in that the digital FM exciter of the system performs the following steps:

Step A: Perform low-pass filtering on the received digital audio signal for use by the exciter code modulation; perform high-pass filtering on the received digital audio signal to obtain the FSK modulated wave of the signaling;

Step B: The correlation detection algorithm of FSK performs demodulation by comparing the maximum value of the square law of the output of the correlator. Correlate the obtained FSK modulated wave with the in-phase component and quadrature component of the frame header data to obtain two sets of correlation calculation results, perform square calculation on the two sets of values point by point, add the results, and save the calculation results . At the same time, the maximum value is calculated in the result, and the corresponding point of the maximum value found within a frame period is the frame synchronization point, thereby recovering the relative positional relationship of the accurate sampling points of the entire transmission frame, and realizing sampling point synchronization, bit synchronization and frame synchronization . the

Step C: Perform CRC check and frame analysis on the obtained signaling data to obtain correct network management information, and calculate the audio delay by combining the 1PPS standard time scale signal of GPS. the

Preferably, in step B, the modulated wave is used to perform a correlation operation with the frame header to realize frame header synchronization and to realize sampling point synchronization. the

Preferably, the system clocks at both ends of the transceiver are phase-locked with the 10MHz reference frequency standard provided by GPS; the starting position of demodulation is fine-tuned at regular intervals. the

The beneficial effect that technical scheme of the present invention brings is:

The channel-associated audio signaling provided by the present invention is transmitted in a frequency division multiplexing manner, and the receiving end only needs to perform high-pass filtering and FSK demodulation to obtain the channel-associated audio signaling added by the sending end. This channel-associated audio signaling The joining method is suitable for any audio transmission system with a transmission bandwidth of 20KHZ, and has the widest applicability, and this method does not need to add any additional equipment, which brings great convenience to the user in the actual networking process. the

Description of drawings

Fig. 1 is a schematic diagram of FM synchronous broadcasting according to the present invention;

Fig. 2 is the schematic diagram that the FM synchronous broadcasting system of the present invention calculates audio time delay under GPS standard time scale signal synchronization;

Fig. 3 is the schematic diagram of time-division multiplexing of channel-associated audio signaling and audio signals;

Fig. 4 is a schematic diagram of a channel-associated audio signaling transmission scheme of time division multiplexing;

Fig. 5 is a schematic diagram of frequency division multiplexing of channel-associated audio signaling and audio signals;

Fig. 6 is the hierarchical architecture of the channel-associated audio signaling transmission system of frequency division multiplexing;

Fig. 7 is the specific composition of each layer of the channel-associated audio signaling transmission system of frequency division multiplexing;

Figure 8 is the transmission sequence of various functional frames;

Fig. 9 is a block diagram of the hardware platform;

Fig. 10 is a schematic diagram of the software function module of the sending end;

Figure 11 is a flow chart of the CRC check fast algorithm;

Fig. 12 is the realization block diagram of FSK modulation module;

Fig. 13 is a schematic diagram of the software function module of the receiving end;

Figure 14 is a schematic diagram of sampling point synchronization and frame synchronization using a correlation and square law judger;

Fig. 15 is a block diagram of data demodulation processing;

Detailed ways

The present invention will be further described below in conjunction with the accompanying drawings and specific embodiments, but not as a limitation of the present invention. the

In the FM synchronous broadcasting of the present invention, the 1PPS second pulse signal provided by GPS is used as the time scale signal, which is multiplexed with the audio stream and transmitted in the channel. As shown in Figure 2, at the sending end, the audio front-end server adds the pulse-per-second signal provided by the GPS into the audio stream as an associated audio signaling. At the receiving end, the exciter extracts the associated audio signaling, and compares the time scale signal with the local GPS second pulse to obtain the audio delay caused by the transmission process. In addition, the downlink management information of the network can also be included in the channel-associated audio signaling, so that the remote control of each exciter can be realized, which provides great convenience for the management of the entire network. the

For the transmission of audio signals mentioned above from the broadcast control center to the FM exciters of each transmitting station, it can be realized by utilizing the special network provided for FM simulcast. These specialized networks can be optical fiber networks, microwave networks or PDH/SDH networks of telecommunications. However, this special network has a general disadvantage that it is expensive to build and difficult to promote. the

So I hope to borrow the already mature DVB transmission link. First look at the DVB satellite transmission link. At present, most of the central and provincial TV stations already have DVB-S digital TV satellite programs. Moreover, one channel of satellite TV program can also include six channels of audio in addition to the TV audio of the program itself. The audio signal to be transmitted can be combined and transmitted as an additional audio and TV program. Since there is no need to create another channel separately, there is no additional channel occupation fee. In addition, satellite broadcasting covers a wide range, which is suitable for the establishment of a large-area FM synchronous broadcasting network. Moreover, satellite broadcasting is not limited by geographical conditions, and some places with harsh geographical conditions that are not suitable for laying wired lines can also receive signals transmitted by satellites. At present, the cable TV network of DVB system has a very high penetration rate in cities, and the audio transmission scheme realized by satellite links can be almost completely transplanted to the cable TV network. All of these make DVB transmission link the first choice for setting up FM synchronous broadcasting network. However, whether it is a dedicated network or a DVB network, it may cause uncertain changes in audio delay, so no matter which transmission method is used, delay adjustment is required. the

Aiming at the most commonly used DVB transmission link in the FM synchronous broadcasting network, the following considerations have been made on the method of adding audio signalling: some MPEG-2 encoders used in the DVB system only support analog audio input, and the network The provider is generally not allowed to modify the existing equipment, so it is necessary to add the associated audio signaling with the time scale signal to the analog audio stream to ensure that the two always maintain the same delay during transmission. The method of joining is to modulate the associated audio signal in digital form first, and then multiplex the modulated signal with the analog audio. Since the channel-associated audio signaling data rate is relatively low, and FSK modulation and demodulation are considered to be relatively simple, this embodiment adopts the FSK modulation mode. the

There are two ways to multiplex the modulated waveform of the channel-associated audio signaling and the analog audio: time division multiplexing and frequency division multiplexing. the

First look at time division multiplexing, as shown in Figures 3 and 4. First, the input audio stream is compressed in the time domain by sampling rate conversion to allow time slots for signaling. When receiving, first use a higher sampling rate to receive the composite signal, and then separate it. During the separation process, the rotary switches at both ends of the transceiver must be opened or closed at the same time. If there is an error in the synchronization process, the signaling will be mixed into the audio signal, which will seriously interfere with the quality of the broadcast audio signal. In the actual implementation process, due to the possibility of spurious signaling in the signal, it is difficult to guarantee this strict synchronization, which is absolutely not allowed in broadcast-level equipment, so this is not used in this recommended embodiment way of reuse. the

Then look at frequency division multiplexing, as shown in Figure 5. Since the spectrum of most audio signals is concentrated below 15KHz, the idea of frequency division multiplexing is to modulate the accompanying audio signaling to the high end of the frequency spectrum of the audio signal above 15KHz through FSK modulation. In this way, at the receiving end, only a simple low-pass filter is required to filter out the influence of signaling on the audio signal without affecting the quality of the broadcast audio signal. Moreover, this multiplexing method is applicable to any system with an audio transmission bandwidth of 20KHz, and has wide versatility. Therefore, in this embodiment, frequency division multiplexing is selected to realize multiplexing of channel-associated audio signaling and analog audio. the

According to the previous analysis, it can be seen that the most commonly used audio transmission method in FM synchronous broadcasting is the transmission link of DVB. In this transmission link, need to use MPEG-2 coder. The use of this MPEG-2 coder affects the transmission of frequency division multiplexed associated audio signaling. Because the encoding standard of MPEG-2 makes full use of the psychoacoustic characteristics of the human ear, the audio signal is lost and compressed. The human ear is insensitive to high-frequency components in audio, so if the high-frequency components are too small in magnitude or masked by time or spectrum, the encoder will not encode and transmit them. Since the channel-associated audio signaling is modulated to the high end of the frequency spectrum of the audio signal when frequency division multiplexing is adopted, it is likely that the signaling is not encoded and transmitted for a certain period of time after passing through the MPEG-2 encoder. the

Aiming at the characteristics of the MPEG-2 encoder, this embodiment carefully selects the transmission parameters of the channel-associated audio signaling. Here, the modulation frequency of FSK should be as low as possible, such as 17KHz and 18KHz. The modulation amplitude requirement of FSK is moderate, and -24dBFS is selected in this embodiment. The duration of the FSK modulation symbol is chosen to be 1 ms. The duration is short, so that when a certain section of FSK modulation waveform is lost due to MPEG-2 encoding, a new complete frame of signal can be found as soon as possible, and correct demodulation can be performed again. Due to the low data rate of signaling, the lost data can be compensated by the retransmitted data later, and the audio delay does not need to be adjusted every second, so the signaling performance will not be affected. In addition, at the sending end, a 15KHz low-pass filter is performed on the audio signal to reduce the interference of the audio signal to the signaling. In the designed frame structure, a CRC check is also added to detect possible demodulation errors. the

This application designs a layered architecture of frequency division multiplexing channel-associated audio signaling transmission system. It mainly includes four layers, namely physical layer, data link layer, transport layer and application layer. The system structure of this system corresponds to ISO/OSI and TCP/IP as shown in Figure 6. the

The system adopts a layered architecture, which can bring two benefits: First, according to the abstract layering of different characteristics, each layer implements a clear function, which can avoid the confusion of the functions of each layer. Second, each layer is relatively independent, so that the tasks assigned to each layer can be realized independently. This way, when a solution provided by one of the layers is updated, it does not affect the other layers. the

The specific composition of each layer is shown in FIG. 7 . the

Channel-associated audio signaling defines three functional frames: synchronization frame, site information frame and command frame. The synchronization frame contains the delay reference point as a time scale signal, the station information frame provides parameter control for each exciter, and the command frame can centrally set the parameters of all exciters in the entire network. During transmission, synchronous frames and non-synchronous frames are sent in turn, and only one synchronous frame is sent per second. The frame header of the synchronization frame is aligned with the second pulse signal provided by GPS, as shown in Figure 8. the

The signaling transmission system is implemented on a general-purpose processing hardware platform for baseband audio signals. This hardware platform includes seven modules as shown in Figure 9: man-machine interface module, digital audio transceiver module, analog audio CODEC module, FPGA and CPLD module, DSP module, power supply module and clock module. the

Among them, the analog audio CODEC module and the digital audio transceiver module are the input and output modules of this platform. Audio signals in analog or digital form are sent to the post-stage module for processing through these two modules; or the system can output the processed audio data through these two modules for use by the next-level equipment. the

The DSP module is the core of the entire hardware platform, and almost all signal processing functions are completed in this module. the

FPGA and CPLD modules are used to provide signal transmission and data cache for the above modules. Some signal processing functions are also implemented in this module. the

The man-machine interface module provides a communication interface for man-machine dialogue. The power supply and clock module are the necessary conditions for the normal operation of the whole circuit. the

The sending end, that is, the audio front-end server, has the following functions to be realized:

(1) Communicate with the single-chip microcomputer that controls the LCD panel and the computer serial port, and obtain the user data of the system application layer;

(2) Select the data frame type to be sent by polling;

(3) Data framing from the application layer to the data link layer;

(4) Carry out FSK modulation to the data of data link layer;

(5) Receive the digital audio signal from the input terminal, add it to the FSK waveform modulated by the accompanying audio signaling after 15KHZ low-pass filtering, and output it to the subsequent equipment through the analog audio interface after CODEC D/A conversion. the

The schematic diagram of the software function module of the sending end is shown in FIG. 10 . the

For CRC check, DSP has adopted a kind of fast CRC check algorithm calculated by byte based on look-up table. The idea of this look-up table algorithm is to first construct a single-byte information remainder coding table off-line, and then perform table look-up and XOR operations based on this coding table to obtain the CRC remainder of multi-byte information. Since the single-byte information has 8 binary code units, there are 256 different combinations in total. Take the most widely used generator polynomial G(x)=x ¹⁶ +x ¹² +x ⁵ +1 suggested by CCITT as an example, each combination is divided by the generator polynomial G(x) to generate a check value of two bytes , so the single-byte remainder code table accounts for 256 words (512 bytes) in total.

The flow chart of the fast algorithm for CRC check is shown in Figure 11. the

For the FSK modulation of data link layer data, the method of table lookup and keying is adopted. This method has three advantages: (1) fast (2) can ensure phase continuity (3) easy to implement. The schematic diagram of its algorithm is shown in Figure 12. the

The receiving end of the system, that is, the digital FM exciter end, mainly has the following functions:

(1) Perform 15KHZ low-pass filtering on the received digital audio signal for code modulation of the exciter. the

(2) Perform 15KHZ high-pass filtering on the received digital audio signal to obtain the FSK modulated wave of the signaling. the

(3) Correlating the obtained FSK modulated wave with the frame header to simplify the demodulation of FSK. the

(4) Carry out CRC check and frame analysis on the obtained signaling data to obtain correct network management information, and calculate the audio delay by combining the 1PPS standard time scale signal of GPS. the

The schematic diagram of the software function module of the receiving end is shown in Fig. 13 . the

filter design

In this system, digital filters are used to filter digital signals in DSP. There are two main types of digital filters: finite impulse response (FIR) filters and infinite impulse response (IIR) filters. The FIR filter has a finite-length impulse response and is a filter with ideal linear phase and constant group delay. The advantages of the FIR filter are summarized as follows:

●Can provide ideal linear phase response. the

●It can be implemented with a very simple algorithm. the

• Reduced algorithm sensitivity to limited arithmetic precision and round-off errors. the

●The stability is guaranteed. the

The ideal linear phase of an FIR filter means that the phase response of the filter decays linearly with frequency (ie, the rate of phase change is constant). Constant group delay is of particular value for applications that require preservation of signal waveform shape, such as high-speed data transmission or high-fidelity audio or video applications. Our calculation of audio delay is based on this point, so the filter design here is realized by FIR digital filter. the

Window function design method is the basic design method of FIR filter, which has been widely used. In this embodiment, a Kaiser window can be used when designing a filter using the window function design method. Because the Kaiser window is one of the most useful and optimal window structures. For a given stopband attenuation, it provides the smallest main lobe width, that is, the steepest transition band, and in this respect, it is optimal. In the actual implementation, the filter parameters N and β are calculated according to the empirical formula of the Kaiser window, and then the desired filter coefficient vector is obtained through the firl function provided by MATLAB, and then called in the DSP program in the form of a coefficient table. the

Sampling point synchronization

Sampling point synchronization is a synchronization concept proposed for this system. Sampling point synchronization requires that the bit synchronization timing error recovered during demodulation does not exceed an audio sampling point interval, that is, the bit synchronization timing can be accurate to the sampling point. Only in this way can the accuracy requirement of audio transmission delay calculation be met. The realization of sampling point synchronization is particularly important, which is the basis of subsequent series of calculations. the

FSK demodulation

FSK demodulation uses a correlation and square law decision device to complete sampling point synchronization and frame synchronization. The method is shown in Figure 14. the

The correlation detection algorithm of FSK demodulates by comparing the maximum value of the square law output by the correlator. Correlate the obtained FSK modulated wave with the in-phase component and quadrature component of the frame header data to obtain two sets of correlation calculation results, perform square calculation on the two sets of values point by point, add the results, and save the calculation results . At the same time, the maximum value is calculated in the result, and the corresponding point of the maximum value found within one frame period is the frame synchronization point, so as to restore the precise relative position relationship of the sampling points of the entire transmission frame, and realize sampling point synchronization, bit synchronization and frame synchronization . When the maximum output position is the same for three consecutive calculations, we believe that the system enters the state of accurate synchronization of sampling points. By adopting the demodulation algorithm mentioned above, bit synchronization, frame synchronization and sampling point synchronization can be realized at the same time. the

After synchronization is achieved, the transmission frame data is demodulated, and the demodulation method is shown in Figure 15. After demodulating and obtaining the entire transmission frame data, perform CRC check. If the CRC check result is correct, save the demodulated data and output it after analysis. If the CRC check result is incorrect, the data is discarded. the

This application proposes a method for adding and extracting channel-associated audio signaling in a digital synchronous FM broadcasting system. Through the analysis of possible audio transmission methods used in FM synchronous broadcasting network, a general method for adding channel-associated audio signaling to analog audio signals is proposed. This method firstly modulates the channel-associated audio signal to the audio high-end frequency through FSK modulation, and then sends it to the transmission link together with frequency division multiplexing of the audio signal. The receiving end only needs to perform high-pass filtering and FSK demodulation to obtain the channel-associated audio signaling added by the sending end. This method for adding channel-associated audio signaling is applicable to any audio transmission system with a transmission bandwidth of 20 KHz, and has the widest applicability. Moreover, this method does not need to add any additional equipment, which brings great convenience to the user in the actual networking process. the

The architecture design of the channel-associated audio transmission system with frequency division multiplexing has fully considered the impact of the MPEG-2 encoder in the most commonly used DVB transmission system on the audio frequency components, that is, the channel-associated audio signaling. In view of these influences, various parameters and strategies used in the signaling transmission process are specified in detail. the

The above-described embodiments are only preferred specific implementations of the present invention, and ordinary changes and replacements performed by those skilled in the art within the scope of the technical solution of the present invention should be included in the protection scope of the present invention. the

Claims (3)

1. A method for solving FM synchronous broadcasting system synchronous audio delay calculation and adjustment, its transmission bandwidth is the audio transmission system of 20KHz, it is characterized in that, the digital FM exciter of described system performs the following steps: Step A: Perform low-pass filtering on the received digital audio signal for code modulation by the digital FM exciter; perform high-pass filtering on the received digital audio signal to obtain the FSK modulated wave of the signaling; Step B: The FSK correlation detection algorithm demodulates by comparing the maximum value of the square law output by the correlator; Correlate the obtained FSK modulated wave with the in-phase component and quadrature component of the frame header data respectively to obtain two Group correlation calculation results, square the two groups of values point by point, and add the results to save the calculation results; at the same time, find the maximum value in the result, and find the corresponding point of the maximum value in one frame period is the frame synchronization point, so as to recover the precise relative position relationship of sampling points of the entire transmission frame, and realize sampling point synchronization, bit synchronization and frame synchronization; Step C: Perform CRC check and frame analysis on the obtained signaling data to obtain correct network management information, and calculate the audio delay by combining the 1PPS standard time scale signal of GPS. the 2. The method for calculating and adjusting the synchronous audio time delay of the FM synchronous broadcasting system as claimed in claim 1, wherein in step B, the frame header synchronization is realized by utilizing the modulated wave and the frame header to carry out correlation operations, and sampling Click Sync. the 3. the method for solving FM synchronous broadcasting system synchronous audio time delay calculation and adjustment as claimed in claim 1 is characterized in that, the system clocks at both ends of the transceiver are all phase-locked with the 10MHz reference frequency standard provided by GPS; every once in a while, The starting position of demodulation is fine-tuned. the

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