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ENCYCLOPEDIA OF RADIO ELECTRONICS AND ELECTRICAL ENGINEERING We accept stereophonic sound. Encyclopedia of radio electronics and electrical engineering
Encyclopedia of radio electronics and electrical engineering / Телевидение On November 14, 2003, Channel One of Russian television began to regularly broadcast a number of programs with stereo sound accompaniment. They are marked on the image with a special icon in the form of two stylized TV screens superimposed one on top of the other. Of course, the transmission of a monophonic sound signal has also been preserved. Such broadcasting became possible in connection with the commissioning of a new transmitter at the Ostankino television tower to replace the old one, which had been operating since 1967. - from the date of the start of broadcasting from the television center in Ostankino. The old transmitter will be used as a backup for now. Residents of Moscow and the Moscow region can receive stereo sound if their TVs are equipped with demodulators - decoders of the NICAM signal transmitted by the DQPSK phase modulation method at a subcarrier frequency of 5,85 MHz. Recall that the separation between the carrier frequencies of the image and the usual monophonic sound in radio channels is 6,5 MHz, as provided for in our standards D (for MB) and K (for UHF). How the NICAM stereo sound signal is formed, transmitted and received is described in this and subsequent parts of the published material. Until recently, stereophonic sound accompaniment of on-air television programs was not carried out in our country, so there was little interest in such broadcasting systems. At the same time, they are successfully operated abroad. One of the most popular among them is the NICAM (Near Instantaneously Companded Audio Multiplex) stereo sound system for television broadcasting. It was developed by the British Broadcasting Company BBC (BBC) and first presented to the CCIR in 1987. It entered service in 1988 and is now widely used in the UK, Sweden, Denmark and other European countries both in terrestrial and and in satellite television broadcasting. Glossary of terms
Since the "First Channel" of television broadcasting began to conduct stereo sound accompaniment of a number of its programs using this particular system, the reader should be familiarized with the principles of NICAM signal generation, its transmission and reception according to radio frequency standards B, G, H, I, as well as specific schemes television receiver signal decoders. Since the system provides transmission with a total speed of 728 kbps, in the literature it is often called NICAM-728 [1-4]. In accordance with CCIR Recommendation 707, the system is used in cases where terrestrial television devices, together with the transmission of analog video signals, additionally require the introduction of digital audio. For its transmission, two carrier frequencies are used (Fig. 1), the main of which f3 ocn is modulated, as usual, in frequency by an analog monophonic sound signal for television programs, and the additional f3 additional is modulated by a NICAM digital stereophonic sound signal.
The audio carriers are 5,5 MHz (primary) and 5,85 MHz (secondary) MHz for the B, G, H standards and 6 and 6,552 MHz for the I standard from the image carriers f. L (left) and R (right) channel signals. The NICAM sound carrier in the B, G, H, I standards is located in frequency slightly higher than the normal sound carrier, but within the frequency band of the radio channel. The main parameters of the NICAM system are shown in the table.
Let us consider the principle of signal formation of the NICAM system according to the simplified structural diagram of the transmitter shown in Fig. 2. Before applying analog audio signals from the L and R channels to the multiplexed ADC, pre-emphasis is introduced into each of them. They are required by international standards (CCITT Recommendation J.17) to provide some boost to the RF components of the signals. Pre-emphasis allows you to reduce the level of noise, which are located mainly in this interval. In the receiver, the ratio of LF and HF components is restored by pre-distortion correction circuits, which reduce the amplitude of the HF components.
It is known that to obtain high-quality sound from home equipment, an audio frequency band of 15 kHz is sufficient. It follows that the minimum sampling (sampling) frequency when converting an analog audio signal to digital should be equal to twice the upper audio frequency, i.e. 30 kHz. However, in practice, to prevent signal aliasing and related distortion, a slightly higher sampling rate of 32 kHz is used. Sampling in the L and R signals occurs simultaneously, after which the ADC converts a group of three L signal samples into a 14-bit coded word, followed by the same group of R signal samples, then again the L word, and so on in turn. The ADC output signal consists of successive data segments, which are groups of 32 samples of each channel. 14-bit digitization of signals allows you to get a large number of quantization levels (16384), which is quite acceptable for high-quality sound reproduction. Under the mentioned conditions for digitizing signals with a sampling rate of 32 kHz, a rather high data rate is required and, consequently, a very wide frequency band, which does not fit into the frequency band of the radio channel. Therefore, in practice, almost instantaneous digital companding is used (as the name of the system indicates), which allows you to reduce the number of bits per sample from 14 to 10 and the bit rate without degrading the quality of the reproduced signal. The method of digital companding is based on the fact that the value of each bit of the binary code depends on the level of the sound signal, which at each moment represents a specific coded sample. So, with loud sounds, i.e., with large signal amplitudes, the influence of the least significant bits is very small and they can be neglected. With quiet sounds (counting values do not exceed 100 ... 200 μV), the least significant bits cannot be neglected. Therefore, the NICAM digital compander turns the 14-bit code into a 10-bit code: for weak signals, the original 14-bit samples are retained, and for high-level signals, one to four least significant bits are discarded. For more efficient companding, some higher bits are also excluded in some cases. For example, the 13th bit will be excluded if it matches the 14th; The 12th bit - if it matches both the 13th and the 14th, etc. The 14th bit is always present, since it indicates the polarity of the signal. When the most significant bits are removed, the system provides a way to restore them at the receiver, called coding with a scale factor. It is a three-bit code that tells the receiver the number of excluded high bits for their subsequent recovery. The next stage of signal processing is to add a parity bit to the code of each sample and form an 11-bit code. The parity bit is needed to check the six most significant bits for the presence of an error in them. At the output of the device for adding parity bits from 32 11-bit samples L1 - L32 (in the L channel) and R1 - R32 (in the R channel), groups are formed, called segments (Fig. 3), which first arrive at the block shaper, and then - to the loop-forming multiplexer. Before the formation of cycles (frames, frames), the data stream is organized into 704-bit data blocks, each of which contains two segments (one from each channel), and the blocks are multiplexed as shown in Fig. 4.
Each block of audio data is preceded by an additional 24 bits of information necessary for synchronization and control (Fig. 5). The framing word synchronizes the NICAM receiver of the TV and is always 01001110, and bits C0-C4 are needed to control and synchronize the decoder, with the CO bit called the frame flag.
Next, bitwise interleaving is applied. It is required to minimize bit errors (bursts of errors), which are caused by noise and interference and can distort several neighboring bits. The bit interleaver separates adjacent bits from each other by 16 clock cycles (i.e., there are 15 other bits between them). Therefore, since a packet of errors usually does not exceed 16 bits (and this is most likely), on a TV it will be dispersed over various samples in the form of single bit errors, and this practically does not affect the sound quality. The bit interleaver contains a RAM where data of a 704-bit block is first written to and then read from it in the above sequence. The reading order is stored in ROM, otherwise called the address sequence sensor. A similar ROM was used in the TV to restore the original bit sequence there.
In order for the signal to be perceived as random, i.e., to have a uniform distribution of energy, and to reduce the influence of the normal audio signal from the frequency modulator on the NICAM audio signal, the bit stream is passed to the scrambling device. Obviously, the bits of the framing word are not scrambling. The TV performs the reverse procedure, called descrambling the audio data bits, to restore them to their original form. In the NICAM system, the QPSK (Quadrature Phase Shift Keying) sound carrier phase keying method is used to transmit a digital signal over a radio channel. However, the scrambled digital audio data stream is differentially encoded before being fed to the modulator, so the keying is also called Differential (DQPSK). This is necessary so that the TV could use not only synchronous demodulation, but also a simpler one - difference. Phase shift keying is the most economical form of modulation in which the frequency of the carrier remains constant while its phase changes according to the state of the data bits. Quadrature phase shift keying, also called quad-position keying, has four phase values: 45°, 135°, 225°, and 315°. To obtain them, the carrier phase is first shifted by 90° and two quadratured data signals are formed: I and Q. As a result, a signal is created with a resulting phase of 45°. Then, to form the remaining resulting vectors, these two signals are subjected to a phase change of 180° (Fig. 6). Each of the vectors can be represented by two bits of a binary number:
Therefore, the presented bit patterns change the phase of the carrier by different angles with respect to the phase of the previous signal, as shown in the timing diagram of Fig. 7. To provide such phase manipulation, conversion of a serial digital audio data stream into a parallel two-bit format is provided. As a result, the bit rate is reduced by half, which leads to a narrowing of the bandwidth occupied by the signal.
The modulated DQPSK signal and the FM mono signal are sent to the frequency converter, where they are transferred to a given carrier frequency. The RF signal is amplified and radiated by the antenna. Consider a fragment of the block diagram of a TV set with a built-in NICAM demodulator and decoder (Fig. 8).
As usual, the broadcast television signal is fed to the antenna input of the channel selector (tuner), in which the selection and conversion of the received radio frequency signals into IF image and sound signals take place. Reinforced and passed through the SAW filter, they pass into the corresponding processing paths of the TV. The NICAM band pass filter (at 5,85 MHz for B, G, H, D, K or 6,552 MHz for I) separates the NICAM IF signals, which are amplified and fed to the NICAM demodulator (Figure 9). Its operation is based on the same principles as a conventional FM demodulator, in which changes in the phase or oscillation frequency lead to a change in the output DC voltage. However, with quadrature modulation, in addition to the in-phase phase detector, a quadrature phase demodulator is also used, to which a 90 ° phase-shifted signal from a carrier generator is applied.
From the detector and demodulator outputs, the I and Q data signals pass through the low-pass filter to the differential logic decoder, the clock bit recovery device, and the PLL. The latter, as usual, if necessary, generates an error signal that adjusts the frequency and phase of the carrier generator. The sync bit restorer enters the second PLL locked to the bit rate. To ensure bit rate synchronization, a multiple of the bit rate is used as the system frequency. The bit rate is obtained by dividing the system clock frequency by 8. The differential logic decoder converts the I and Q data streams into the corresponding two-bit parallel data, which is then passed to the parallel-to-serial converter, which restores the original serial data stream. The NICAM decoder (Figure 10) provides descrambling, deinterleaving, data expansion, original 14-bit word recovery, and DAC control.
The encoded data from the NICAM demodulator is fed to a frame alignment word detector and a descrambler for frame detection and descrambling. The descrambled data arrives at the deinterleaver, which outputs the original two-channel (L and R) data along with the desired channel identification signal. For deinterleaving, by analogy with the transmitter, first, the data stream is written to the ROM cells block by block, and then, to reproduce the correct bit order, the contents of the cells are read in accordance with the program recorded in the ROM. The descrambled data also passes to the operating mode selector, which decodes the control bits C0-C4 (see Fig. 5) and transmits information about the type of transmission to the expander and other nodes of the decoder, as well as the TV. In it, in particular, a mono sound channel blocking signal is generated when a stereo sound is received. This blocking prevents interference and noise from the mono channel from entering the 3H amplifier. Restored in the correct order by the deinterleaver, each 11-bit word (recall: 10 data bits + 1 parity bit) is expanded by the expander to a 14-bit format. The expander uses scale factors embedded in the parity bits, which expand the 10-bit sample codes to 14 bits. The error checker uses parity bits to correct the bit stream. The data is then corrected for pre-emphasis and fed to the DAC control unit, which generates three signals: a data bit stream, an identification signal, and a clock signal. Usually, one DAC is used, which works alternately on the code words of the L and R signals. Analogue 3-hour signals are formed at the DAC outputs, which are fed to the corresponding power amplifiers. Let us now consider the schematic diagram of the NICAM receiver (board K) of the PHILIPS TV - 29PT-910V / 42 (58), assembled on the chassis FL2.24, FL2.26 or FL4.27 (AA) (Fig. 11). The receiver is designed in such a way that it can process signals of both standards B, G, H, and standard I.
The NICAM IF signal is applied to the board input pins 1N43 and 1N50 (IF INPUT). Two bandpass filters 1002 and 1004, connected in parallel, ensure the separation of the signals of the mentioned standards. The cascade on transistor 7008 plays the role of an emitter follower, and on transistor 7009 - an IF signal amplifier. Next, the NICAM (DQPSK) signal is applied to pin 3 of the 7000 chip, which acts as a demodulator of the NICAM audio spectrum components. It also includes the restoration of time intervals (bits) of the digital code, the conversion of the parallel code of the data signal into a serial one, and the phase locked loop of the frequency of the double carrier generator. The block diagram of the TDA8732 chip is shown in fig. 12. Through the limiting amplifier inside the microcircuit, the signal arrives at the in-phase phase detector and the quadrature demodulator. One of them received a subcarrier signal without changing the phase, and the other - shifted by 90°.
The I and Q signals generated at the outputs of these devices through pins 7 and 6 of the microcircuit, low-pass filter (choke 5001, capacitor 2005 and choke 5000, capacitor 2004 in Fig. 11), pins 8 and 5 of the microcircuit pass to the differential logic decoder (Fig. 12) , a clock bit recovery device, and a PLL device. The first one converts the I and Q signals received in parallel to two-bit digital data, and the data converter included later restores them to the original serial stream. At the output of the CLK LPF bit recovery device (pin 1 of the microcircuit), a low-pass filter (capacitors 2042, 2012, 2014, resistors 3011, Z010) and a varicap 6006 are turned on (see Fig. 11). Under the influence of the voltage level formed at pin 1 of the microcircuit, the capacitance of the varicap changes, as a result of which the quartz resonator 1001 is automatically adjusted. This ensures synchronization of the frame synchronization word detector located in the 7001 microcircuit. A low-pass filter (capacitors 9, 7000, resistor 2006) and a 2007 varicap are connected to the output of the PLL device (pin 3005 of the 6005 microcircuit). , and a double carrier frequency generator (Fig. 9). This is how the system synchronization of the demodulator devices takes place. The 7000's data converter is clocked by external PCLK clocks applied to the timer clock via pin 16 of the IC (see Figure 11) from the 7001's internal oscillator. The serial data stream DATA from pin 15 of the 7000 passes through pin 21 of the 7001 (Figure 13) to the framing word detector and descrambler. The operation of most devices of the SAA7280 chip coincides with that already described in Fig. 10 in the previous part of the article does not require comments.
It is only necessary to add that from the operating mode selector, through pin 22 of the microcircuit (see Fig. 11), the control voltage is supplied to the audio signal switch and ensures that the channel of ordinary monophonic sound is blocked when stereophonic is received. The remaining outputs of the operating mode selector (see Fig. 11 and 13) are not used in this particular TV. The devices of the 7001 microcircuit are controlled by signals from the 1C digital bus, therefore, an interface for this bus is provided inside the microcircuit (Fig. 13). SCL clock signals are applied to it through pin 26 of the microcircuit (see Fig. 11), resistor 3027 and pin 4N43 of the board, and SDA data signals are received and removed through pin 24 of the chip, resistor 3026 and pin 5N43 of the board. From the control device of the DAC of the 7001 microcircuit (Fig. 13), through pins 10, 8 and 9, the digital signals of SDAT data, SCLK synchronization and STIM identification, respectively, pass to pins 3, 2 and 1 of the 7007 microcircuit (TDA1543), which acts as a DAC. At its outputs (pins 6 and 8), stereo audio signals of the left (L) and right (R) channels are generated, fed to the 3H amplifier.
Figure 14 shows a fragment of the circuit diagram of the sound board (AUDIO) of SAMSUNG - CS6277PF / PT TVs assembled on the SCT51 A chassis. It should be noted that in the demodulator-decoder all fixed resistors, except for RJ08, RJ11, and all non-polar capacitors are for surface mounting (CHIP). The NICAM signal processing channel in TVs is built on one LSI ICJ01 (SAA7283ZP), which performs the functions of a DQPSK signal demodulator, a demodulated signal decoder and a DAC (Fig. 15).
The quadrature (phase) modulated DQPSK NICAM signal through the SIF (QPSK) pin of the CN601 connector (see Fig. 14) of the sound card and pin 29 of the microcircuit (Fig. 15) enters the band-pass filters built into it (5,85 and 6,552 MHz) and an amplifier covered by AGC and controlled by an internal AGC controller. The DQPSK signal is detected by a phase detector with carrier loops, on which (depending on the received standard) an error voltage is emitted, which is then converted by the VCO into a control voltage (in our case, at pin 27, see Fig. 14). It also affects the contour adjustment circuit. The generated I and Q signals come (see Fig. 15) to the synchronization bit recovery device, which, through pins 39 and 40 of the microcircuit, acts on the crystal oscillator. The NICAM decoder descrambles, deinterleaves, and expands the data signals. The decoded data after the digital filter is amplified, passes the pre-distortion correction device and is converted by the built-in DAC chip into analog audio signals of the L and R channels. The L and R signals passed through the output switches from pins 15 and 8 of the microcircuit, respectively, are fed to the 3H amplifier. Other audio signals can also be applied to the output switches, such as a mono signal of normal audio in the absence of stereo accompaniment. In the module under consideration, a monophonic sound signal comes through pins 7 and 16 of the microcircuit, capacitors CJ28 and CJ23 and the SECAM-L pin of the CN601 connector. All nodes of the microcircuit are controlled by a controller combined with a NICAM decoder and ROM. Control is provided by digital bus l2C. To do this, pin 49 of the microcircuit receives the SCL clock signal, and pin 50 is supplied and the SDA data signal is removed from it. Literature
Author: A. Peskin, Moscow
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