Coded data generation or conversion – Digital code to digital code converters
Reexamination Certificate
1999-07-09
2001-08-28
Tokar, Michael (Department: 2819)
Coded data generation or conversion
Digital code to digital code converters
C348S738000, C381S002000
Reexamination Certificate
active
06281813
ABSTRACT:
BACKGROUND OF THE INVENTION
The invention relates to a method and a circuit for decoding a Sound Intercarrier Frequency (SIF) signal using the frequency (SIF) BTSC standard.
In the United States, stereo sound in television and video is transmitted and processed using the BTSC standard established by the Broadcast Television System Committee (BTSC). This standard is also known as Multichannel Television Sound, (MTS).
According to the BTSC standard, the sum R+L and the difference L−R of the signal R of the right stereo channel and the signal L of the left stereo channel are formed at the transmitter. The sum signal and the difference signal are further processed and encoded into separate branches so that an encoded sum signal and an encoded difference signal are formed that are decoded again at the receiver in order to obtain a stereo audio signal.
The sum signal R+L, the amplitude-modulated difference signal L−R, a pilot signal, the frequency-modulated Second Audio Program, (SAP) and the frequency-modulated professional channel are combined into the MTS signal
At the transmitter, a BTSC encoder encodes a multiplex signal that has been compressed using the DBX noise suppression described in U.S. Pat. No. 4,539,526.
According to the BTSC standard, the difference signal L−R is amplitude-modulated with twice the video line frequency. Because of the parabolic noise distribution in the transmission of frequency-modulated signals, the noise level in the transmission channel on which the difference signal L−R is transmitted is approximately 15 dB higher than in the transmission channel on which the sum signal L+R is transmitted, although the difference signal L−R is modulated by 6 dB more than the sum signal. For this reason, a dynamic noise suppression system is provided for the channel to transmit the difference signal L−R, (e.g., the above mentioned DBX noise suppression system). However, because a DBX noise suppression system is provided in only one channel, (i.e., the channel for transmission of the difference signal), it must meet strict requirements regarding accuracy and scaling. Even small inaccuracies result in a deteriorated separation of the stereo sound. In order to avoid these disadvantages, a noise suppression system of this kind would be required in both channels, because the errors in dematricization are compensated and therefore act only on the frequency response. Because of the high expense, however, a second noise suppression system is not used.
The basic principle of the DBX noise suppression system is the so-called masking. If a signal level in one spectral range is much greater than the noise level in this range, a listener will not notice the noise in this range. In order to make the noise inaudible, the DBX noise suppression system encodes the signal so that the signal level of the transmitted signal is much greater than the noise level of the transmission channel. In the DBX noise suppression system, the audio signal to be transmitted in various spectral ranges is compressed so that the level of the transmitted signal is sufficiently high relative to the noise level and the signal energy is distributed as uniformly as possible over the entire frequency range of the signal.
In order to be able to meet both requirements, a DBX encoder described in the above mentioned U.S. Pat. No. 4,539,526 is equipped with first and second compression stages.
In the first compression stage, a variable pre-emphasis filter amplifies the high-frequency signal component as a function of the energy in this spectrum in order thus to achieve a distribution of the signal energy over the entire frequency spectrum that is as uniform as possible. The variable pre-emphasis filter is controlled by a control variable obtained by feeding its output signal back through a high-pass.
In the second stage, the level of the signal is raised or lowered as a function of the total energy by a compressor, which includes a controllable amplifier whose control signal is derived from the output signal of the variable pre-emphasis filter by a bandpass.
The control signal for the variable pre-emphasis filter is derived from the high-frequency components, while the control signal for the controllable amplifier is derived from the entire spectrum.
According to the BTSC standard, at the transmitter, a carrier with a frequency of 4.5 MHz is frequency-modulated with the MTS signal, which must be adjusted precisely to a specific level that always relates to a certain frequency modulation deviation. The modulated MTS signal is the Sound Intercarrier Frequency (SIF) signal. Thus for example it is defined that the bandpass of the encoder has an amplification of 0 dB when a signal with a frequency of 300 Hz is applied at the input which has modulated the carrier with a deviation of 4.495 kHz. Known analog frequency modulation demodulators supply an output voltage that is proportional to the frequency of the frequency-modulated input signal. However, the voltage of the frequency modulation demodulator is also proportionally dependent on an unknown factor which varies as a result of internal component tolerances and is subject to drift as a result of temperature changes and component aging. Since no known relationship exists for this reason between the frequency modulation deviation and the output voltage of the frequency modulation demodulator, the input level for each module must be very accurately adjusted by a potentiometer for example. As a result of the above mentioned drift caused by component aging, an adjustment must be made after a few years in order to achieve good stereo reproduction once again.
SUMMARY OF THE INVENTION
Hence the goal of the invention is to design a method and a circuit for decoding an analog audio signal according to the BTSC standard in such fashion that all the adjustments, both adjustment during manufacture and also subsequent adjustment caused by drift due to component aging, become superfluous.
The invention achieves this goal according to the method by the fact that initially a digital SIF signal is provided in a first step, is demodulated in a second step to produce the digital MTS signal, and that the digital MTS signal is decoded in a third step using a digital BTSC decoder.
The invention achieves this goal devicewise by the fact that a digital SIF signal is applied at the input of a digital frequency modulation demodulator whose output is connected with the input of a digital BTSC decoder, at whose outputs the sum signal R+L and the difference signal L−R can be obtained.
REFERENCES:
patent: 4472830 (1984-09-01), Nagai
patent: 5357284 (1994-10-01), Todd
patent: 363296418 (1988-12-01), None
Hilpert Thomas
Mueller Stefan
Noeske Carsten
Vierthaler Matthias
Winterer Martin
Chang Daniel D.
Micronas GmbH
Samuels , Gauthier & Stevens, LLP
Tokar Michael
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