Professional music equipment. Effect Processor. The screen clearly shows all metering functions of the processing structure in use. If you get lost, ESC will always bring you back home. Step-by-step instructions are on the screen. It can be configured to interface ideally with any commonly-found transmission system in the world.
Its pre-emphasis control is almost never audibly apparent, producing a clean, open sound with subjective brightness matching the original program. Only one processing structure can be active at a time. See page for information on programming the remote control interface. It can run on systems using VGA graphics and can drive a modem or create a direct connection between the computer and the through their RS serial ports.
There are two serial connectors. The other connector meets the RS standard. The audio processing circuitry in OPTIMOD-FM produces a signal that is pre-emphasized to either the 50 s or 75 s standard curve, is precisely and absolutely high frequency-con- trolled and peak-controlled to prevent over-modulation, and is filtered at 15kHz to protect the 19kHz pilot and prevent distortion caused by aliasing-related non-linear crosstalk.
If this signal is fed directly into a stereo encoder, peak modulation levels on the air will be precisely controlled. Peak modulation will increase, but average modulation will not. The modulation level must therefore be reduced to accommodate the larger peaks.
Reduced average modulation level will result in reduced loudness, and a poorer signal-to-noise ratio at the receiver. Frequency response errors and non-constant group delay are typically introduced by land line equalizers, transformers, and 15kHz low-pass filters and pre-emphasis networks in stereo encoders.
There are three criteria for preservation of peak levels through the audio system:. If low-pass filters are present, this may. The deviation from linear phase must not exceed. Low-pass filters including anti-aliasing filters in digital links , high-pass filters, transformers, distribution amplifiers, and long transmission lines can all cause the above criteria to be violated, and must be tested and qualified.
It is clear that the above criteria for optimal control of peak modulation levels are most easily met when the audio processor directly feeds the stereo encoder. In the , no circuit elements that might distort the shape of the waveform are interposed between the audio processor and the stereo encoder.
We therefore recommend using the with its built-in stereo encoder whenever practical. This situation is not ideal because artifacts that cannot be controlled by the audio processor can be introduced by the link to the transmitter, by transmitter peak limiters, or by the external stereo encoder.
All audio processing must be done at the studio, and you must tolerate any damage that occurs later. Transmitter protection limiters should respond only to signals caused by faults or by spurious peaks introduced by imperfections in the link. To ensure maximum quality, all equipment in the signal path after the studio should be carefully aligned and qualified to meet the appropriate standards for bandwidth, distortion, group delay and gain stability, and such equipment should be re-qualified at reasonable intervals.
However, the link should have low noise, the flattest possible frequency response from ,Hz, and low non-linear distortion. You will achieve a louder sound on the air, with better control of peak modulation, than if you use an external stereo encoder.
Its instruction manual contains complete information on its installation and application. If possible, bypass the pre-emphasis network and the input low-pass filters in the encoder so that they cannot introduce spurious peaks. Except for the composite baseband microwave STL and certain digital links, all these links carry the left and right channels directly or in some encoded form other than the standard 19kHz pilot-tone stereo baseband.
These links are normally fed both left and right audio channels in non-encoded form, and their output is in the form of left and right channels. The composite baseband microwave STL carries the standard pilot-tone stereo baseband, and is therefore fed from the output of a stereo encoder located at the studio site. Thus, no stereo encoder is needed at the transmitter.
In general, a composite microwave STL provides the highest audio quality, as long as there is a line-of-sight transmission path from studio to transmitter of less than 10 miles 16 km. If not, RF signal-to-noise ratio, multipath distortion, and diffraction effects can cause serious quality problems. Although a microwave STL exhibits satisfactory stereo separation, it is nevertheless not unusual for it to bounce because of a large infrasonic peak in its frequency response caused by an under-damped automatic frequency control AFC phase-locked loop.
This bounce can increase the peak carrier deviation by as much as 1dB, reducing average modulation. Some consultants presently offer modifications to minimize or eliminate this problem. If your exciter or STL has this problem, you may contact Orban Customer Service for the latest information on such services. There are several types of digital links presently available.
One type encodes the entire composite stereo baseband on the link, functionally replacing a composite microwave STL. Other digital links pass the audio in left and right form, and may apply data-rate-reduction processing to the signal to reduce the number of bits per second required for transmission through the digital link. Such processing may distort peak levels, and such links must therefore be carefully qualified before you use them to carry the peak-controlled output of the to the stereo encoder.
Older-technology links may use straightforward PCM pulse-code modulation without data rate reduction. These can be very transparent and can exhibit accurate pulse response provided that their input anti-aliasing filters and output reconstruction filters are rigorously designed to achieve constant group delay over the frequency range that contains significant program energy.
This is not particularly difficult to do with modern over-sampled converter technology. Older-technology converters usually exhibit rapid changes in group delay around cut-off because their analog filters are ordinarily not group-delay equalized.
Additionally, they may exhibit quantization distortion unless they have been correctly dithered. The installing engineer should be aware of all of these potential problems when designing a transmission system. The quality of signal received at the transmitter through this type of STL is high.
However, the high carrier frequencies make these links subject to rain fading. Other potential problems include very sharp high-frequency cut-off, rapid changes in group delay around cut-off, and quantization distortion. The Sony and dbx encoders are no longer manufactured, but may be found on the used market.
Analog land line quality is extremely variable, ranging from excellent to poor. Whether land lines should be used or not depends upon the quality of the lines locally available, and upon the availability of other alternatives. Even the best land lines tend to slightly veil audio quality, due to line equalizer characteristics and phase shifts. Slight frequency response irregularities and non-constant group delay characteristics will alter the peak-to-average ratio, and will thus reduce the effectiveness of any peak limiting performed prior to their inputs see Optimal Control of Peak Modulation Levels on page However, problems include gainand phase-matching of the left and right channels, overloads induced by pre-emphasis, and requirements that the audio applied to the microwave transmitters be processed to prevent over-modulation of the microwave system.
Unless carefully designed, dual microwave STLs can also introduce non-constant group delay in the audio spectrum, thus distorting peak levels. Many such links have been designed to be easily configured at the factory for composite operation, where the entire FM stereo baseband is passed, including the pilot tone and stereo subchannel.
The requirements for maintaining stereo separation in composite operation are similar to the requirements for high waveform fidelity with low overshoot. Therefore, most links have the potential for excellent waveform fidelity if they are configured for composite operation even if a composite FM stereo signal is not actually being applied to the link. Nevertheless, in a dual-microwave system, the is usually located at the main FM transmitter and is driven by the microwave receivers.
If the STL microwave uses pre-emphasis, its input pre-emphasis network will probably introduce overshoots that will increase peak modulation without any increases in average modulation. This frees the system from political overshoot. The Orban A Co-Operator can be easily configured to produce a pre-emphasized output. If the audio link between the studio and the transmitter is noisy, you can minimize the audibility of this noise by performing the gain-riding function at the studio site.
Gain-riding applied before the link to the transmitter improves the signal-to-noise ratio because average level on the link will be greater, so its noise will have less effect on the overall noise level of the broadcast. The A also contains high-frequency and peak control circuitry to protect the STL transmitters or digital encoders from over-modulation. When you use any studio level controller, defeat the gain-riding function in the main processing. This is covered in the setup information in Section 2 page or If you are using an Orban Transmission Limiter to protect your STL, do not defeat the gain-riding function in the main processing.
The Orban is a transmission system overload protection device; it is normally operated below threshold. Earlier in this section, we discussed at length what is required to prevent STLs from overshooting. There are similar requirements for FM exciters. Nevertheless, in some installations some overshoot is inevitable.
Studio engineers and transmission engineers consider audio levels and their measurements differently, so they typically use different methods of metering to monitor these levels. The VU meter is an average-responding meter measuring the approximate RMS level with a ms rise time and decay time; the VU indication usually under-indicates the true peak level by 8 to 14dB.
The PPM has an attack time of 10ms, slow enough to cause the meter to ignore narrow peaks and under-indicate the true peak level by 5dB or more. The absolute peak-sensing meter or LED indicator shows the true peak level. It has an instantaneous attack time, and a release time slow enough to allow the engineer to easily read the peak level. Figure shows the relative difference between the absolute peak level, and the indications of a VU meter and a PPM.
The studio engineer is primarily concerned with calibrating the equipment to provide the required input level for proper operation of each device, and so that all devices operate with the same input and output levels.
This facilitates patching devices in and out without recalibration. For line-up, the studio engineer uses a calibration tone at a studio standard level, commonly called line-up level, reference level, or operating level. Line-up level is usually at this same maximum meter indication.
The transmission engineer is primarily concerned with the peak level of a program to prevent overloading or over-modulation of the transmission system. This peak overload level is defined differently, system to system.
In AM modulation, it is negative carrier pinch-off. In digital, it is the largest possible digital word. For metering, the transmission engineer uses an oscilloscope, absolute peak-sensing meter, calibrated peak-sensing LED indicator, or a modulation meter.
A modulation meter usually has two components — a semi-peak reading meter like a PPM , and a peak-indicating light which is calibrated to turn on whenever the instantaneous peak modulation exceeds the over-modulation threshold.
Program material produces frequent peaks that will read full-scale due to the absolute instantaneous peak response of the meters. Reference tone should be set to indicate 0dB on these meters. The meter is scaled so that 0dB is calibrated to the highest left and right peak modulation level, before de-emphasis, that the processing will produce, under any program, processing, or setup condition except when the processing is switched to bypass.
The meter indication is not affected by the setting of the output level control. For example, in the U. The frequency and modulation level of the line-up tones can be adjusted from the front panel. A sync reference input that can accept 10 MHz or 1 x wordclock 32, The digital input and output conform to the professional AES3 standard. They both have sample rate converters to allow operation at 32, The level, de-emphasis, and other parameters of these outputs are set in System Setup and are the same regardless of whether the is operating in its audio processor or stand-alone stereo encoder modes.
The left and right analog inputs are on XLR-type female connectors on the rear panel. Input impedance is greater than 10k ; balanced and floating. The left and right analog outputs are on XLR-type male connectors on the rear panel. Output impedance is 50 ; balanced and floating.
The outputs can drive or higher impedances, balanced or unbalanced. Level control of the analog inputs and outputs is accomplished via software control through System Setup see step 3 on page and step 8 on page or through PC Remote. The stereo encoder has two unbalanced analog baseband outputs on two BNC connectors on the rear panel. Independent level control of each output is available via software see step 6 on page The stereo encoder has two unbalanced 60 subcarrier SCA inputs with rearpanel BNC connectors to accept any subcarrier at or above 23 kHz.
The subcarriers are mixed into each composite output and their level is not affected by the composite level control for that output. Subcarrier inputs sum into the composite baseband outputs. Thus both inputs accommodate subcarrier generators with output levels as low as mV p-p. The correct peak level of the stereo program applied to the stereo encoder sometimes depends on the number of subcarriers in use.
Some regulatory authorities require that total baseband peak modulation be maintained within specified limits. You define the amount of reduction in percent using the procedure in step 21 on page 2- See page for information on programming the remote control.
A jumper J6 on the circuit board can reconfigure the SCA 2 input to provide the stereo pilot tone only, which can provide a pilot reference for an RDS subcarrier generator. They can control various functions of the Switch the stereo encoder to stereo, mono from left, mono from right, or mono from sum audio input. Independently activate and defeat the diversity delay applied to the analog, digital, and composite outputs. Reduce the stereo main and subchannel modulation to compensate for transmitter overshoot and subcarrier inputs SCAs.
The remote control of overshoot compensation and SCA modulation see page is not latching. You must supply a continuous current to the programmed remote input to hold the gain at its compensated level. The tally outputs can be programmed to indicate the following:. Input: Analog: Indicates that the is processing audio from its analog input. Analog Input Silent: Indicates that the level at either or both analog input channels is below the threshold set in step A on page AES Input Silent: Indicates that the level at either or both digital input channels is below the threshold set in step in step A on page When the chip detects such an error, it automatically switches the input to Analog.
You can reconfigure the functions of the inputs and outputs via System Setup. For example, if you are not using the stereo encoder, the three inputs ordinarily dedicated to controlling the state of the stereo encoder can instead be re-configured to call three additional presets. These computer interfaces support remote control and metering, and allow downloading software upgrades.
See Networking and Remote Control on page for more information. It accepts a 1x 5V p-p squarewave wordclock signal at 32, You can configure the to lock its 19 kHz pilot tone and output sample frequency to this input. If the output frequency is different, the output sample frequency will be the product of a quotient of integers times the reference frequency.
If the reference frequency is 48 kHz and the output frequency is set to It is precisely and absolutely high-frequency-controlled and peak-controlled to prevent overmodulation, and is filtered at 15 kHz to protect the 19 kHz pilot and prevent distortion caused by aliasing-related non-linear crosstalk.
If this signal is fed directly into a stereo encoder, peak modulation levels on the air will be precisely controlled. Peak modulation will increase, but average modulation will not. The modulation level must therefore be reduced to accommodate the larger peaks.
Reduced average modulation level will cause reduced loudness and a poorer signal-to- noise ratio at the receiver. Landline equalizers, transformers, and 15 kHz low-pass filters and pre-emphasis networks in stereo encoders typically introduce frequency response errors and nonconstant group delay. There are three criteria for preservation of peak levels through the audio system:.
If low-pass filters are present,. The deviation from linear-phase must not exceed 1 from ,Hz. An all-pole deemphasis network like the classic series resistor feeding a grounded capacitor is not appropriate. However, this network could be corrected by adding a second resistor between ground and the capacitor, which would introduce a zero. Low-pass filters including anti-aliasing filters in digital links , high-pass filters, transformers, distribution amplifiers, and long transmission lines can all cause the above criteria to be violated, and must be tested and qualified.
It is clear that the above criteria for optimal control of peak modulation levels are most easily met when the audio processor directly feeds the stereo encoder. In the , no circuit elements that might distort the shape of the waveform are interposed between the audio processor and the stereo encoder.
We therefore recommend using the with its built-in stereo encoder whenever practical. This situation is not ideal because artifacts that cannot be controlled by the audio processor can be introduced by the link to the transmitter, by transmitter peak limiters, or by the external stereo encoder. All audio processing must be done at the studio and you must tolerate any damage that occurs later.
However, if the digital link employs lossy compression, it will disturb peak levels. Transmitter protection limiters should re-. To ensure maximum quality, all equipment in the signal path after the studio should be carefully aligned and qualified to meet the appropriate standards for bandwidth, distortion, group delay and gain stability, and such equipment should be re-qualified at reasonable intervals. You can achieve the most accurate control of modulation peaks by locating OPTIMOD-FM at the transmitter site and then using its stereo encoder to drive the transmitter.
However, the link should have low noise, the flattest possible frequency response from ,Hz, and low non-linear distortion. See Figure on page This separation is comfortably above the separation approximately 20 dB that starts to cause perceptible changes in the stereo image. You will achieve a louder sound on the air, with better control of peak modulation, than if you use most external stereo encoders.
Adkins and Robert D. Audio Engineering Society , vol. Subjects listened to Hz tones, broadband noise, and stereophonic program material through earphones and adjusted the channel separation, via a manual control, until the degradation of the spatial effect became detectable. Mean channel separations ranged from 10 to The results are discussed in terms of existing data on detectable stereo separation and on the discrimination of interaural time differences.
Its instruction manual contains complete information on its installation and application. If possible, bypass the preemphasis network and the input low-pass filters in the encoder so that they cannot introduce spurious peaks.
STLs are used in three fundamentally different ways. The three applications have different performance requirements.
In general, a link that passes unprocessed audio should have very low noise and low non-linear distortion, but its transient response is not important. A link that passes processed audio does not need as low a noise floor as a link passing unprocessed audio. However, its transient response is critical. We will elaborate below. Digital links may pass audio as straightforward PCM encoding, or they may apply lossy data reduction processing to the signal to reduce the number of bits per second required for transmission through the digital link.
Such processing will almost invariably distort peak levels, and such links must therefore be carefully qualified before you use them to carry the peak-controlled output of the to the transmitter. While the desired program material may psychoacoustically mask this noise, it is nevertheless large enough to affect peak levels severely.
For any lossy compression system the higher the data rate, the less the peak levels will be corrupted by added noise, so use the highest data rate practical in your system. Because the uses multiband limiting, it can dynamically change the frequency response of the channel. This can violate the psychoacoustic masking assumptions made in designing the lossy data reduction algorithm. This is also true of any lossy data reduction applied in the studio such as hard disk digital delivery systems.
Some links may use straightforward PCM pulse-code modulation without lossy data reduction. If you connect to these through an AES3 digital interface, these can be very transparent provided they do not truncate the digital words produced by the devices driving their inputs.
Currently available sample rate converters use phase-linear filters which have constant group delay at all frequencies. If they do not remove spectral energy from the original signal, the sample rate conversion, whether upward or downward, will not add overshoot to the signal.
This is not true of systems that are not strictly bandlimited to 15 kHz, where downward sample rate conversion will remove spectral energy and will therefore introduce overshoot. It uses a block-companded floating-point representation of the signal with J. Older technology converters including some older NICAM encoders may exhibit quantization distortion unless they have been correctly dithered. Additionally, they can exhibit rapid changes in group delay around cut-off because their analog filters are ordinarily not group-delay equalized.
The installing engineer should be aware of all of these potential problems when designing a transmission system. The digital input and output accommodate sample rates of 32 kHz, The composite baseband microwave STL carries the standard pilot-tone stereo baseband, and therefore receives the output of a stereo encoder located at the studio site. The receiver output of the composite STL is the stereo baseband signal, which is. Thus, no stereo encoder is needed at the transmitter. In general, a composite microwave STL provides good audio quality, as long as there is a line-of-sight transmission path from studio to transmitter of less than 10 miles 16 km.
If not, RF signal-to-noise ratio, multipath distortion, and diffraction effects can cause serious quality problems. Uncompressed digital composite baseband microwave STLs, if properly designed, have excellent performance and we recommend them highly. However, the fact that they are digital does not eliminate the requirement that they have low frequency response that is less than 3 dB down at 0.
Any such STL should be qualified to ensure that it meets this specification. Dual microwave STLs use two separate transmitters and receivers to pass the left and right channels in discrete form.
However, problems include gainand phasematching of the left and right channels, overloads induced by pre-emphasis, and requirements that the audio applied to the microwave transmitters be processed to prevent over-modulation of the microwave system. Lack of transparency in the path will cause overshoot. Unless carefully designed, dual microwave STLs can introduce non-constant group delay in the audio spectrum, distorting peak levels when used to pass processed audio. The can only be located at the transmitter if the signal-to-noise ratio of the STL is good enough to pass unprocessed audio.
Of these, the and are currently manufactured as of this writing and are the preferred choices because their AGCs are identical to the AGC in the If the is located at the transmitter and fed unprocessed audio from a microwave STL, it may be useful to use a companding-type noise reduction system like dbx Type 2 or Dolby SR around the link.
This will minimize any audible noise buildup caused by compression within the Many such links have been designed to be easily configured at the factory for composite operation, where an entire FM stereo baseband is passed.
The requirements for maintaining stereo separation in composite operation are similar to the requirements for high waveform fidelity with low overshoot. Therefore, most links have the potential for excellent waveform fidelity if. Nevertheless, in a dual-microwave system, the is usually located at the main FM transmitter and is driven by the microwave receivers.
If the STL microwave uses pre-emphasis, its input pre-emphasis filter will probably introduce overshoots that will increase peak modulation without any increases in average modulation. This frees the system from potential overshoot. The Orban ST can be readily configured to produce a pre-emphasized output. Further, it is common for a microwave STL to bounce because of a large infrasonic peak in its frequency response caused by an under-damped automatic frequency control AFC phase-locked loop.
Many commercial STLs have this problem. Some consultants presently offer modifications to minimize or eliminate this problem.
If your exciter or STL has this problem, you may contact Orban Customer Service for the latest information on such services. Analog landline quality is extremely variable, ranging from excellent to poor. Whether landlines should be used or not depends upon the quality of the lines locally available, and upon the availability of other alternatives.
Due to line equalizer characteristics and phase shifts, even the best landlines tend to veil audio quality slightly. They will certainly be the weakest link in a FM broadcast chain.
Slight frequency response irregularities and non-constant group delay characteristics will alter the peak-to-average ratio, and will thus reduce the effectiveness of any peak limiting performed prior to their inputs. It is not a pdf file. Please tick the box below to get download link:.
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