Sparky

Impedance matching is only relevant when transferring power from the driving source to a load. In the case of a broadcast transmitter output amplifier (at the connector) it is designed to have a 50 ohm source impedance. The transmitter load is an antenna with a termination value of 50 ohms resistive. In order to minimize the power transfer loss between the two devices, the conduit (aka coax) must exhibit the same impedance as the source and the load.

Coax cable (or wave guide) at high frequencies is essentially a distributed combination of series inductors and parallel capacitors creating a "balanced out" component. Or in other terms, the series inductive reactance gets canceled out by the parallel capacitive reactance leaving only resistance as a loss component. Reactive components only store energy, where as resistive components convert energy into work (heat). So an ideal cable is essentially a lossless conduit for RF energy (think of it like a bucket brigade where no water dribbles out or is spilled). Real world cables do exhibit losses from two areas. The most significant loss is the dielectric absorption properties from the plastic, foam, or air that fills the space between the center conductor and the sheath. The other loss (minor) comes from the actual resistance of the conductor metals (copper).

In this situation where you are connecting a soundcard mpx output to an exciter input, true impedance matching is not much of an issue for a couple of reasons.
1. The source impedance of a sound card varies greatly depending on the brand, the output driver circuit design, and the type of output being used ie. headphone output (8-200 ohms), line output (200 ohms to several 1000’s of ohms or more).
2. The exciter mpx input designs are mostly if not all high impedance inputs. Typically 10k ohms or more.
3. Due to the low frequencies that make up the mpx signal (practically DC compared to 100’s or 1000’s of MHz) the series inductive component of the coax is so small it essentially provides little contribution in balancing out the parallel capacitance of the cable. In this case 50 ohm coax is nothing more than fancy shielded wire where the impedance of the cable has little relevance.

If the sound card output and the exciter input were both 50 ohms, then yes, the cable impedance would become a factor worthy of consideration. But since this is not the case, worrying about 50 ohm connectors and cable impedance matching is a moot point.

What you need to concentrate on most in creating a low loss connection between the soundcard output and the exciter mpx input is to keep the cable lengths as short as possible, and use the highest quality cable that exhibits the lowest capacitance per unit length (i.e. xxx pF per foot/meter).
Using low quality cable with high distributed capacitance essentially creates a low pass filter that rolls off the higher frequency components of the mpx signal. Leif has been clever enough to design in tilt compensation to mitigate this roll-off effect (special output pre-emphasis to boost the higher mpx frequencies), but it has a finite boost range. So it’s up to you to use the best cable available that ensures the mpx signal remains properly equalized (flat) when it arrives at the exciter input.

Note: When making sound card tilt adjustments it’s very important that you measure the mpx signal at the exciter input, not at the sound card output. Doing so allows you to compensate for the cable roll-off effects.

Looks great. As a RF hardware engineer I can truly appreciate the work you have done and all the effort involved. 😉
I would be curious to see what your carrier phase noise is at 1kHz, 10kHz and 100kHz offsets. If you do not want to publicly share this info send me a PM if you wish.

Sparky

I find that the use of two PLL mixing stages to provide the RVR exciter DDS signal up conversion for FM band frequencies to be a bit unusual and an inefficient design approach.
It’s far easier to use class C signal multiplier stages for the up conversion. Frequency doublers (or triplers or quadruplers) is a tried and true simple method to take a low frequency signal sources and make them higher. But not having taken a look at the inner workings of the RVR exciter I can’t validate the PLL up-conversion approach though.

There is nothing wrong with PLL designs provided the loop filter is properly engineered. As a designer I have achieved PLL designs with low phase noise in the order of 110-130dBc routinely. Noise this low down is completely inaudible. From a listeners perspective, more phase noise is introduced by the local PLL oscillator mixer of their consumer grade receiver than from the transmitter exciter. Also DDS synthesizers are not immune to noise either. Source clock jitter will provide phase error noise on the DDS baseband output along with lousy post DSS reconstruction filtering.

Applying modulation indices and bandwidths as required to satisfy the broadcast standards to a directly modulated crystal oscillator is not possible. This is because the motional mass of the quartz crystal is too great and cannot provide sufficient deflection to give the needed deviation. Plus the dampening effect of the motional mass acts like a low pass filter whereby the MPX signal will be filtered out. Long ago many TX manufacturers used ovenized crystals as the frequency determining component for their exciters, but had to apply phase modulation to generate the modulation indices for FM broadcast. These crystals were typicaly in the 100’s of kHz’s and fed a whole string of class-C multiplier stages to arrive at required frequency.
GE, Gates and Harris had their own unique way of applying the phase modulation to the multiplier stages. (i.e. Serrasoid to name one)

Leif, DDS synthesis does not use a processor running an "algorithm". It’s essentially a clocked state machine that uses a 1/4 sine wave sine look-up table to drive a D/A converter. This converter output through smoothing filters reconstructs the FM signal (much like the sound card makes the MPX signal). By taking a numeric constant (modulation point) and adding it to the internal phase accumulators will generate a phase modulated output signal (FM). Because all the modulation is done in the digital domian, it requires a CPU to prep the audio sample point (plus any MPX signal data) into the proper numeric value prior to sending it on to the DDS chip.

Want to know more about DDS devices visit the Analog Devices Web site. They make lot’s of these kinds of chips.
http://www.analog.com/en/rfif-component … index.html

There is an application note that shows how to build a FM transmitter using one of their DSS chips with a companion DSP processor.

40dB down doesn’t tell you enough on the first analyzer photo. One major problem I see is the two analyzer photos have different settings. The first photo has RBW/VBW’s of 3kHz, the second has RBW/VBW’s of 10kHz? You’re not consistant with your test setup comparisons to be meaningful. Peak amplitude accuracy comes at the expense of narrow RBW and VBW’s (and slow sweep speeds).

In order to get precision measurements you need to be measuring with a RBW of 300Hz, with VBW of 100Hz. The narrower the better. Just be prepared to spend some time waiting for the slow sweep speeds to write the trace to the screen. The reward for your patience is very accurate amplitude measurements of the carrier null. Remember you don’t need to see all the modulation sidebands during these measurements (hence your widebandwidth settings), just the energy right at the carrier center freq, +/-100Hz. So zoom into this region and wait.

You keep this up and I won’t have anything left in my air chain rack to impress people with 😉
The next rack slot to be vacated will be where the Orban 222A spatial enhancer currently lives when you release this latest addition.

OK uber coder… here’s another feature to think about.
How about adding a stereo synthesizer to convert mono source material into a pseudo stereo image… i.e. Orban 275A?

Interesting. Is this 65Hz cutoff based on speaker cone physical displacement, or does it include enclosure resonance?

Marketing gimicks and advertizing one upmanship… total BS. 🙄
Remember it was done because they can not because it was needed.

A very good example of BBP misuse and abuse. I never made it through a single song before I tuned out. My poor bleeding ears… (and I like Greek music too)

Hey Camclone… what’s the point?
In case you haven’t noticed there is a very powerful tool that’s standard issue for all media players each listener has control over…. A VOLUME CONTROL !!!
You don’t have to turn it up for them by crushing the audio into sonic mush!