DSP Speech Processor Experimentation 2012-2020

Version 2.1 June 2013

A DSP speech processor based on the Microchip dsPIC33 Digital Signal Controller, designed to maximize voice levels on communications transceivers.
 
 

A homebrew speech processor designed for SSB applications. Flattens the speech power spectrum and improves the peak-to-average ratio of speech to about 6dB, while maintaining good audio quality. Box size: 110 mm x 70 mm x 24 mm. Hardware features: RF filtering on inputs and outputs. Analog AGC on mike input to prevent input clipping. Ground loop suppression on Tx and Rx. Analog bass EQ boost for transmitter compensation. Rx audio processing input. TDA2003 audio amp for receive output. Polyswitch 12V rail protection. 3 spare microphone pin connections via 8-pin Mini-DIN connectors. Front panel Headphone/Line output for monitoring or recording (both Tx and Rx audio). 44.1 kHz 3.5 mm mini TOSLINK S/PDIF optical digital 2-channel audio output for recording. Power consumption: Under 5 watts (Transmit mode). No power switch - Always-on design. Software features: Dynamic pre-emphasis for improved clarity and reduced distortion. Microphone Flatness adjustment to compensate non-flat microphones. Six-band look-ahead digital audio compressor with delayed recovery and noise gating. Look-ahead digital output compressor for well controlled clipping level. Two-band low-distortion Hilbert Transform clipper. "Virtual clipping" mode for even lower distortion. Multiple processing levels (Bypass, Singleband, Multiband Local through DX). Selectable processing band limits: 60 Hz, 125 Hz or 250 Hz and 3 kHz or 4.5 kHz. Receive mode LMS Noise Reduction. Receive audio AGC (Volume limiter). Single tone, two-tone, modulated tone, swept tone, toneburst and white noise for testing purposes. Automatic tone sequences and white noise in 10 dB steps for processor Self-test mode. Concise processing audio chain overview: Low noise, low-distortion 2-transistor mike preamp with software AGC control to prevent clipping. Gain freezing on input AGC to stabilize level and prevent pumping and noise increase. Dynamic pre-emphasis boost, compressing about 18dB above 1kHz. 20 stages of phase rotation centred on 100Hz (adjustable). 10dB to 24dB of 6-band compression with noise gating. About 6dB (up to ~30dB) of single-band output compression with EQ sidechain. About 6dB (up to ~18dB) of dual-band Hilbert clipping. And it fits in your shirt pocket! :) PCB cost: AUD$21 including AirMail delivery see below. Estimated parts cost: Around $50USD (not including PCBs). Estimated time to build: About two days - depending on skill level.

Controls (Left to right) Button 1: MODE : Resets the display and operating mode to normal Transmit mode. Button 2: SELECT : Steps through the Select menu, to select different options. Button 3: - (minus) : Decrements selected option. Button 4: + (plus) : Increments selected option. LED Bargraph: LED 0 = Pre-emphasis Limiter. LED 1 = Band 1 (60 Hz-125 Hz). LED 2 = Band 2 (125 Hz-250 Hz). LED 3 = Band 3 (250 Hz-500 Hz). LED 4 = Band 4 (500 Hz-1 kHz). LED 5 = Band 5 (1 kHz-2 kHz). LED 6 = Band 6 (2 kHz-3 kHz or 2 kHz-4.5 kHz). LED 7 = Output Compressor. LED 8 = Hilbert Clipper.

Using the economical 28 pin SPDIP dsPIC33FJ64GP802 by Microchip. (or the dsPIC33FJ128GP802)
40 MIPS @ 3.3V, 16k RAM, 64k FLASH, built-in stereo audio DAC with 256x oversampling, 1 x 10/12 bit ADC, 16 bit CPU design, and many other features.
This processor is a real DSP unit, and has many features to speed up processing such as:
Harvard architecture (separate data and program busses).
Hardware support for multiply, divide, MAC and data move instructions.
Two 40 bit DSP accumulators, hardware multiplier, hardware divider (max 32/16 bit in only 18 CPU cycles), instruction prefetching, hardware saturation, hardware rounding, barrel shifter, memory mapped IO, multiple DMA channels, multiple IRQs, multiple clock sources, in-system low-voltage programming... the list goes on!
 

If you are interested in trying to build this unpublished design for an Amateur Radio or CB radio application, contact the author at this email address: If no reply is received after two weeks, try again - Spam filtering may have deleted your message. Remember to check your own spam filtering to ensure you can receive a reply from me! Please note: This project is NOT a kit. This project is NOT a commercial product. There are no panels or components for sale at this time. PCBs are sold out! This project is shown as a demonstration of what is possible, and to share ideas. Schematics are shown below. Software is still in development. dsPIC code is available via email for personal use only. PCB artwork is available via email for personal use only. This unit is a work in progress and continues to be improved. Details, features and audio levels may change in future versions. This version is unable to support VOX operation due to ADC switching. No guarantees can be made - use at your own risk. This project was designed to provide "Medium-Fi" processing for older, homebrew or low-budget transceivers without speech processing. It can also be used on radios with SSB RF clipper processors to improve quality. Modern DSP radios that lack good processing can also work very well with this unit. This processor is designed for maximum loudness, and doesn't try to be "Hi-Fi" - though on-air it sounds very good - almost like a shortwave broadcasting station. This is primarily due to the use of phase rotation, multiband compression and multiband clipping - similar techniques to those used in broadcasting.

Standard audio processors do not suit Amateur radio use: * For SSB, limiting audio level does not control PEP levels accurately. * Most audio processors do not produce a flat output power spectrum required for best SNR on SSB and AM. * Most audio processors do not have the output ground loop isolation required to drive an unbalanced transceiver mike input. What this processor does right: * A Hilbert Clipper is used (IQ vectorsum controlled limiting) for best SSB PEP control. * A flat output power spectrum is produced. This matches the flat noise floor of SSB and AM. * Transceiver ground loops are prevented by using a ground loop cancellation output amplifier circuit.

Some Speech Processing Theory The difference between the quietest and loudest dB levels in an audio signal is referred to as its "Dynamic Range". Due to the large dynamic range of unprocessed speech (greater than 30 dB according to various references), speech processing is widely used to reduce this range for consistent modulation. This is especially important for broadcasting and communication where the maximum peak level is limited by the maximum transmitter power level or the defined modulation standard. Both compression and peak clipping are used to reduce the dynamic range of audio content in these applications. AM and SSB systems have a flat noise spectrum, because they respond equally to noise across the passband. The unprocessed voice has a bass heavy spectrum, as the fundamental and low harmonics of the vocal cords are far stronger than the higher harmonics. FM systems have a noise spectrum that rises with frequency. Pre-emphasis is used in both AM/SSB and FM systems to compensate for these factors. "Loud, bright, clean - pick any two" is a well known choice in FM broadcasting. This is because the receiver de-emphasis makes low frequencies sound louder. To increase brightness, less bass energy is available - reducing loudness. To increase loudness, more clipping can be used which then increases distortion. Since most receivers have some type of de-emphasis or hi-cut filtering, this effect can also happen in communications gear. Multiband compressors are often used in broadcasting and music mastering applications because their action produces less undesirable effects than broadband compression (e.g. "pumping" and increased IMD). Essentially the audio spectrum is divided into bands by a filter bank, then compressed individually before being recombined - sometimes after additional filtering to suppress any compressor distortion products. The multiband compressor also has the effect of producing a more even spread of energy across the audio band, producing a flatter power spectrum. This is useful, as it matches the flat noise spectrum of SSB and AM systems. It gives all the voice frequencies a better chance of being heard above the noise floor, while maintaining a higher perceived quality. This is the approach used here. The multiband compression is teamed up with phase rotation, additional wideband compression and Hilbert clipping to significantly increase the average speech power without adding harmonic distortion, or excessive amounts of intermod distortion. This design will outperform any conventional analog processor, and any standard audio DSP processor.* With its reasonably flat audio spectrum, the processor produces a typical peak-to-average ratio of about 6dB (3kHz BW, Mode 4, clipping ON), while maintaining good quality. * Unprocessed audio with excessive bass can produce high average RF power due to the bass having a high duty cycle. This looks good on a power meter, but results in poor readability in bad conditions.   SSB hates Square Waves! It has long been known that square wave audio fed into an SSB transmitter produces a very spiky RF output with a low average power. This is primarily due to the inherent Hilbert transform that occurs when generating SSB - either by phasing, Weaver, or filter methods. Virtually all SSB modulators also have a highpass response, causing a low frequency rolloff of the audio. The flat tops on a squarewave are essentially DC pulses, and they require an extrodinarily low frequency response in a modulator to faithfully reproduce the waveform without "tilt". This is impractical in a standard SSB transmitter due to the close spacing of these low frequencies to the opposite sideband. Ordinary peak clipping of speech waveforms causes flat tops that resemble square waves. It also adds harmonic distortion that makes the voice sound harsh, raucous and unpleasant. This makes it undesirable for SSB speech processing. Instead, clipping and filtering the SSB RF output produces a higher average power with only an increase in on-channel IMD (Intermod Distortion). IMD sounds far less objectionable on speech, so SSB RF clipping is far better for speech processing than audio clipping. SSB RF clipping has an equivalent, known as "Hilbert Clipping". Hilbert Clipping is straightforward to do in DSP, by using a "Hilbert Transform" to generate a "Q" (Quadrature) signal from the "I" (In-phase) input, then finding the vector sum of the I and Q signals and using it as a gain control signal. Effectively, the system acts as a compressor with zero attack and recovery time. The waveforms are limited without any squaring, causing IMD from the changing gain - but (ideally) no harmonic distortion. The output can be clipped heavily without generating square waves. Since the SSB RF envelope is proportional to the vector sum of the I and Q signals, it produces accurate clipping of the RF envelope. This produces higher average SSB RF power with low harmonic distortion, plus some on-channel intermod distortion.

Page Shortcuts

LATEST DEVELOPMENTS

• Oct 04: Multiband Compressor IIR filters have been pipelined for increased speed and efficiency.
• Oct 06: The modular IIR filter code designed for the Multiband Compressor is now also being used for other IIR filters.
• Oct 06: The total size of the current software hex code is about 8k.
• Oct 07: The IIR Phase Rotator has been updated to 32-bits to reduce spurious products from roundoff noise. Spurious products are now -84dB or better.
• Oct 08: The Multiband Compressor code has been modularized and streamlined using macro instructions. Efficiency slightly improved.
• Oct 11: The Multiband Compressor has been updated to include "Delayed Recovery" time constants, to increase loudness and quality.
• Oct 14: The Hilbert Clippers have been modified using peak-hold and zero-crossing techniques to reduce low frequency harmonic distortion by over 20dB (now -55dB @ 86Hz, Mode 4).
• Nov 06: The Multiband Compressor has been upgraded to use look-ahead + current amplitude values for lower overshoot & ripple distortion.
• Nov 20: New LMS NR recording.
• Nov 21: OpAmp voltage listings added.
• Dec 01: "Construction Notes" updated.
• Dec 05: Software bug in Delayed Recovery routines found and fixed - Harmonic distortion is now measurably lower.
• Dec 05: Dynamic Pre-emphasis changed to Delayed Recovery mode, harmonic distortion decreased.
• Dec 06: Distortion measurements added - see below.
• Dec 07: Receive audio AGC changed to Delayed Recovery mode for faster response and low distortion.
• Dec 07: More distortion measurements added - see below.
• Dec 11: Utility modes expanded, to allow a choice of highpass response at 3 kHz bandwidth. Choices are: 60 Hz, 125 Hz, 250 Hz.
• Dec 15: New LMS NR recording.
• Punctuation used on this page is being modified to improve the performance of language translation tools.
• LED bargraph numbering referred to on this page now starts at 0, for better agreement with function numbering.
• Dec 22: Setup information added. Targets added to webpage for improved navigation.
• Dec 30: Multiband compressor modified to have all filtering before each compressor stage. Should increase loudness further, and may reduce compressor distortion.
• 2014 -----------------------------------------
• Mar 14: The PCB design is being updated to a standard acceptable for commercial PCB production techniques. This may take some time to complete.
• Mar 15: This webpage has been reformatted to fix a word wrap problem on small screens.
• Apr 20: Some hole sizes on the pushbuttons and 3.5 mm sockets were too small. They have been corrected.
• Apr 24: Select modes updated: Phase Rotator number of stages is now selectable. Minimum 1, maximum 26. Improves average SSB power for specific voices.
• Apr 25: Programming Adaptor Data I/O wiring error corrected. **IMPORTANT**
• Apr 26: Troubleshooting section added.
• May 02: Compression gain setting added. Works on Modes 1 through 7.
• May 06: Swept tone flatness test added on Mode "b". Easily checks whether output EQ matches transceiver.
• May 06: Hardware output EQ modified. 10k resistor on U1b pin 6 changed to 4k7 for improved flatness with typical transceivers.
• May 07: Decoupling network added to electret microphone phantom power supply.
• May 07: Microphone input coupling capacitors reduced to 1 uF.
• May 07: PCB version 221 artwork available.
• May 08: Minor EQ changes (spectral skew) are currently being evaluated to reduce transceiver distortion and upgrade processing performace. (Currently -1 dB on Band 1 & Band 2.)
• May 09: Audio I/O pinout charts added.
• May 16: Division overflow handler added to compressor code.
• May 16: Receive audio input OpAmp resistors changed, for lower gain and better common-mode rejection.
• May 17: LMS NR filter bug fixed. Performance improved.
• May 18: Pulsed sinewave tonebursts added to tone generation.
• May 22: Receive audio input OpAmp resistors changed to commonly obtainable values.
• July 04: Gerber PCB artwork is available. The Gerber files appear correct, but have not been fully checked for errors at this time.
• July 21: Two-tone SSB test tone separation changed to 2 kHz.
• Aug 07: Minor webpage updates.
• Aug 07: Dynamic pre-emphasis filter changed to 1 kHz IIR highpass pipelined filter. Improves power spectrum flatness on voice.
• Aug 10: Bandwidth And Noise section added to web page.
• Aug 10: Noisy Audio Intelligibility Tests added to Bandwidth and Noise web page.
• Aug 16: Worst DDS spurious levels reduced by about 13 dB.
• Aug 16: Multiband compressor distortion reduced on Band 5.
• Aug 16: New distortion measurements - Lower harmonic levels.
• Aug 16: New frequency response graphs.
• Aug 26: Dynamic pre-emphasis filter changed to 700 Hz IIR HPF with a Q of 0.4, for improved EQ.
• Aug 26: New voice frequency analysis plots added to Bandwidth and Noise page.
• Aug 27: Power spectrum flatness improvements.
• Aug 27: Noisegate added to dynamic pre-emphasis.
• Aug 29: Output compressor performance improved slightly by using fast RMS detection.
• Aug 30: New off-air audio recording added.
• Sep 20: Phase rotator centre frequency adjustment changed to fixed presets.
• Sep 20: Shaping filter added after multiband compressor to improve EQ.
• Sep 21: Experiments with modified EQ continue, to allow increased clipping with less perceived distortion.
• Sep 25: Phase rotator presets now also load an optimized setting for the number of rotation stages for each frequency choice.
• Sep 26: Multiband compressor output levels are now dynamically loaded, allowing custom EQ output level curves for each mode.
• Oct 04: Output compressor gain setting added.
• Oct 04: Modified output compressor sidechain allows reduced low frequency clipping, improving quality.
• Oct 14: Output EQ stage changed to a peaking filter. Significantly improves transceiver EQ compensation, improving flatness.
• Oct 14: New S/N ratio test recording added to Bandwidth and Noise page.
• Oct 19: Receive EQ filter added (bass boost). Significantly improves received speech quality, and improves readability at low volume settings with typical ambient noise levels.
• Oct 28: New HDSDR recording.
• Nov 02: DSP circuit updated to revision 5.7. This adds a PNP inverter, which prevents PTT activation if the DSP negative power rail is lost.
• Nov 22: DSP circuit updated to revision 5.8. This adds the 1k resistors on the LED Data and LED Clk lines.
• Nov 24: PCB updated to v235. This corrects some mechanical problems and offers improvements. Gerber format PCB artwork nears completion.
• Nov 26: LED alignment graphic added.
• Nov 29: Schematic parts have been renumbered to avoid duplication. This may cause some errors in the text references to part numbers.
• Dec 06: DSP circuit updated to revision 6.0. Some component IDs and placement corrected.
• Dec 11: DSP circuit updated to revision 6.1. More corrections.
• Dec 12: PCB updated to v244. More component designators added, and wire links eliminated (flying leads have not been eliminated).
• Dec 19: PCB updated to v246.
• Dec 20: Duplicated designator on R72 corrected. Resistors numbered above R72 now redesignated. Display PCB 1 resistor designations now begin at R100.
• Dec 21: Error in Display PCB schematic LED wiring diagram corrected. Display PCB LED designations now start at D100. PCB layouts unaffected.
• Dec 21: Bill Of Materials (parts list) added.
• Dec 21: PCB updated to v247.
• Dec 23: Schematic updated to v6.8.
• Dec 27: PCB updated to v251.
• Dec 29: PCB updated to v252.
• 2015 -----------------------------------------
• Jan 06: SW1 to SW4 renumbered to match layout and usage.
• Jan 06: Schematic updated to v6.9.
• Jan 06: PCB updated to v253.
• Jan 24: Analog and Digital schematics updated. PCB updated to v254.
• Jan 26: Diodes renumbered. Analog and Digital schematics updated. PCB updated.
• Jan 30: Track error on Display PCB 1 corrected. PCB updated.
• Jan 31: Diode D3 rearranged for consistency. Overlays, copper and outline adjusted to keep all elements within board outline. PCB updated to version 256.
• Feb 07: Discrete LED pads on Display PCB 2 adjusted for better alignment. J10 TOSLINK socket pads adjusted for better alignment. PCB updated.
• Apr 20: Work on this project has been slowed due to personal circumstances.
• Apr 25: PCBs have been submitted for prototype manufacture. The later production PCBs will be free of the errors mentioned below.
• Two prototype PCBs have been manufactured for evaluation. Their properties are:
• 1.6mm double-sided fibreglass boards (later production runs will use 1.0mm to improve clearances).
• Nickel-Gold plated pads.
• Top and bottom white silkscreen overlays.
• Top and bottom green soldermask.
• Plated-through vias replace the wired grounding vias.
• No jumpers required (flying leads are still used in four places).
• Three PCB set - Main PCB, Display PCB1 & Display PCB2.
• All PCBs are edge routed and finished - no cutting, trimming or filing required.
• Apr 29: Prototype PCB manufacturing is progressing.
• May 02: New prototype PCB pictures added.
• May 04: Modifications list added.
• May 09: Concise processing chain information added.
• May 13: Construction notes updated. Programming section added.
• May 16: Bill Of Materials updated with modified capacitor heights to clear updated Display PCB boards.
• May 17: Prototype 2 picture added.
• May 17: Important note to constructors - Error on Reg 2 heatsink pad.
• May 19: Updates to Bill Of Materials and PCB.
• May 21: Page shortcuts added. Code examples page added.
• May 22: Noisegating added to Output Compressor to improve noise gating further.
• May 23: New demonstration video and audio.
• May 25: Added pricing estimate to web page.
• May 27: Updated BOM with changes to v260.
• June 01: Design work for v3.0 has begun.
• June 10: Front panel drilling template added.
• June 11: Capacitor values on Display PCB 1 and BOM updated. Q4 configuration changed to grounded Collector. Analog schematic updated.
• June 11: PTT state filtering and level ramping (T/R click suppression) added to software.
• June 12: D12 added to PTT line for CPU protection. Schematic updated.
• June 14: C15 increased to 220n to fix HF oscillation and overheating of U4 in prototype 2.
• June 25: Clock speed dropped slightly to fix audio glitches in prototype 2 with TOSLINK feeding a C-Media USB Sound device.
• June 27: Mini-DIN pinouts updated.
• July 02: Pre-emphasis filter changed to pipelined type. Pipelined de-emphasis shelf filter added on S/PDIF output for Tx audio.
• July 03: A production run of 50 PCBs has been ordered from an Asian supplier.
• July 07: Compressor sidechain filtering updated to a -3dB 600Hz high shelf filter, for a smoother sound.
• July 12: Macros for cascaded multiband compressor filters on bands 4, 5 and 6 combined to reduce CPU usage.
• July 13: Macros for cascaded 32 bit multiband compressor filters on bands 1, 2 and 3 combined to reduce CPU usage.
• July 13: The 50 new commercially made PCBs have arrived. Boards are now available for purchase. These PCBs have no errors.
• July 14: New PCB260 picture added.
• July 16: Construction overlay added to web page, with the locations of all resistors marked.
• July 19: Menu functions for saving settings added.
• July 29: Save settings function is now operational.
• Aug 01: Mini-DIN plug section added.
• Aug 02: Updates to construction information.
• Aug 05: Analog schematic updated.
• Aug 22: Constructors section added.
• Sep 02: Bug in transmit EQ mode Flash memory routine fixed.
• Sep 08: Bug in Tx level ramping fixed.
• Sep 15: Bug in multiband compressor gain Flash memory routine fixed.
• Sep 23: Error in BOM fixed. New Gridded BOM added.
• Dec 12: Firmware updated to save & load LED bright/dim setting.
• Dec 16: LMS NR filter changed from 64 taps to 32 taps to improve quality.
• Dec 23: Default NR settings adjusted for improved quality.
• Dec 24: Error in Rx clipping indicator level fixed.
• Dec 29: Bug in Frequency Sweep tone limits fixed. Sweep rate reduced.
• Dec 31: Input EQ improved. Dynamic pre-emphasis increased. 3kHz filters improved. Programming mode updated.
• 2016 -----------------------------------------
• Jan 01: Dynamic microphone amplifier circuit added to webpage.
• Feb 07: Schematic updated to 7.4e. C58 position corrected. New distortion measurements added.
• Feb 15: Select modes updated. Default startup mode is now user selectable.
• Feb 18: Shaping filter (for mode 2 and 4) changed to 500Hz low-cut -3dB shelf. Multiband levels tweaked for improved flatness on mode 5.
• Feb 24: More tweaks to improve mode 5 flatness to within about 1dB.
• Feb 25: LED adjustment display added for input and output level settings.
• Feb 28: Bugs in Hilbert transform fixed. Efficiency improvements in Hilbert clipper code.
• Feb 29: Bug in look-ahead buffer modulo limit fixed. Multiband attack rates slowed for lower distortion.
• Mar 17: Compressor code updated to a more modular structure for coding neatness. This update introduced a distortion bug. See Sep 09 entry.
• Sep 09: Compressor code has been reverted to Feb 29 version to fix a distortion bug. Output compressor attack time constant adjusted to prevent clipping.
• Sep 20: Testing is underway to optimize Mode 4 for 3kHz bandwidth applications. Changes have been made to the multiband compressor EQ and the multiband clipper crossover frequency.
• Oct 12: Extra RF filtering section added to web page.
• Oct 14: New demo recording added to web page.
• Oct 19: EQ and output level modifications section updated.
• Oct 21: New demonstration recording added.
• Oct 21: Changes to EQ on Mode 6 and Mode 7, for a flatter response on voice. Band 1 level reduced 6dB. Output compressor gain reduced.
• Oct 25: Section on Grounding added to page.
• Nov 01: Output compressor moved to be after Hilbert transform stage. Now uses vector sum of I and Q channels to measure signal magnitude to improve peak control when Hilbert clipper is disabled.
• Nov 02: Changes to compressor timing and level parameters to improve sound quality.
• Nov 02: Bug fixed in Output compressor noisegate.
• Nov 05: Improvements to Output compressor to improve peak control and average power and reduce distortion.
• Nov 06: Output compressor "Virtual Clipping" level adjustment added. (Select menu "c").
• Nov 10: Microphone Wiring diagram added to page.
• Nov 15: Microphone "Flatness" adjustment added (Select menu "_") to compensate for non-flat microphones. "Virtual clipping" boosted by an extra 6dB in Mode 7, with Hilbert clipping off.
• Nov 25: Impulse response test signal added. Select menu operation modified for easier setting.
• Nov 25: New demonstration video.
• Dec 19: Mini-DIN pinouts added to Microphone Wiring section.
• 2017 -----------------------------------------
• April 17: Work has begun on an SDR based system, to support advanced SSB modulation and demodulation methods. This is the next step beyond DSP speech processing, and will include a new DSP design to process I and Q signals.
• Sep 18: Mode 4 multiband gain EQ adjusted to reduce response at the extremes. Band 1 -6dB, Band 2 -3dB, Band 5 -3dB, Band 6 -6dB.
• Nov 27: Analog AGC Gain Freeze Disable option added (Select menu option 28 ["]). This option allows the DSP to recover automatically when large audio inputs (like pop noise) temporarily desense the input AGC. This avoids the need to manually reset the AGC gain by pressing button 1. AGC gain ducking will still be noticed, as will any loud audio noises that cause the AGC desense.
• 2018 -----------------------------------------
• Jan 13: Input AGC settings updated to improve AGC temperature stability. New firmware is available.
• Page updated to show dsPIC33FJ128GP802 option. This chip has twice the FLASH memory, is compatible with the original design, and only costs a little more.
• Jun 12: Phase rotator specs on web page corrected. Firmware unchanged.
• Jun 14: Demo video audio format changed to fix playback problem in Firefox.
• Jun 22: Experiments with Faust DSP are ongoing. Various DSP v2.1 processing improvements are being tested using Faust DSP code.
• Nov 26: New firmware using Weaver baseband SSB is currently being developed. This should slightly improve processing performance.
• 2019 -----------------------------------------
• Mar 30: New firmware using Weaver baseband SSB is being tested. This has improved audio quality and average output. Also includes CESSB overshoot compensation.
• Mar 31: New firmware using Weaver baseband SSB is available. Ask for it by email.
• 2020 -----------------------------------------
• May 27: The last two PCBs have been sold. If more are manufactured, it will be announced here.
• A video of the unit in operation is featured below.

Bandwidth And Noise Tests - Now moved to its own page. Some interesting results here with speech buried in noise.

PCBs

Version PCB260.

Three PCBs in one set:

1. PCB260A = Main board
2. PCB260B = Display PCB 1
3. PCB260C = Display PCB 2 (LED board)

These PCBs have no errors. See pictures of the boards here.

The thinner 1.0mm PCB material improves mounting clearances of the LEDs, audio IC, and connectors.

Constructors

Frank has built the PCBs into a 1U rack case to expand the I/O connectors and add extra controls:

An early test can be heard here.

Processor demonstration. G0GSR (Frank) in QSO with M3ZDZ (another Daz), recorded from Hack Green WebSDR. EDITED. This DSP Speech Processor is demonstrated, along with a DBMax for comparison.
 
 

PCB sales and known working units, by country:
* Prototype/test units
 

Country	PCBs	Units
Australia	5/2*	4
Brazil	2	0
France	1	0
Germany	3	1
India	8	13
Liechtenstein	1	0
Netherlands	3	0
New Zealand	1	0
Norway	1	0
Poland	1	0
Portugal/Azores	2	0
UK	8	3
Ukraine	1	0
USA	3	0

 

Demo video & audio Nov 25, 2016. Video of the DSP Speech Processor v2.1 in action - H.264/MP4 Format. This recording has been lowpass filtered to more closely replicate the sound of passing through transmit filtering. The "r" in receive mode shows Rx volume adjust is selected (- and + buttons change Rx Volume). The default transmit behaviour is to show Tx processing level number (- and + buttons change mode number). The LED "bargraph" is actually configured to show limiting action of each compressor or limiter stage. The LED ordering is: Pre-emphasis limiter, band 1 through band  6, output compressor and Hilbert Clipper stages. Audio was recorded via S/PDIF interface for best quality. Audio Audio recording via HDSDR Flat receive (no de-emphasis) October 28, 2014. CB chat August 8, 2015 NEW Demonstration in 4.5kHz bandwidth. 27Mhz CB transceivers, 12W PEP over a 15.7km non-LOS path. DSP Speech Processors used on both radios. Some EQ applied to flatten receiver audio responses. Default Mode 5 on both units. Recorded via TOSLINK output at each end of the QSO, then edited together. SSB frequency correction was applied in Spectrum Lab to correct mistuning.   LMS Noise Reduction test - 40m Amateur band 2013-06-24-NR-test-7155kHz-1730-local.mp3 NR is turned off for a short while in this MP3 to show effectiveness. Some audio clipping is heard - AGC routines are still being adjusted.   2013-11-20-7150kHz-0150-local.mp3 NR is turned on at 21 seconds into this recording.   2013-12-13-VK2IY-VK2QA-40m-2100-local.mp3 NR is used throughout this recording. New Rx AGC is used. What a great story by VK2IY Angus! Measured specs (S/PDIF output) White Noise Spectrum Plot 25/02/2016 @-6dBFS - Mode 5.   White Noise Spectrum Plot 16/08/2014 @-6dBFS - Mode 4. The response flattens out at high levels. White Noise Spectrum Plot 16/08/2014 @-16dBFS - Mode 4. The bass response is designed to drop slightly as the level drops 10dB. White Noise Spectrum Plot 16/08/2014 @-26dBFS - Mode 4. The bass response drops more as the level drops 20dB. White Noise Spectrum Plot 16/08/2014 @-36dBFS - Mode 4. At very low levels the dynamic pre-emphasis dominates the response. Measured 02/03/2016. Sinewave Distortion, Aliasing & Spurious levels - Processing mode 5 - Tested in Self-test mode. Frequencies are rounded to the nearest Hz - Hanning window. dB is dB below full scale via S/PDIF output. dBFS Fundamental Frequency dBFS Worst Spurious Frequency -14 86 -66 259 (3rd harmonic) -12 172 -76 862 (5th harmonic) -12 345 -74 1035 (3rd harmonic) -12 689 -81 1379 (2nd harmonic) -12 1379 -90 2759 (2nd harmonic) -12 2758 -97 3184 (spurious) -12 4137 -81 2795 (spurious) Some of the distortion measured is due to spurii produced in the DDS sinewave routine. Harmonic distortion levels are very low in all stages. The main distortions are now clipper IMD, residual aliasing and IIR filter noise. Measured 07/02/2016. Sinewave Distortion, Aliasing & Spurious levels - Processing mode 5 - Tested in Self-test mode. Frequencies are rounded to the nearest Hz - Hanning window. dB is dB below full scale via S/PDIF output. dBFS Fundamental Frequency dBFS Worst Spurious Frequency -13 85 -65 259 (3rd harmonic) -12 172 -74 1207 (7th harmonic) -12 345 -71 1035 (3rd harmonic) -12 689 -82 2068 (3rd harmonic) -12 1379 -89 4138 (3rd harmonic) -12 2758 -96 4881 (spurious) -12 4137 -87 1379 (DDS spur) Some of the distortion measured is due to spurii produced in the DDS sinewave routine. Harmonic distortion levels are very low in all stages. The main distortions are now clipper IMD, residual aliasing and IIR filter noise. Measured 07/02/2016. Noise level, measured by waveform statistics, S/PDIF output @5.5kHz BW. Electret unplugged, gain at minimum. Mode 5: -38.59dBFS RMS - Noise Gate off. Mode 5: -59.39dBFS RMS - Noise Gate on. Measured 10/11/2013. Noise level, measured by waveform statistics, S/PDIF output @5.5kHz BW. Mode 0: -84.07dBFS RMS - Electret unplugged, gain at minimum. Mode 4: -42.81dBFS RMS - Noise Gate off. Mode 4: -57.61dBFS RMS - Noise Gate on. Compression @689Hz  = 29.15dB. Compression @4136Hz = over 36dB. (updated 05/08/2014) Measured 19/08/2013 after more input gain optimization. Noise level, measured by waveform statistics, S/PDIF output @5.5kHz BW. Mode 0: -86.00dBFS RMS  - Mike input shorted. Measured 17/08/2013. Noise level, measured by waveform statistics, S/PDIF output @5.5kHz BW. Mode 0: -84.07dBFS RMS - Electret unplugged, gain at minimum. Mode 0: -72.24dBFS Peak - Mike input shorted. Mode 0: -84.30dBFS RMS  - Mike input shorted. Mode 0: -85.39dBFS RMS  - ADC input shorted. The microphone amplifier appears to be making a minimal contribution to the input noise.

Power Consumption Under 5 Watts (Transmit mode). Current drain: as of 06/11/2013 @ 13.8 Volts. Transmit, Bright mode: 330mA. Transmit, Dim   mode:  200mA. Receive,  Bright mode: 310mA+   Expect current to rise with received audio driving a speaker. Receive,  Dim  mode:  180mA+   Expect current to rise with received audio driving a speaker. If current consumption is much higher than this, test for audio amp oscillation, a short circuit, or a failed SMD regulator.

Firmware Update (new)

The new firmware has some differences:

• The Output compressor look-ahead has been removed, as it is not beneficial (overshoots are allowed before the clipper).
• The Output compressor is no longer in IQ mode, just single channel.
• The Hilbert Transform is removed, and Weaver SSB mode is used instead. Processor output is still audio only.
• Weaver mode produces flatter response & more precise phase for lower harmonic distortion.
• The Hilbert clipper no longer requires peak-hold and zero crossing functions to reduce LF distortion.
• The Hilbert clipper is single band. Weaver mode makes a 2-band Hilbert clipper impractical.
• Clipping accuracy has been improved, by using an "Equiripple" magnitude estimation routine. Error is < 1%.
• "Virtual clipping" is no longer available, and is not required.
• CESSB overshoot compensation follows the Hilbert clipper.
• The processed Weaver SSB is FIR bandpass filtered to remove distortion products, and reduce transmitter overshoots.

Signal processing structure - Original Firmware (old)

Multiband look-ahead compressor
A multiband compressor reduces the dynamic range of the audio with few compression artifacts, and flattens the spectrum power of the voice for more even clipping at all frequencies. Produces large amounts of bass, treble and midrange energy simultaneously. This improves quality, and allows a greater amount of clipping. The current version uses six bands, each one octave wide. Each band has a 13mS delay buffer. Now includes HF gain trimming for microphone flatness compensation (Select menu "_" Flatness adjust).

This multiband compressor is a composite of 24 IIR Butterworth filters, and 6 individual compressors.
 

The look-ahead structure used in all the compressors allows a slower attack time without signal clipping. This reduces distortion. The multiband compressors are essentially overshoot free. Short peaks are best removed by the Hilbert clipper stage, so the Output compressor doesn't attempt to be overshoot free, just low distortion. The compressors measure the signal magnitude by simply taking the absolute value of the audio samples. The magnitude is converted to an inverse function by dividing it into a large number to produce a control signal. The control signal is then slew-limited for slower attack, smoothed for controlled release, and clipped to limit the maximum gain. This method has good performance and very predictable levels. A small RAM buffer delays the audio before the samples are multiplied by the control signal, which brings the output to a constant level (above threshold). This produces a compression ratio of infinity, and the forward acting structure ensures unconditional stability. Placing the integration stage after the inverse divider stage has reduced compressor distortion, probably by improving the smoothness of the gain control signal.

The Multiband Compressor now uses the DSP accumulator to scale and sum each band output for fine control over individual band levels.
Input gains are adjusted for some pre-emphasis at low levels, and output levels are set for a reasonably flat response at high levels.

Oct 04, 2013: These IIR filters have now been pipelined and modularized for faster execution, and neater code. This has produced some benefits:

• The filter gains are better matched, and the combined frequency response is now much flatter (within 2dB from 80Hz to 3kHz).
• The multiband compressor now has higher usable gain, and hence is capable of more compression.
• Filter overflow is now eliminated, as the new code structure clips cleanly if overdriven.
• The new code macros can be used elsewhere in the processing for other filtering purposes.

Oct 08, 2013: The compressor code has now been rewritten into modular form. A single macro instruction repeats the compressor and noisegate code in each of the six stages. This makes the code easier to understand and upgrade.

Oct 11, 2013: The Multiband Compressor code now includes "Delayed Recovery", to reduce control ripple while speeding up recovery time. This increases loudness, and reduces distortion. Delayed recovery works by setting the compressor recovery time constant to infinity whenever a signal increase is detected. After a delay (inversely proportional to the frequency), the decay time constant progressively speeds up to rapidly adapt to the new audio level.

Dec 30, 2013: The filters used in the Multiband Compressor were inititally placed both before and after each compressor stage. They have now all been moved in front of the compressors. This should improve compressor action (increasing loudness) and reduce some compressor distortion products, by reducing out-of-band energy at the compressor input.

May 08, 2014: Tests are being conducted to evaluate subjective audio quality using non-flat (spectral skewed) output from the multiband compressor. This may reduce bass response by between 1 and 6 dB, and increase treble.

Sep 20, 2014: The spectral skewing has been assisted by a following shelf filter. This can dramatically increase quality by reducing bass distortion in the clipper. Using a shelf filter reduces problems with filter leakage in the multiband compressor due to widely varying band output level settings required for EQ correction.

Oct 16, 2014: The spectral skewing is now assisted by a following -10 dB 300 Hz peak filter (notch). This increases quality by reducing bass distortion in the clipper at moderate clipping levels.

Sep 23, 2016: The Mode 4 spectral skewing has been removed. Mode 4 is now optimized for 3kHz bandwidth use. Band 1 level has been reduced 7dB to keep the voice fundamental at a similar level to the higher bands. Since band 1 has only a single voice frequency present, all the power is concentrated on one frequency - causing it to be out-of-proportion if adjusted for flatness on white noise (like Mode 5 is). Compressor gain on band 4 and band 5 has been reduced 6dB to improve voice quality in 3kHz bandwidth.

Sep 23, 2016: In Mode 5, band 4 and band 5 have had their compression gains reduced 3dB to improve voice quality.

Nov 02, 2016: Compressor timing modified to improve performance.

Nov 15, 2016: Microphone Flatness adjustment added, to trim the gains on band 5 and 6 to compensate non-flat microphones.

Per-band noise gate
Downward expands the compressor output below a preset threshold, to reduce background noise.

The noise gate also removes IIR filter noise, which is noticeable when the multiband compressor is active.
Currently the noise thresholds are preset in software. If possible, they will be made automatic.
The noise gating is limited to about 20dB. This helps prevent gating artifacts, and also helps the other station to hear when you are still transmitting.

Shaping filter
Modifies the response going into the output compressor to provide EQ or filtering. Implemented as a 2nd order biquad, allowing a variety of IIR filter types to be used. Shaping the spectrum going into the output compressor can increase quality by reducing unwanted distortion. Using a separate filter to perform this EQ avoids band leakage problems from excessive scaling of the multiband compressor outputs. The shaping filter is now only used in Mode 2.

Oct 16, 2014: This filter was a -10dB peak (notch) at 300 Hz. When combined with the multiband compressor output level EQ, this produces a dip in the voice energy around 300 Hz. This reduces some very obvious bass clipping distortion in the region 300 to 600 Hz, resulting in better sounding audio at moderate clipping levels. At high clipping levels, the results improve with a flat multiband output. This filter is only used in mode 2.

Feb 18, 2016: This filter is now a 500Hz highpass 3dB shelf filter. This produces less clipping under 500Hz to reduce bass distortion.

Sep 23, 2016: Shaping filter is now only used in Mode 2. It is now disabled in Mode 4. Mode 4 is now optimized for 3kHz bandwidth audio.

Hilbert Transform filter
The Hilbert Transform filter adds a 90 degree phase shift to the entire audio spectrum, generating a Q (Quadrature) channel, from the I (In-phase) input. This enables accurate measurement of audio levels and helps avoid aliasing distortion from non-linear processes causing beats between audio harmonics and the sample rate. As this doubles the workload of the following processes, it was placed here instead of earlier in the signal chain.

Look-ahead output compressor
Makes up for variations in level due to the multiband compressor outputs adding in random phases. Boosts loudness. Increases average power. Gives the impression of more bass. Provides a more constant drive level to the Hilbert clipping stage. Produces a very well controlled output level. Uses a 1.45mS look-ahead buffer. This compressor now has dual time constants to deal with fast peaks and slower average signal variations simultaneously ("Virtual clipping"). The gain control signal is now lowpass filtered to reduce audible distortion.

Limiting action is shown by lighting the last orange LED (LED 7).
Has a slower attack time than other compressors to improve loudness, and reduce compression artifacts.
Uses Look-ahead peak vector sum computation of the I and Q channels to compute the signal magnitude.
Includes Delayed/Adaptive Recovery to allow a faster recovery time while avoiding control ripple distortion.
Delayed recovery works by setting the compressor recovery time constant to infinity whenever a signal increase is detected. After a set time delay, the decay time constant starts to recover.
Adaptive recovery speeds up the compressor recovery until another peak is detected. This makes the compressors progressively speed up to adapt to the new audio level.
NEW "Peak stretching" added to level detector to improve control sensitivity to fast signal peaks.
NEW "Virtual clipping" added to improve peak-to-average ratio without creating excessive distortion. This adds a second fast time constant in parallel with the normal compressor time constant, to deal with fast peaks.
NEW Control signal lowpass filtering added to slow down gain changes and reduce distortion.
"Virtual clipping" is now disabled when Hilbert clipping is enabled.
"Virtual clipping" level is now adjustable via Select menu "c" (Global setting = affects all modes). Adjustable from 0 to 6dB. Level is boosted 6dB more on Mode 6 and 12dB more on Mode 7.

Oct 29, 2014: The Output Compressor now has a bypass mode in the Select menu. When active, it feeds the Multiband Clipper from the Multiband Compressor directly. This changes the loudness/distortion performance. Peak clipping level increases, but average clipping level decreases.

Aug 29, 2014: Fast RMS level detection has now been added. This is the same routine used in the Hilbert clippers. Removed, due to processing order changes.

May 22, 2015: A noise gate has been added to the Output Compressor, to improve noisegating performance further.

Nov 01, 2016: The compressor structure has been updated to use the vector sum of the I and Q channels to improve accuracy. This is important when Hilbert clipping is disabled, to still provide good SSB power control.

Nov 02, 2016: Compressor parameters modified to improve performance.

Nov 05, 2016: "Peak stretcher" added. "Virtual clipping" added. Control signal lowpass filtering and matching audio delay added. Latency reduced.

Nov 06, 2016: "Virtual clipping" is now disabled when Hilbert clipping is enabled. When Hilbert clipping is disabled, Virtual clipping level is boosted 6dB in Mode 6 and 7. Control signal lowpass filtering reduced to 500Hz for lower distortion.

Nov 15, 2016: "Virtual clipping" now boosted by 12dB in Mode 7.
 

EQ sidechain
Fixed lowpass EQ on the compressor level sensing input reduces compression on high frequency signals. This improves loudness, especially with large amounts of pre-emphasis. It also reduces level "pumping" or "flutter" of low frequencies by strongly emphasised high frequency voice components. Makes the high frequencies clip harder in the Hilbert Clipper. Only enabled when Hilbert clipping is enabled.

Nov 05, 2014: The EQ sidechain is now disabled if Hilbert Clipping is disabled, to prevent overshoots.

Oct 04, 2014: The EQ sidechain now includes a mix setting to increase compressor sensitivity to low frequencies when the shaping filter is active, reducing low frequency clipping for improved quality. This is only useful on modes 2 and 4, as they are the only modes using the shaping filter. Obsolete.

July 07, 2015: The EQ sidechain now uses a biquad shelf filter for more predictable response. Currently, it is a -3dB 600Hz high shelf. This causes frequencies above 600Hz to clip up to 3dB harder than frequencies below 600Hz. This appears to produce a cleaner processed sound.

Nov 15, 2016: Output compressor envelope (Hilbert clipping off, virtual clipping active, Mode 5) - Average power has been greatly increased.

Clipping is used after the compressors to increase the average power, while decreasing peak levels.
Larger amounts of clipping produce a louder signal, but increase distortion.
Three techniques are used to allow higher clipping levels, while keeping distortion lower:

• Hilbert Clipping: Produces mostly intermod distortion (IMD) - Equivalent to SSB RF clipping, but performed at baseband.
• Peak-hold and zero-crossing techniques: Lowers residual harmonic distortion by preventing the Hilbert Clipper from changing the waveform curves excessively.
• Multiband clipping: Limits the IMD produced by the Hilbert Clippers to smaller frequency ranges.

The vector sum calculation is the same math technique used for finding the perimeter of a circle from a central point. The I & Q channels describe a sinewave as a rotating vector. The Hilbert Clipper works to limit the radius of the vector which equates to the signal amplitude. All real signals can be represented as a combination of sinewaves, so the process works for all waveforms.

The vector sum computation is a fast approximation that produces better than -30 dB average error. Suppresses harmonic generation better than 30 dB at high frequencies. At low frequencies, harmonic distortion is reduced further by using peak-hold and zero-crossing reset techniques.

Version 1 of the DSP Speech Processor used a fast audio compressor that had its level control synchronized to the waveform zero crossings. This allowed it to maximize the waveform half-cycles while keeping harmonic distortion levels low. A similar approach is applied to the Hilbert Clippers to reduce harmonic distortion from imperfect Hilbert Transforms and vector sum computations. A peak-hold routine is added to the magnitude estimation (vector sum) output. The peak-hold value is reset to the current magnitude output value at each zero crossing.

Oct 14, 2013: The Hilbert Clippers have now been modified to use peak-hold and zero-crossing reset. This reduces low frequency harmonic distortion by an additional 20 dB or more, without significantly compromising limiting effectiveness. This creates a limiter with zero attack and recovery time, and low harmonic distortion and aliasing. Produces higher average power levels on SSB.

A Multiband Hilbert Clipper clips the audio in two separate frequency ranges. The clipper outputs are combined by a matrixing process. This restricts the bandwidth of each clipper to reduce distortion, and simultaneously allows both bands to reach nearly full output. Methods to further reduce distortion are being investigated.

Allows higher clipping levels while maintaining good quality. Audio sounds strong and clear.
Appears to have good peak control.

Sep 23, 2016: The Multiband Hilbert clipper now uses a fixed crossover frequency of 2kHz in all modes and bandwidths.

Nov 15, 2014: Output envelope - Mode 5. Updated. "The".

Nov 15, 2014: Output waveform - Mode 5. Updated. "E".

Oct 23: Output waveform - Mode 7 (Narrow DX).

Output filtering and transceiver EQ
Cleans up some high frequency IMD from the Hilbert Clipper, and provides high & low frequency boost to help compensate for transceiver filtering. Running an outboard speech processor is always a compromise, as the accurately controlled output level then goes through the transceiver filtering, altering the frequency and phase response of the nicely limited audio. The output EQ tries to compensate for the bandpass effect of the transceiver audio and RF filtering. Eventually it will be made adjustable. This digital process is assisted by the hardware EQ that adds a fixed amount of boost at low frequencies. This helps improve the dynamic range of the DAC, and helps avoid low frequency digital clipping.

Oct 06: The two IIR filters used for EQ have both been upgraded to the new pipelined IIR code. The bass EQ filter uses the 32-bit version for lower noise.

Oct 29: A separate EQ mode has been added for AM use. It is switchable via the Select menu. At this time it is not adjustable. OBSOLETE.

May 06: Output EQ modes expanded. Choices are Disabled, SSB1, SSB2, AM. Disabled has higher output level, as it needs no headroom. This selection will be expanded further.

Output gain compensation
This switches audio levels to compensate for different filter gains and processing levels, to maintain constant peak output.

DDS test tones
Test tones are useful for testing levels and distortion performance.

At this stage, levels are only approximately set to -6 dBFS unless noted otherwise.

An assortment of tones are available + white noise:
Single tone sinewave: For testing CW power on SSB.
2-tone sinewave: For measuring IMD with a spectrum analyzer. Changed to 2 kHz separation 21/07/2014.
Triangle modulated sinewave: For checking transmitter linearity with a scope. Distortion is far more visually obvious on this than on a 2-tone waveform.
Swept tone flatness test: For checking output EQ matching to transceiver. Level is about 6dB below limiter output.
Pulsed sinewave tonebursts: For testing transmitter tuning without overheating. 5 cycle bursts of 1 kHz sinewaves, 5 mS long, 50 mS apart. 10% duty cycle.
Impulse response test signal: Single sample impulses, used for testing transceiver amplitude and phase response.
White noise: For testing frequency response using a spectrum analyzer.
* All of these test tones are sent through the DSP digital EQ (if enabled) and the analog EQ stages, to compensate for transceiver filtering.

Self-test mode
Uses the DDS and white noise routines to generate stepped sequences of tones and noise. Feeds the tones and noise into the processor input, for evaluating processing performance. Generates seven frequencies that are carefully chosen to minimize DDS spurs. Roughly: 86, 172, 344, 689, 1378, 2756 and 4136 Hz. Steps down through four levels in 10 dB steps, 1 second each for tone : 5 seconds each for white noise. Includes settings to freeze the current tone or noise output.

DDS distortion and spurs were initially higher than expected at -69 dBFS (3rd Harmonic of 86Hz), but this has now been improved to -82 dBFS by using a quadrature combining technique [16/08/2014]. The improved test signals make distortion measurements more accurate, because the processor's pre-emphasis and multiband compression amplify harmonics and spurii. The sinewave source may eventually be updated to an even lower distortion version, such as a DSP phase-shift oscillator.

* Self-test mode is designed for testing the DSP compressors and clippers, not as a general purpose audio generator.

Roger Beep (future option)
A Roger Beep produces a short tone to indicate end-of-transmission. When enabled, the PTT line is kept on for a short time after manual PTT release, and an audio tone is sent to the transceiver. Single tone, multi-tone or modulated tone outputs are possible. Morse code is also possible, such as the letter "K" -.- indicating "invitation to transmit". The delay is typically about 250 mS to 1 S.

S/PDIF over TOSLINK (optical digital audio output)
The S/PDIF output is sent using the dsPIC DCI (Data Converter Interface) 16-bit shift register. The shift register is clocked by the divided down CPU clock to produce the correct frequency S/PDIF bit clock. This requires a CPU clock other than the standard 40 MHz. The bit clock is now provided via Timer3 OC3, transferred to the DCI via pin 11 (RP4). This allows a wide choice of divide ratios. The DCI interface is run in DMA mode, to reduce CPU usage.

The audio is encoded and formatted into BMC (Biphase Mark Coding) using a 256 entry lookup table. This is very fast. BMC encoding generates two bits out for each bit in. A 16 bit word becomes 32 bits long after coding.
The S/PDIF preambles (sync headers), channel status bits, and parity bits are added, and the data is written directly into the DMA buffer in "ping-pong" mode. This method saves over 3% CPU time compared to the previous double buffering scheme.

No antialiasing is performed on the upsampled data, to reduce CPU load.

If recorded at the original sample rate of 11.025 kHz, the alias frequencies are automatically removed. Alternatively, the audio can be recorded at 44.1 kHz, and resampled without filtering to 11.025 kHz, as there is no content above 5.5 kHz to cause aliasing. The TOSLINK output now features EQ to improve the sound quality, so it no longer has accurate peak control.

An interesting observation about the S/PDIF format is that the even parity and BMC coding always results in the last BMC parity bit being a logic 0. This means the signal polarity never changes, as can be observed on an oscilloscope. This also means the parity doesn't need to be calculated. Instead, the first BMC parity bit can just be made to compliment the preceding BMC bit! This further speeds up encoding.

The S/PDIF output is proving invaluable for testing audio processing performance.

June 25, 2015: Main CPU clock speed has been dropped back slightly to decrease S/PDIF clock speed to fix an audio glitching problem when feeding a C-Media USB sound device. These glitches occurred despite the S/PDIF clock being within 1% of the correct value.

July 06, 2015: The S/PDIF monitor output and headphone output now have lowpass shelf filtering added to improve sound quality. Currently the filtering consists of a 500Hz -4dB high shelf, followed by a second order 4.5kHz lowpass filter.

November 15, 2016: The S/PDIF monitor and headphone output lowpass shelf filter has been disabled, to make audio testing easier during firmware development.

PTT state filtering and click suppression (level ramping) (added 11/06/2015)
The DSP Accumulator is used as an integrator to count up if the PTT is activated, and down if it is not activated. The Accumulator saturation function limits the values and prevents overflow. When the accumulated value is positive the unit switches to transmit mode. When the value is negative the unit switches to receive mode. The value is also used as a multiplier to fade the transmit and receive audio. This neatly filters the PTT action, and provides click suppression at the same time producing very smooth switching. The fade period is about 23mS.

Transmit timer and Watchdog timer (future options)
Since the PTT line is controlled by the CPU, a transmit mode timeout is a wise option. A software timer keeps track of how long the unit has been in transmit mode. If no audio is present, the timer counts faster.
After a set time, the PTT output is temporarily disabled to shut off the transmitter. Releasing the PTT switch resets the timer.

In over two years of operation, the functional software has never locked up in transmit mode, despite some serious bugs.

A watchdog timer keeps track of whether the CPU is executing code. If the CPU locks up for any reason, the watchdog timer times out, and the CPU is reset. The CPU then starts again normally. This is a safeguard against malfunction due to ESD, intense RF interference or power supply glitches that could cause CPU errors.
 

Receive Mode

Input overload is indicated by lighting the red LED (LED 8 - the last one). This means the radio volume level is too high, and should be reduced.

FIR oversampling filters (as above)

DC blocking (as above) [Replaces IIR highpass filter - DC blocking has superior overload performance during impulse noise.]

Receive EQ
Typical old style SSB radios have a steep drop in audio response below about 400 Hz in SSB mode, due to the crystal IF filter response. This can be corrected to some degree by low frequency boost. The required amount of boost is very high - over 16 dB. On AM or FM mode this boost must be disabled, or greatly reduced.

Nov 08, 2014: Low frequency EQ boost significantly flattens the receive frequency response on a typical transceiver. This improves the perceived quality of received audio, and allows much better readability at lower volume settings with typical household ambient noise levels.

Nov 08, 2014: This IIR biquad filter can be programmed with a variety of response curves. Currently, it is a 120 Hz peak filter with a Q of 1.2 and a gain of 20 dB.

Noise Limiter (future option)
A Hilbert Clipper could be placed here to act as a noise limiter. Since a Hilbert Clipper produces minimal harmonic distortion, it will sound better than a traditional audio noise limiter (more like an IF clipper).
Will help protect the Rx AGC (Volume Limiter) from impulse noise. Any distortion produced by this stage will be reduced by the LMS Denoiser, when active. Clipping will be indicated by lighting one of the front panel LEDs.

Phase Rotator (future option)
It may be worth adding a Phase Rotator after the Noise Limiter, to disperse noise pulses after limiting to lower their peak level.

LMS Denoiser
Based on an "Adaptive predictor", this filter reduces random noise on the signal by adapting to pass the frequencies in the signal with the greatest power. LMS stands for "Least Mean Squares", which is the adaption method used. It has three adjustments: delay, beta (error gain) and leak (coefficient decay). The standard LMS design is modified by controlling the beta value and audio input level in an AGC loop, to reduce clipping caused by excessive enhancement.

This AGC action is indicated by lighting the first orange LED (LED 6).

The output is taken from y(n).

Rx AGC (Volume limiter)
A low gain audio compressor with adaptable recovery time, allows weak signals to be heard more easily and prevents blasting on strong signals. Using "Delayed Recovery" provides rapid recovery from audio peaks and noise impulses, while keeping distortion levels low.

This AGC action is indicated by lighting the last orange LED (LED 7).
 

Configuration saving
To avoid the need for resetting levels each time the processor is powered up, the settings can now be saved into Flash memory.

On startup, the Flash memory is scanned for the start of the data block. The start is marked by the number 12345. The starting address is then saved in RAM for later access, and the data is read into RAM. During save operations, a simple wear levelling scheme adds an offset to the starting address, incrementing the page of Flash that the settings are saved to each time. After the new page is written, the old page is erased. Bounds checking sets the page back to the start of free Flash when the last block is reached. At this stage no verify is done, but this may be added later.

The dsPIC33FJ64GP802 has a specified endurance of more than 10,000 erase/write cycles. Together with the wear levelling that spreads the writes over about 30 Flash memory pages, this should allow more than 300,000 saves before device failure.

The DSPIC33FJ128GP802 can be used instead, and has twice the FLASH memory. This would extend the FLASH endurance, and allow more complex firmware to be loaded for future updates.
 

Currently, these settings are saved:

Global settings: (always saved regardless of mode)
Receive EQ (bass boost).
Utility mode (narrowband settings).
Input level.
Output Level.
Phase Rotator settings.
Microphone Flatness setting.
Output compressor on/off.
Hilbert clipper on/off.
Output EQ setting.
Default startup mode.
Virtual clipping level.

Settings saved individually for each mode:
Input EQ setting (pre-emphasis).
Multiband gain settings. Currently there is no way to adjust this setting.
Multiband level settings. Currently there is no way to adjust this setting.
Output compressor gain setting.
Hilbert clipper drive level.

During firmware updates, all settings will revert back to the defaults.
 

• The transceiver EQ is not user-adjustable. [Mar 21: Some EQ settings have been added - more to follow.]
• The Hilbert Clipper adjustment was overflowing and malfunctioning if adjusted too high. [Fixed 12/12/2013.] Still has a saturation problem. [Fixed 6/01/2016]
• Self-test tone 1 clips in the OpAmp stage when Mode 0 (Bypass) is selected. [Fixed 27/03/2014. Output level set now works in Mode 0 (Bypass)]
• Output level control is non-functional in Mode 0 (Bypass). [Fixed 27/03/2014.]
• Mode 1 and mode 3 frequency response was not flat, due to EQ changes made to improve other modes. [Fixed 26/09/2014.]
• Self test mode was conflicting with main DDS modes. Main DDS modes now take priority. [Fixed 06/05/2014.]
• The Phase Rotator centre frequency adjustment was working in reverse, as a larger coefficient caused a decrease in frequency. [Fixed 05/09/2014. Changed to presets 20/09/2014.]
• User settings couldn't be saved (if power is lost the user adjustments were lost also). [Implemented 29/07/2015.]
• When normally adjusted, bypass mode level is too low to drive transceiver to full output. [Now boosted by 6dB.]
• The 300 Hz phase rotator frequency configuration has too much gain and is non-flat. Coefficients need optimizing. Improved.
• The noise gating is not adjustable. (It is hoped to make this an automatic function, but a fixed level may work well enough).
• The multiband compressor EQ is not user-adjustable (pre-emphasis is adjustable though). [Microphone Flatness adjustment added 15/11/2016.]
• Each test tone has a different level.
• Two-tone audio test needs an amplitude balance adjustment.
• Two-tone audio test may need a frequency selection option.
• Other listed features need to be added.
• S/PDIF clock rate varies with dsPIC FRC CPU clock - May need a software frequency trimming option to correct for unit-to-unit variations.

Current Hardware Problems and fixes:

• Two solder joints on Display PCB 1 are very hard to reach due to component crowding. [Solder Display PCB 2 to Display PCB 1 *BEFORE* soldering the SMD display driver IC.]
• Display PCB 1 conflicted slightly with the 7-segment LED pins protruding from Display PCB 2. [Fixed in PCB260A.]
• The 3 mm bargraph LEDs are very crowded. [PCB v235 has a combined 3 mm LED and SMD LED front panel to solve this issue.]
• The display PCB hasn't been firmly positioned. [Pin header heights may need adjustment and spacers, but works well as-is.]
• The extruded aluminium box has an anodized coating that prevents the PCB slots grounding the board edges. [Mount solder lugs under the rear panel screws for grounding.]
• Standard 3 mm thick acrylic front panel is 0.5 mm too thick for the headphone jack. [Recess the back surface of socket hole, or use 2 mm material. PCB 204 moves socket forward 0.5 mm.]
• Receive input gain is excessive. Needs reduction by at least 6dB more, without affecting OpAmp common-mode rejection. [Level decreased 16/05/2014.]
• Keypad backlighting has been improved. The new PCBs feature a white LED behind the buttons.
• Mini-TOSLINK and 3.5 mm audio input sockets are too close for a mini-TOSLINK adaptor to fit properly. A mini-TOSLINK to TOSLINK cable works fine.
• Large signals on the transmit ADC input were causing distorted crosstalk on the receive ADC input. [Fixed. Resistor value change and software error corrected.]
• Loss of the DSP unit negative power connection was causing the PTT output to activate. The solution to this was to use a PNP inverter for Q3. [Added 02/11/2014.]
• If used, the extra microphone wires (sometimes used for scan buttons) have to be wired with links due to lack of space available for PCB tracks.
• Front panel labelling hasn't been designed yet.
• Rear panel drilling templates haven't been designed yet.

ToDo List:

• Add menu functions to adjust various gains, time constants and filter frequencies from user input. [in progress]
• Find intuitive ways to navigate menu functions. [in progress]
• Tidy up code, use more macros to make code easier to maintain. [in progress]
• Make distortion, noise and frequency response measurements and graph results. [in progress]
• Improve noise rejection of Rx Volume limiter. [in progress]
• Write LMS filter code for Rx NR. [done]
• Fix LMS filter delay glitches at certain values. [done]
• Gain compensate output for constant peak level in each mode. [done]
• Build new display PCB and install TOSLINK S/PDIF socket. [done]
• Write code for S/PDIF TOSLINK interface. [done]
• Add a highpass filter option or a bass-cut EQ. [done] : Variable EQ 11/10/2013
• Pipeline the code for IIR bandpass filters to improve speed and efficiency. [done] 4/10/2013
• Convert the Phase Rotator to 32 bit, to reduce roundoff noise. [done] 07/10/2013
• Add a software setting to switch Hilbert Clipper modes from zero-crossing to standard. [done] 31/10/2013 [Now removed.]
• Add a software setting to modify the output EQ for AM operation. [done] 29/10/2013 Added via Select menu.
• Add an output EQ disable setting - This could be added to the AM EQ setting above. [done] 21/03/2014
• Add a user controlled compression gain setting, or software mike gain. [done] Added 02/05/2014.
• Add a swept frequency test tone, for easy EQ checks. [done] Added 06/05/2014.
• Add noisegating to dynamic pre-emphasis. Added 27/08/2014.
• Add highpass filters for Modes 1 & 2 in narrow utility modes. Mode 2 now uses the shaping filter to provide a highpass shelf.
• Add PTT click suppression (level ramping). [done] Added 11/06/2015
• Add PTT delay to compensate for audio latency. [done] Added 11/06/2015
• Add PTT state filtering.[done] Added 11/06/2015
• Add user adjustable configurations. Settings saving is now functional. Added 29/07/2015.
• Add more output EQ settings to adjust for individual transceivers.
• Add a separate AM output level setting, or a separate AM mode setting.
• Improve band isolation of multiband clipper for lower distortion. May not be possible, due to other demands on CPU power.
• Test whether the Phase Rotator can be made more efficient by using a single delay instead of multiple delays.
• Test changing the last FIR oversampling filter to IIR type to reduce CPU usage.
• Add IIR highpass filters for Utility modes to improve bass response at frequency limits.
• Standardize test tone frequencies. May not be possible, due to DDS spurs.
• Add "Roger Beep" options.
• Add a watchdog timer function. (In case of a software crash due to ESD, power glitches etc.)
• Add a transmit timeout option. (e.g. 30 seconds if unmodulated, 3 minutes if modulated.)
• Modify 7-segment LED driver software for sequential digit display, producing more useful user feedback. e.g. "E1" for Band1 EQ etc.
• Make front and rear panel drilling templates.
• Write a user manual. Setup and Troubleshooting webpage sections added.

• Add self-test functions - eg: Switch white noise into processor input to check output spectrum. [done]
• Test whether "Multiband Clipping" is practical to improve quality or loudness. [done] Works well.
• Test adding hang timers to various compressors to reduce distortion from control ripple. [done] Works well - Delayed Recovery is an enhancement of this.
• Test using a proper windowed averaging look-ahead method, instead of the current delayed look-ahead method. [done] Too sensitive to RMS level.
• Test modifying the Hilbert Clippers to become fast zero-crossing synchronized limiters. [done] Works well, reduces low frequency harmonic distortion by around 20 dB.
• Test bypassing the Output Compressor. [done] 29/10/2013 Switchable via Select menu - Changes loudness/distortion performance, decreases compression and background noise.
• Test the multiband compressor with all audio filtering placed before the compressors. [done] 30/12/2013 Seems to work well. Improves compressor action and may reduce distortion.
• Test subjective quality of shaped audio spectrums on distortion performance. [done] 16/10/2014 A dipped response around 300 Hz seems to improve quality at moderate clipping levels.
• Modify DX mode settings to improve readability and loudness: Added low bands back in, increased EQ around 2 kHz and reduced EQ above 3 kHz for better loudness. [done] 16/08/2014.
• Change processing to use Weaver mode SSB at baseband. [done] 26/11/2018.
• Test dynamically modifying multiband EQ depending on the amount of clipping occurring, to improve both quality and loudness.
• Test using a DSP phase shift oscillator instead of a DDS to improve spectral purity of tone.
• Test methods of efficiently combining the Multiband Compressor noise gate and compressor code.
• Test adding adjustable transmitter EQ by allowing direct IIR filter coefficient adjustment from a menu.
• Test adding adjustable receiver EQ by allowing direct IIR filter coefficient adjustment from a menu.
• Expand 7-segment display capability by software multiplexing, alternately flashing 2 or 3 digit alphanumeric codes for better user feedback.
• Check if IIR oversampling filters are practical to save CPU cycles.
• Test adding an IIR antialias filter for the 44.1 kHz S/PDIF output.
• Add an autonotcher to the NR filter to remove heterodynes.
• Test adding more bands to the Multiband Clipper.
• Test using an LMS adaptive notch filter as an adaptive equalizer to flatten the voice spectrum.
• Test using an LMS adaptive filter to enhance spectral contrast (pronunciation) to improve intelligibility.
• Test using an LMS adaptive filter to clean distortion from heavily clipped audio.
• Test using adaptive comb filtering for noise reduction and speech enhancement.
• Test adding a Hilbert Clipper before the receive NR filter for noise limiting.
• Test adding a Phase Rotator after the receive noise limiter for noise dispersal.
• Check if FFT/iFFT processing is a viable alternative to the filter/compressor structure.
• Test if frequency shifting the output by +300 Hz is practical to improve transceiver bass response (if implemented, this would require a Tx frequency offset of -300 Hz).

• Dynamic phase rotation. A great idea - dynamically modify the phase rotation to minimize peak level. VERY HARD to do in practice, as it required evaluation of many different phase shifts simultaneously.
• Feeding the multiband clipper with a 2 kHz split path from the compression stage. This caused excessive distortion, as there was no way to reduce clipper drive when both bands had high signal levels.
• Real RMS look-ahead signal measurement in the compressors. This caused too much gain reduction on signals with high average power, and not enough gain reduction on signals with high peaks, causing excessive distortion.

• Careful adjustment of the phase rotation significantly reduced distortion and improved loudness by improving waveform symmetry and staggering the timing of the signal peaks of different frequencies at the clipper.
• Careful adjustment of the EQ applied after the multiband compressor reduced bad sounding mid-bass distortion and allowed more clipping in the midrange where distortion is not so obvious.
• Delayed recovery in the compressor stages allowed much faster gain recovery while keeping distortion low. This makes the compressors "smarter", adapting their response to suit the waveform.
• Adding peak-hold to the Hilbert clipping stages greatly reduced harmonic distortion from low frequency clipping.
• Adding a second Hilbert clipper in a split-band approach allowed much more clipping to be used, while still maintaining good quality.
• Adding a peak stretcher and lowpass filtering to the Output compressor gain control signal allowed it to act as a low-distortion "Virtual clipper" when Hilbert clipping is disabled.
• Changing the Hilbert Clipper to operate in Weaver mode improved both the processing quality and average output power. NEW 2019 FIRMWARE ONLY.

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Adjustments

HARDWARE
Tx output level (R43/R45).
Rx input level (R25/R26). These resistor values also affect receive input common-mode rejection.
Tx analog EQ freq (C31).
Tx analog EQ shelving (R30).

SOFTWARE
LED dimming function bright/dim/dynamic - Manual bright/dim working 23/06/2013.
Save user settings to Flash memory.
Load user settings from Flash memory.
Reset default user settings.

Transmit
Eight audio processing levels 0-7. (Bypass, Singleband, Multiband Minimal... through Narrow DX.)
Analog AGC hang time.
Analog AGC freeze threshold.
HF boost EQ on/off/slope adjustment. - Slope adjustment added 20/06/2013. - Now only used on modes 1 & 2. OBSOLETE.
Dynamic pre-emphasis EQ adjustment. - Added 10/07/2013.
Phase Rotator on/off. - Added 24/08/2013.
Phase Rotator centre frequency adjustment. - Presets added 20/09/2014..
Multiband compressor on/off. - Off in modes 0, 1 and 2. On in modes 3, 4, 5, 6, 7.
Compression gain adjustment. Adjusts gain on the multiband or output compressors on modes 1 through 7. Added 02/05/2014..
Six bands of multiband compressor EQ gain adjustment. - Implemented via mode presets.
Microphone Flatness setting - scales gain on Bands 5 and 6.
Six bands of multiband compressor recovery time adjustment. - No longer required since Delayed Recovery has been implemented.
Six bands of noise gating threshold adjustment. Will be made automatic if possible.
Six bands of noise gating recovery time adjustment.
Output compressor on/off. Added 29/10/2013.
Output compressor gain (post-multiband compression). - Implemented via modes, and Input level adjust setting.
Output compressor recovery time adjustment. - No longer required since Delayed Recovery has been implemented.
Hilbert clipper on/off. Added 07/07/2013. Works on modes 1 to 5.
Hilbert clipping level. 1 = HF - Added 24/09/2013.
Hilbert clipping level. 2 = LF - Added 24/09/2013. [Disabled] LF clip level is a fixed amount above HF clip level for best results.
Tx digital EQ bypass. Added 21/03/2014.
Tx digital EQ mode setting. Added 21/03/2014.
Tx digital EQ LF peak frequency.
Tx digital EQ LF gain.
Tx digital EQ HF peak frequency.
Tx digital EQ HF gain.
Tx output level - Added 19/06/2013.
Tx headphone monitor on/off.
Alignment tones: Single tone, 2-tone, triangle modulated sinewave, pulsed sinewave and swept sinewave for linearity, EQ and level tests + psudo white noise.
Debug mode - allows testing of each processing stage to evaluate audio quality and identify problems.

Receive
Noise Reduction on/off.  (LMS adaptive filter.) Added 21/06/2013.
Noise Reduction "delay". Added 21/06/2013.
Noise Reduction "beta". (Adaptation gain.) Added 21/06/2013.
Noise Reduction "leak". (Decay factor.) Added 21/06/2013.
Audio AGC on/off. (Volume limiter.)
Receive audio level. (Volume.) - This will also affect the analog Tape/Headphone output level. (TOSLINK S/PDIF digital output is unaffected.)

DISPLAY Information
7-segment display. - Current processing mode 0-7, plus tones: 8, 9, A, b, c, + white noise: d.
7-segment display. - Menu state/options (Alphanumeric codes and graphics).
Modes are shown below.
Decimal point. - Keypress indicator.
LED multistage limiter display.
LED audio level/adjustment level bargraph.
Red LED Hilbert clipping indicator (Tx).
Red LED Input overload indicator (Rx) Added 16/06/2013. For consistency, this will be moved to the Tx/Rx indicator.
Green/Red power LED Rx/Tx indicator + Analog AGC limiting indicator (Yellow).

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Controlling the processor

Display and Buttons:

LED Bargraph:
LED 0 = Pre-emphasis Limiter.
LED 1 = Band 1.
LED 2 = Band 2.
LED 3 = Band 3.
LED 4 = Band 4.
LED 5 = Band 5.
LED 6 = Band 6.
LED 7 = Output Compressor.
LED 8 = Hilbert Clipper.

KEYPAD FUNCTIONS: (left to right)
Button 1 = Normal mode. (Resets functions to default starting positions, also resets analog input AGC to maximum gain.)
Button 2 = Select - Tx default is 0 (mode number), Rx default is 1 (volume). This is for convenience. Debug overrides both.
Button 3 = "-" minus Decrements selected setting.
Button 4 = "+" plus Increments selected setting.
 

Select Menu - For new 2019 Weaver firmware:

• Frequencies and Q listing corrected June 12, 2018 - firmware is unchanged.
• LED 0 = 100 Hz, 20 stages, Q = 0.98 Original type. Default. Smoother and rounder sounding. "Broadcast" sound. Appears to reduce clipper intermod distortion on voice.
• LED 1 = 100 Hz, 6 stages. New type. Sharper and deeper sounding.
• LED 2 = 110 Hz, 6 stages. New type.
• LED 3 = 140 Hz, 6 stages. New type.
• LED 4 = 200 Hz, 6 stages. New type.
• LED 5 = 400 Hz, 6 stages. New type.
• LED 6 = 600 Hz, 6 stages. New type. Gain is too high - tends to distort.

• LED 0 = Disabled. MAXIMUM LEVEL.
• LED 1 = SSB1 +3dB @100Hz, +3dB @4.5kHz. Updated 27/10/2014 to a 3 dB 100 Hz peak filter with a Q of 0.2.
• LED 2 = SSB2 +9dB @150Hz, +3dB @4.5kHz. Default. Updated 1/10/2016 to a 9 dB 150 Hz peak filter with a Q of 1, for improved flatness with 4kHz BW CB radios.
• LED 3 = AM 150Hz HPF (much less bass). Needs more work. Ideally this mode should cancel some of the hardware EQ, which is excessive in AM mode.
• LED 4 = SSB3 +6dB @80Hz, +6dB @3kHz. Designed to suit some amateur transceivers. Added 31/12/2015.
• LED 5 = SSB4 +6dB @3kHz. Designed to suit some amateur transceivers. Added 8/1/2016.

• Utility (Narrowband settings) LED bargraph shows number. Note: 3kHz bandwidth affects both transmit and receive.
• LED 0 = 60Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 1, Default. Utility off. Rx BW = 4kHz).
• LED 1 = 125Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 2, Rx Bw = 4kHz).
• LED 2 = 250Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 3, Rx Bw = 4kHz).
• LED 3 = 60Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 1, Rx Bw = 3kHz).
• LED 4 = 125Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 2, Rx Bw = 3kHz).
• LED 5 = 250Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 3, Rx Bw = 3kHz).

Select Menu - Old Firmware:

• Frequencies and Q listing corrected June 12, 2018 - firmware is unchanged.
• LED 0 = 100 Hz, 20 stages, Q = 0.98 Original type. Default. Smoother and rounder sounding. "Broadcast" sound. Appears to reduce clipper intermod distortion on voice.
• LED 1 = 100 Hz, 6 stages. New type. Sharper and deeper sounding.
• LED 2 = 110 Hz, 6 stages. New type.
• LED 3 = 140 Hz, 6 stages. New type.
• LED 4 = 200 Hz, 6 stages. New type.
• LED 5 = 400 Hz, 6 stages. New type.
• LED 6 = 600 Hz, 6 stages. New type. Gain is too high - tends to distort.

• LED 0 = Disabled. MAXIMUM LEVEL.
• LED 1 = SSB1 +3dB @100Hz, +3dB @4.5kHz. Updated 27/10/2014 to a 3 dB 100 Hz peak filter with a Q of 0.2.
• LED 2 = SSB2 +9dB @150Hz, +3dB @4.5kHz. Default. Updated 1/10/2016 to a 9 dB 150 Hz peak filter with a Q of 1, for improved flatness with 4kHz BW CB radios.
• LED 3 = AM 150Hz HPF (much less bass). Needs more work. Ideally this mode should cancel some of the hardware EQ, which is excessive in AM mode.
• LED 4 = SSB3 +6dB @80Hz, +6dB @3kHz. Designed to suit some amateur transceivers. Added 31/12/2015.
• LED 5 = SSB4 +6dB @3kHz. Designed to suit some amateur transceivers. Added 8/1/2016.

• Utility (Narrowband settings) LED bargraph shows number. Note: 3kHz bandwidth affects both transmit and receive.
• LED 0 = 60Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 1, Default. Utility off. Rx BW = 4kHz).
• LED 1 = 125Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 2, Rx Bw = 4kHz).
• LED 2 = 250Hz to 4kHz Modes 3-7, 60Hz to 4kHz Modes 1&2 (Wide 3, Rx Bw = 4kHz).
• LED 3 = 60Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 1, Rx Bw = 3kHz).
• LED 4 = 125Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 2, Rx Bw = 3kHz).
• LED 5 = 250Hz to 3kHz Modes 3-7, 60Hz to 3kHz Modes 1&2 (Narrow 3, Rx Bw = 3kHz).

Processing Modes: (Tx processing level)

*UPDATED 25/11/2016*
0 = Bypass - Unprocessed. Analog AGC only. Preserves input quality. Analog output EQ still on. For testing or feeding from external input for replays etc.
1 = Singleband, flat EQ. Multiband off, everything else on. No noise gating. Maintains spectral balance of input. For equalized inputs only, or manual EQ adjustment. Not suitable for normal microphone input.
2 = Singleband, voice EQ. Multiband off, everything else on. Maintains spectral balance of input, apart from Dynamic Pre-emphasis and shaping filter.
3 = Multiband, flat EQ. Good quality, no pre-emphasis. For equalized inputs only, or manual EQ adjustment. Not suitable for normal microphone input.
4= Multiband, voice EQ. Tuned for better flatness on voice. Mode 4 is now optimized for 3kHz bandwidth use. UPDATED 23/09/2016.
5 = Multiband, voice EQ. Tuned for better flatness on white noise. Slightly more bass on voice. Good loudness at increased clipping levels. Default mode.
6 = Multiband, voice EQ. DX mode. 3dB more Hilbert clipping. Still sounds OK but with some minor distortion. Loud. UPDATED 21/10/2016.
7 = Multiband 3kHz, voice EQ. Heavy DX mode. 6dB more Hilbert clipping, max bandwidth 60 Hz-3 kHz. Higher average power than modes 4, 5 and 6. Very loud with good DX quality, but high distortion.
8 = Single tone DDS sinewave - 800Hz. Peak level is -6.17 dBFS on S/PDIF date: 29/04/2014.
9 = Two-tone sinewave SSB IMD test - 1628.95Hz & 2628.56Hz. Peak level is -6.88 dBFS on S/PDIF date: 29/04/2014. Frequencies updated 03/07/2014.
A = Triangle modulated sinewave linearity test - ~2628.56Hz, with ~43Hz triangle. Peak level is -7.34 dBFS on S/PDIF date: 29/04/2014.
b = Swept frequency DDS sinewave flatness test. Added 06/05/2014. Peak Level is around -12dBFS 10/01/2016.
c = Pulsed sinewave tonebursts - 5mS long, 50mS spacing, 10% duty. Added 18/05/2014. Peak level is around -6dBFS.
d = Impulse response test signal. Added 25/11/2016. NEW. This test signal allows a transceiver frequency and phase response to be tested.
E = White noise Added 01/07/2013. Peak level is about -4.35 dBFS on S/PDIF date: 29/04/2014.

Modes 6 and 7 are DX modes that have higher distortion than other modes. Due to improvements in processing, the lower modes are only about 1dB quieter than the DX modes, but have much higher quality.

Customized Processing.
Many of the processing stages can be disabled, to customize the processing as required (although this is not recommended).

Examples:
Single band compression and pre-emphasis only: Mode 2 selected, Hilbert clipper disabled.
Multiband compression and pre-emphasis only: Mode 4 selected, output compressor disabled, Hilbert clipper disabled.
High quality, minimal processing: Mode 2 selected, output compressor disabled, Hilbert clipper disabled. (No peak control.)
High quality, minimal processing but louder: Mode 2 selected, output compressor enabled, Hilbert clipper disabled. (Good peak control using latest firmware.)
High quality, minimal processing but louder still: Mode 5 selected, output compressor enabled, Hilbert clipper disabled. (Good peak control using latest firmware.)

Controllable stages:
Input EQ (dynamic pre-emphasis) adjustable (Adds brightness and clarity to audio.)
Phase rotator on/off (Increases loudness by reducing peaks. Adds no harmonic or IM distortion.)
Multiband compressor off = Mode 1 or 2, on = Mode 3 to 7 (Increases loudness and clarity of audio. Adds no significant distortion, or artifacts.)
Output compressor on/off (Keeps the clipper drive level more constant. Adds punch.) (On newest firmware, has excellent peak control when Hilbert clipper is off.)
Hilbert clipper on/off (Increases loudness by removing unwanted peaks from the audio. Only adds IM distortion.)
Output EQ off/choose (Helps to flatten the transceiver's transmit filtering response.)
 

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Save Settings (Programming Mode)

• Press and hold button 1 (M) then press button 2 (S) once. Release both buttons. The current mode number will start flashing.
• Press button 2 (S) to save and overwrite the displayed mode with the current settings.
• OR press button 1 (M) to cancel and return to normal operation.

Currently, these settings are saved:

Global settings: (always saved regardless of mode)
Receive EQ (bass boost).
Utility mode (narrowband settings).
Input level.
Output level.
Phase Rotator settings.
Microphone Flatness setting. NEW
Output compressor on/off.
Hilbert clipper on/off.
Output EQ setting.
Default startup mode.
Virtual clipping level. NEW

Settings saved individually for each mode:
Input EQ setting (pre-emphasis).
Multiband gain settings. Currently there is no way to adjust this setting.
Multiband level settings. Currently there is no way to adjust this setting.
Output compressor gain setting.
Hilbert clipper drive level.

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Debugging and test modes

Updated 25/11/2016.

0  = Normal processor output.
1  = 1st FIR antialias filter output.
2  = 2nd stage FIR antialias filter output.
3  = 3rd stage FIR antialias filter output.
4  = DC blocker output.
5  = Dynamic pre-emphasis output.
6  = Phase Rotator output.

* Most of the filter monitoring points have been deleted to save CPU time - they no longer function.
7  = Band 1 filter 1 HPF output.
8  = Band 1 filter 2 LPF output.
9  = Band 1 filter 3 HPF2 output.
10 = Band 1 filter 4 LPF2 output.
11 = Band 1 compressor output.

12 = Band 2 filter 1 HPF output.
13 = Band 2 filter 2 LPF output.
14 = Band 2 filter 3 HPF2 output.
15 = Band 2 filter 4 LPF2 output.
16 = Band 2 compressor output.

17 = Band 3 filter 1 HPF output.
18 = Band 3 filter 2 LPF output.
19 = Band 3 filter 3 HPF2 output.
20 = Band 3 filter 4 LPF2 output.
21 = Band 3 compressor output.

22 = Band 4 filter 1 HPF output.
23 = Band 4 filter 2 LPF output.
24 = Band 4 filter 3 HPF2 output.
25 = Band 4 filter 4 LPF2 output.
26 = Band 4 compressor output.

27 = Band 5 filter 1 HPF output.
28 = Band 5 filter 2 LPF output.
29 = Band 5 filter 3 HPF2 output.
30 = Band 5 filter 4 LPF2 output.
31 = Band 5 compressor output.

32 = Band 6 filter 1 HPF output.
33 = Band 6 filter 2 LPF output.
34 = Band 6 filter 3 HPF2 output.
35 = Band 6 filter 4 LPF2 output.
36 = Band 6 compressor output.

37 = Multiband summing output.
38 = spare.
39 = Stereo I & Q Hilbert transform test on S/PDIF.
40 = Hilbert transform output I.
41 = Hilbert transform output Q.
42 = Stereo I & Q output compressor test on S/PDIF.
43 = Output compressor output I.
44 = Output compressor output Q.
45 = Stereo I & Q Hilbert clipper 1 test on S/PDIF.
46 = Hilbert clipper 1 output I.
47 = Hilbert clipper 1 output Q.
48 = Stereo I & Q Hilbert clipper 2 test on S/PDIF.
** More levels are needed here to test multiband clipper functions. Yet to be added.

Schematics

Updates & Notes

• The 2-transistor microphone amplifier circuit has very low noise and distortion.
• Aug 05, 2015: R15 deleted.
• Jun 14, 2015: C15 increased to 220n to improve audio amplifier HF stability.
• Jun 06, 2015: Q4 changed back to grounded Collector, to reduce muting clicks.
• Jan 26, 2015: D1 has been renumbered to D11.
• Jan 24, 2015: Capacitors C1 to C6b renumbered to C0 to C5. C0 is located at the mike element.
• All components have been renumbered to avoid duplicates.
• Dec 24, 2014: C5 added to improve PWM filtering.
• Nov 01, 2014: Q4 was changed to a BC549 in a normal grounded emitter configuration. This caused greatly increased muting clicks and has been changed back to grounded collector!
• May 07: Extra decoupling network added for phantom power. If phantom power is not required, R1, C10, R14 and C1 can be deleted from the circuit. C1 is then replaced with a wire link.
• May 07: Audio input coupling capacitors reduced in value to 1uF. Tantalum or ceramic SMD capacitors should be used here to avoid leakage current which can cause AGC problems.
• Nov 09: R13 increased to 220 ohms to improve PWM filtering.
• Nov 08: C9 has been added to decrease noise and improve gain slightly.
• The RF input filter has a low cutoff frequency and should be fine on all bands down to 160 metres.
• L1 has been replaced with a 1k resistor (R2) to reduce inductive noise pickup from the CPU rail.
• A 2n2 capacitor has been added across the base/emitter junction of Q2. This capacitor is now connected from base to ground.
• C0, C2 and C4 are ceramic for low inductance at RF.
• The current drain of the entire version 2.0 PCB was around 78 mA, this version (2.1) with the audio amplifier and LEDs is higher - but typically under 200 mA.
• The 8.3 volt regulator circuit supplies the mike amp with a clean supply rail.
• A 1A 3.3V SMD IC linear regulator supplies the dsPIC.
• Some software based hardware AGC is provided via dsPIC pin 6 feeding current to Q1 to prevent microphone input clipping.
• The analog microphone AGC has now been modified to use PWM from Timer2 OC2 for better linearity.
• The AGC filter has been replaced by a 2nd order PWM filter with a cutoff frequency of about 1 kHz (this filter may be modified further for better performance). The attack and recovery filtering is performed in software. The addition of gain "freezing" has made the analog AGC very stable, and the reduced background noise during speech pauses makes the audio sound very good.
• If using the older firmware from before 19/08/2013, C6 must be 10uF.
• D11 has been found to reduce audio distortion when using a bipolar transistor as a variable attenuator element for audio.
• A good quality unidirectional electret microphone is recommended to provide a flat input response and to reduce background noise. Unidirectional microphone inserts require mounting so that the rear of the element is also open to sound. In most commercial mikes, this is accomplished by mounting the element on supports and enclosing it with wire mesh or equivalent. A small baffle of either a disc or a tube can increase mike sensitivity by lengthening the acoustic path from the front to the rear of the mike element. If a tube is used, it normally has slotted side openings to avoid resonance effects.
• The multiband compressor automatically equalizes the audio spectrum to some degree, making the microphone frequency response less important.
• A TDA2003 audio power amp has been added to the PCB to enable speaker drive (similar to LM383).
• Speaker amp muting has been added.
• The 220 ohm audio amp feedback resistor has been moved to the PCB underside to improve thermal design.
• The 220 ohm and 2.2 ohm resistors have since been changed to SMD types to increase display PCB clearance. They can also be standard resistors, surface mounted on the PCB track side.
• A heatsink was added to the TDA2003 by soldering a copper tab to the PCB and bolting it to the IC. With a 4 ohm speaker and heavy audio drive, the IC temperature reached about 54 degrees C during testing. No insulator is required, the tab is grounded.
• The 100n input coupling capacitor on the TDA2003 has been placed on the PCB track side due to space reasons. It is now an SMD type.
• The speaker remains absolutely silent during transmit - no sign of RF breakthrough or supply ripple rejection problems. (Tested on 27 MHz only.)
• Tx audio AGC is set to full on during receive mode to mute the microphone input and prevent any audio crosstalk from large microphone inputs during Rx.
• Ferrite "beads on leads" are used on the speaker output and DC input to help filter RF, as they have very low resistance.
• The extra ferrite bead on the jumper from the speaker socket ground to the audio IC ground prevents it acting like a parallel tuned circuit with the 10n capacitor. (This is a precaution for Amateur radio use, and was not used on the prototype.)
• The mike input RF choke has been removed, as it was picking up strong 3 kHz noise from the CPU supply rail via induction. It has been replaced with a 1k resistor, and the CPU supply rail bypassing has been improved.

• This v2.1 processor should work with any transceiver without requiring an isolated power supply. If an isolated power supply is used, the power supply negative connections must be linked for correct operation.
• The ADC input is on dsPIC pin 2 (Tx)and pin 3 (Rx).
• The dsPIC digital schematic (Part 2) is shown below:

Updates & Notes

• Feb 07, 2016: Schematic updated to 7.4e. C58 position corrected.
• Sep 08,2015: Schematic updated to 7.4d. R33 increased to 10k to prevent distortion at high output level settings.
• Jun 15, 2015: C66 added to U1a to prevent RF breakthrough into headphone output. Pads are already on the PCB for this capacitor, but it is not marked on the overlay.
• Jun 12, 2015: Schematic updated to 7.4. D12 (3.3V SMD Zener) added to protect dsPIC from overvoltage on the incoming PTT line.
• May 19, 2015: Schematic updated to 7.3. C36 increased to 470uF 5x7mm.
• May 16, 2015: C57 increased to 470uF 5x7mm to improve low frequency headphone response and improve clearance to Display PCB 1.
• Jan 26, 2015: Diodes renumbered.
• Jan 24, 2015: Schematic updated to 7.0. C54 replaced with D9. C62 and C63 added.
• Jan 06, 2015: Schematic updated to 6.9. SW1 to SW4 renumbered. C51 noted.
• Dec 23, 2014: Schematic updated to 6.8. Capacitors renumbered.
• Dec 21, 2014: New LED keypad backlight added to main PCB. Capacitor added to help protect Q7. C30 and C31 designators swapped. R82 designator corrected. C54 (now C56) reduced to 1uF.
• Dec 20, 2014: C46 and C48 reduced to 100p to improve ground loop cancellation at high frequencies.
• Dec 06, 2014: C20 position corrected, some text errors corrected.
• Nov 27, 2014: Revision 5.9: All components have been renumbered to avoid duplicates. Text has been updated to refer to NEW version only..
• Nov 22, 2014: Revision 5.8: Both the LED Data and LED Clk outputs of the dsPIC have series resistors for safety. These are now shown on the diagram. Values are 1k.
• Nov 02, 2014: Revision 5.7: The PTT circuit has been improved, by using a PNP transistor inverter. Both supply rails must be present to allow base current to flow and turn Q7 on, before the PTT will activate. If Q7 base current is not present, no voltage will be present at the gate of Q6, leaving it off.
• May 22: Revision 5.4: Changed U2b OpAmp resistors to commonly available values: 7.5k, 15k and 30k.
• May 16: Revision 5.3: Changed U2b OpAmp resistors to fix common mode rejection problem. The AC voltage gain from pin 5 should now be 2.5. The AC input voltage gain is now 0.5, the DC voltage gain is now 2.0, the non-inverting input DC bias is 3.3V and the OpAmp DC output will be 6.6V.
• May 06: Changed 10k EQ resistor on U1b pin 6 to 4.7k, to increase bass EQ. Schematic updated. Some transceivers can't handle the increased EQ and will distort. See "Troubleshooting" below.
• Jan 10: Corrected schematic to match PCB connection of U2b pin 5, 10k resistor to filtered 3.3 volt supply rail.
• Jan 05: Bypass capacitor added to DC input. This capacitor was already marked on the PCB, but not the schematic. An SMD type is used for best RF performance.
• Dec 28: Microphone ground line inductor added. A ferrite bead-on-lead must be used for low resistance. A wire link can be used instead if RF filtering is already adequate. PCB versions before v210 include a link between the pads. This link will need to be cut if an inductor is installed.
• Dec 28: Renamed L5.
• Nov 27: Capacitors in the above diagram have been renumbered.
• Nov 09: 27k resistor added to U2b pin 6, to bring the output voltage of U2b closer to half supply voltage. This resistor is now 7.5k! May 22, 2014.
• Nov 09: ADC current limiting resistors increased to 1k to reduce crosstalk.
• Nov 09: ADC RF bypass capacitors reduced to 1n.
• Nov 09: Schematic ADC pins 23 & 24 swapped, to match PCB.
• Oct 13: EQ capacitor C31 changed from 100n to 47n. (Best value will vary depending on transceiver.)
• Sep 24: Extra 330pF capacitors have been added for RF suppression on U2a.
• Sep 27: Extra 47R and 10n added to headphone output for improved RF suppression. An SMD capacitor was used to save space.
• The CPU digital supply bypass capacitor C36 has been moved closer to the CPU.
• The CPU clock source now uses the dsPIC internal FRC clock to reduce the component count.
• The PTT switching is resistively isolated to avoid ground loops, and defaults to Rx if the CPU is in reset.
• The PTT switching includes DC Rx line switching required by some CB transceivers.
• NOTE: This Rx switching relies on PTT voltage to switch Rx on. If the PTT line has no voltage present, Rx switching will not switch on.
• The 3.3V and 8V regulators have been upgraded to 1A SMD units to reduce operating temperature and save PCB space.
• Transformerless ground loop cancellation has been added to enable normal power supply connections and avoid coupling transformers.
• R49 is for bias purposes, to keep U2a output near 6V.
• R48 and R50 scale the ground loop signal by half, because the OpAmp has 2x non-inverting gain.
• R44 keeps the capacitors correctly polarized in case the microphone output is disconnected.
• Some transceiver low frequency EQ has been added to the DAC OpAmp stage.
• If a flat output is required, the EQ components can be removed, and the U1b OpAmp circuit modified for best dynamic range (by changing resistor values). This will also require disabling the DSP digital output EQ (there is a SELECT menu option for this).
• Two display board add-on connectors have been included for the LED board.
• The four settings jumpers from version 2.0 have been removed.
• Four pushbuttons have been added for user input.
• Hardware I/O for Receiver audio processing has been added.
• Receiver audio input amp has been converted to differential mode to reduce ground loop effects.
• ISP programming interface was simplified to allow re-use of I/O pins for keypad input.
• SIP OpAmps have been used instead of DIP types to save PCB space - Two M5218AL OpAmps replace the TL074.
• The M5218AL OpAmps can drive a headphone output with a current of up to 50mA.
• A differential OpAmp stage is used on the dsPIC DAC output, to provide better noise rejection.The resistor network needs modification to increase common-mode rejection.
• A 27k resistor was added to the ADC bias network to compensate for the voltage drop through the 10k feed resistors, and keep the ADC closer to half-scale.
• Chassis ground has been DC isolated by PCB design - RF grounding is via small capacitors.
• Q8 has been replaced by a bipolar transistor to ensure proper switching at CPU voltage levels.
• PTT ground input has been resistively isolated from the DSP board ground to further reduce ground loops with certain mike wiring.
• More RF bypass capacitors have been moved to the PCB underside - They can be normal or SMD capacitors.
• Monitor/Tape output has been added via a front panel headphone jack. (Tape output is non-muted, and has similar levels for Rx and Tx audio.)
• The speaker output socket has been isolated from local ground to further reduce ground loops (it is grounded at the audio amp side of the PCB instead).
• The audio amp pinout has been redone to improve device mounting.
• The rear sockets have been regrouped as: Power, Rx In, Tx Out, Speaker, Mike.
• A zener diode has been added to the Q6 gate to prevent failure due to ground voltage spikes. (This actually happened when the unit was powered from a switchmode plugpack, due to mains leakage current!)
• OpAmp bias voltages have been optimized, but some coupling capacitors have had to be added.
• A 330p capacitor was added to the feedback path of U2b to prevent OpAmp oscillation.
• Additional RF bypassing capacitors have been added to the Rx audio I/O to improve performance in harsh RF environments.
• The ground link jumper and the EQ jumper have both been removed to simplify the circuit and save space.
• Decoupling the OpAmp bias supply from the CPU rail has reduced the sample clock breakthrough on the audio output by 12 dB.
• More zener diodes have been added to the PTT switching MOSFETs for improved protection against overvoltage. Due to lack of space they must all be mounted on the PCB underside. SMD devices would be the best choice. The commercially made PCB provides space for these through-hole zener diodes. SMD devices are also catered for.
• The rating of the Polyswitch device has been reduced to 1.35 A for increased protection. The alternative SMD version is rated 1.1A.
• The headphone output resistors have been reduced to 220 ohms for improved level matching - they may need to be reduced further.
• Some I/O lines have been reconfigured to allow more flexible S/PDIF bit clock generation. The PCB layout has been updated.

• All components have been renumbered to avoid duplicates.
• Dec 27, 2014: A NEW keypad backlight LED has been added to the main PCB. Its close proximity to the front panel greatly increases brightness.
• Dec 21, 2014: LED ordering error corrected. PCBs were unaffected.
• The LED DISPLAY PCB is now built - It consists of TWO PCBs. Display PCB 1 has the control circuits, and Display PCB 2 is the LED panel.
• The commercially made PCBs include pads for both through-hole and various size SMD LEDs. (untested)
• The 16 LED driver chip is a very cheap ($0.99) SM16126D (SMD).
• The dual colour POWER indicator LED doubles as a Rx/Tx indicator (Green/Red), and analog AGC indicator (Yellow).
• A TDA2822 Stereo Audio Power Amplifier IC has been successfully tested as an efficient dual colour LED driver.
• The TDA2822 was chosen for these reasons:
• Low cost (around $0.20 ea. in lots of 10 on eBay from G&C China) - That's cheaper than many small-signal transistors.
• Bridged output means lower current drain in the LED current limiting resistors and lower temperatures.
• Simplicity - The discrete design used 3 transistors and double the current of the IC version.
• Ruggedness - The IC version uses supply rail current limiting, and will easily survive a continuous short to ground on either output.
• The display includes a software DIMMING function via the LED drivers.
• The 7-segment LED display wiring has been (inverse) segment ordered to make programming easier.
• Decimal point drive has been added (total display LEDs now 18).
• S/PDIF TOSLINK Optical Digital Audio output hardware has been added via the dsPIC DCI data line. This allows the DCI to send data from a RAM buffer via DMA without any CPU overhead.
• ALL I/O pins are now used.
• (Old) An LED backlighting option for the keypad has been added to Display PCB 1, but appears insufficient.
• (Old) Pads have been added to allow the keypad backlight LED on Display PCB 1 to be fed from the dimming circuit, but these are unused.
• Features, components and pin connections may change, but are mostly finalized.
• The LED display board schematics are shown below:

LED Display boards 1 & 2 - Some values may need adjustment to improve LED brightness matching.
Bypass capacitor values are not critical. Only two bypass capacitors were fitted to the prototype.

Caution: Since common-anode LEDs are used, (except for D100) any LED shorts to ground will cause LED failure. The latest Display PCB 2 has a small link that can be cut to insert a safety resistor in the 7-segment LED supply rail. This safety resistor has been omitted from later prototypes. The commercially made PCBs have very well controlled spacing that protects against short circuits when assembled.

In System dsPIC Programmer

If using an add-on printer card, put the card's I/O address under "Programmer Connection" "Advanced" "I/O Addr.".
The card used in this example was a "Redchief" LPT COM1 COM2 serial port expansion card, with LPT I/O address 0xE800. This card works well, as it has a good voltage output and very high value pullup resistors allowing it to work with the revision 1.2 circuit below.
If using standard on-board peripherals, select the LPT number in "Port", instead.
 

Programming Interface Schematic

Parallel Printer Port. (From Wikipedia.)

The dsPIC needs to be powered up to use this programming method. If this circuit fails to communicate with the dsPIC, the port voltage might be a bit low. Try increasing R5 and R6 to 680 ohms or even 1k. These resistors need to be low enough in value to discharge the cable capacitance, and high enough in value to allow adequate voltage. Some printer ports have low value pullup resistors that can upset the voltage levels on the Data line (3). These types of printer ports require buffering of the pin 10 Data In line (ACK) to prevent it pulling up the voltage level on the Data line, and to fully drive the Data In line. Using an LM311 has been proven to work even with troublesome printer ports.

Programming

PICPgm Settings

For initial programming:
Select "Program Code Memory (FLASH)".
Deselect "Program Data Memory (EEPROM)".
Select "Program ID Locations".
Select "Program Configuration Bits".

If Verify fails:
Deselect "Program ID Locations".
Deselect "Program Configuration Bits".

For later updates:
Deselect "Program ID Locations".
Deselect "Program Configuration Bits".

Configuration bits only need updating if the hardware configuration changes, which is very unlikely.

Transformerless ground loop cancellation

A common way to prevent this happening is to use a coupling transformer. The transformer allows differential current flow, but prevents common mode current flow.

But transformers have some problems:

• Limited frequency response.
• Expense.
• Space requirements.
• Increased weight.
• Magnetic hum pickup.
• Added distortion.

The audio is attenuated by a voltage divider for two reasons:

• The audio output is several volts amplitude - way too high for a microphone input.
• The OpAmp noise level is too high for direct connection to a microphone level input.

Since a receive mode DSP input has been added, additional ground loops are possible. A differential input has been used in the Rx ADC preamp circuit to minimize this.

The ground loop cancellation circuit relies on a low-resistance connection to the transceiver microphone ground. A ferrite bead-on-lead has low DC resistance, and has been added to the microphone ground line for better RF filtering.

S/PDIF via TOSLINK

• The audio has already been digitized.
• TOSLINK is immune to RF and ground loops.
• TOSLINK is immune to spikes from the AC mains or lightning.
• TOSLINK connectors are immune to corrosion, poor contacts, and metal fatigue.
• Since receive mode has been added, the digital output is handy for recording.
• PC power supplies and motherboards can generate noise, which can severely affect audio inputs.
• The digital output doesn't feed the speaker, so the level can be fixed.

BMC is a simple two-frequency representation of binary data. A 1 is encoded as a 10 or 01 output, and a 0 is encoded as a 11 or 00 output. The polarity of the new output is the opposite of the previous last output bit. An 8 bit byte becomes a 16 bit word after BMC coding.

The audio is encoded by using a 256 word lookup table to convert to BMC. The data is then written to a dual DMA memory buffer for sending to the dsPIC DCI unit. The SPI unit could also work for this application, and may run at higher clock speeds. Since these are hardware shift registers, they can send high speed 16 bit serial data words with no additional CPU load.

Software encoding was chosen because the special "preamble" sync codes that S/PDIF uses are not in BMC format - which makes hardware encoding difficult. Commercial S/PDIF encoders exist, but they need a clock reference that is not easily available now that the dsPIC has been switched to the internal FRC clock.

Ideally, the S/PDIF would be at 11.025 kHz stereo, but most consumer devices will not recognize this sample rate. The S/PDIF output is instead upsampled (without antialias filtering) to 44.1 kHz stereo with good results. The USB sound device being used works well at this rate. The CPU is now being clocked at 39.5136 MHz, and the correct S/PDIF bit clock is generated by Timer3 Output Compare routed to the DCI serial clock input by setting both lines to use remappable pin RP4.

Since there wasn't room to fit a full size TOSLINK socket on the rear panel, a mini-TOSLINK output connector was built from a 3.5 mm socket and a red LED. It works very well. A mini-TOSLINK to TOSLINK cable connects the DSP unit to the S/PDIF optical input of a cheap (<$15) C-Media USB sound device.

At this stage, the dsPIC internal FRC clock appears to be stable enough for reliable S/PDIF / TOSLINK functioning, but clock frequency errors or drift will cause recorded audio to change pitch slightly.

Recordings are best done at 11.025 kHz, so the recording system performs antialias filtering. Another way is to record at 44.1 kHz, and resample to 11.025 kHz afterward. This does not require antialias filtering, as there is no audio content above 5.5 kHz. The dsPIC software currently does not feature antialias filtering on the S/PDIF resampling, to reduce cpu load. The transmit monitor output now features some EQ to improve the sound quality, and no longer has accurate peak control like the transceiver output does.

Design choices

The DSP Speech Processor needs several connectors to attach the I/O.
The microphone I/O is designed around 8-pin Mini-DIN connectors.
Fly leads can be built to adapt to any kind of microphone system.
Receive audio I/O is via 3.5 mm phone plugs.

An audio amplifier IC has been added to the board. Point-to-point wiring connects the amp to the Speaker socket. If speaker drive is not required, the audio power amp can be left out of circuit, and the Rx Out taken from the OpAmp, but this will disable the transmit muting hardware.

A front panel stereo headphone socket has been added to the PCB. It can also be used as a TAPE output for recording both sides of a QSO. The audio is also available via the digital optical TOSLINK output. The optical output has the advantage of being immune to RF and ground loop effects. It carries processed audio on the left channel, and unprocessed audio on the right channel - for both transmit and receive.

PCB I/O consists of:

Right Mini-DIN (back view).
Microphone audio input.
PTT input.

Left Mini-DIN (back view).
Processed audio output.
PTT output.

PTT Rx switching is only required by some older CB radios.

Rx audio in (3.5 mm).
Rx audio out (3.5 mm).
3.5 mm Socket Tech Drawing - these were the slimmest 3.5mm sockets available anywhere. Known as Lumberg 1503-07:

Digital optical output (TOSLINK S/PDIF) located on Display PCB 1 (3.5 mm Mini-TOSLINK, built from an LED mounted behind a 3.5 mm socket).

DC Power (2.1 mm). A long shaft plug may be required due to rear panel clearances.

Bill of materials

Bill of materials (parts list) with component location grid (thanks to Frank G0GSR)

Special Parts

Two different dsPICs can be used in this design:
The SPDIP dsPIC33FJ64GP802-I/SP can be purchased online from Microchip Direct (minimum order is 2). This was the original design choice.
The SPDIP dsPIC33FJ128GP802-I/SP costs slightly more (about $1 extra), and has twice the FLASH memory. This could be useful for future firmware upgrades, and is compatible with the original design.

DSPIC33FJ64GP802-I/SP
Lead Count:    28
Package Type:    SPDIP
Temp Range:    -40C to +85C
Packing Media:    Tube
http://www.microchipdirect.com/ProductSearch.aspx?keywords=DSPIC33FJ64GP802-I/SP

*OR*

DSPIC33FJ128GP802-I/SP
Lead Count:    28
Package Type:    SPDIP
Temp Range:    -40C to +85C
Packing Media:    Tube
http://www.microchipdirect.com/ProductSearch.aspx?keywords=DSPIC33FJ128GP802-I/SP
 

SM16126D SOIC-24 LED Driver:
http://www.ebay.com/itm/2pcs-16-channels-constant-current-LED-drivers-arduino-SM16126D-STP16CP0-TLC5926-/111902604117

Miniature electrolytic capacitors are available here:
http://stores.ebay.com/CAPPRO/Electrolytic-Capacitors-/_i.html

Ferrite "Beads On Leads":
http://www.ebay.com/itm/50-Ni-Zn-Ferrite-Bead-Core-Lead-3-5x6-0mm-Taiwan-/140329107881

DC power socket:
http://www.ebay.com/itm/10pcs-DC-Power-Supply-Female-Jack-Socket-5-5-x-2-1mm-Barrel-Type-PCB-Mount-/180855002074

Miniature 3.5mm sockets (called "Lumberg 1503-07" sockets - widely available):
http://www.ebay.com/itm/10-Pcs-DIP-PCB-Mount-4-Pins-Female-3-5mm-Stereo-Audio-Jack-Socket-/321099207333

Alternative 8-pin Mini-DIN sockets: * These have been checked, and physically fit the PCB, though they have slightly different dimensions. This supplier has cheaper postage.
http://www.ebay.com/itm/311323907000

The 8-pin mini-DIN sockets: *
http://www.ebay.com/itm/5-Mini-DIN-Connector-Socket-Female-8pin-PCB-pin-2008-8-/140431326834

The 8-pin mini-DIN plugs: *
http://www.ebay.com/itm/5pcs-Mini-DIN-Connector-Male-Plug-8-pin-1102-8-/140431341209

* It is important to purchase mini-DIN plugs and sockets that match the PCB pad outline. Connectors from other suppliers are not guaranteed to fit the PCB. Also, plugs and sockets bought from different suppliers may have differing connector pin configurations, and may not fit each other.

PCB mount right-angle pushbutton switches:
http://www.ebay.com/itm/10-x-Tact-Switch-6-6mm-6-85mm-50mA-12V-DC-Right-Angle-/260528274819

7-Segment common anode white LED display:
http://www.taydaelectronics.com/led-displays/7-segment-1-digit/led-display-7-segment-1-digit-0-56-inch-common-anode-super-white.html

AMS1117 1A 5V SMD Regulator IC:
http://www.ebay.com/itm/10-pcs-AMS1117-5-0-5-0V-1A-Voltage-Regulator-SOT-223-/250725260822

AMS1117 1A 3.3V SMD Regulator IC:
http://www.ebay.com/itm/10-pcs-AMS1117-3-3-3-3V-1A-Voltage-Regulator-SOT-223-/250742607708

1.35A through-hole Polyswitch devices:
http://www.ebay.com/itm/20pcs-RUEF135-1-35A-30V-PPTC-PolySwitch-Resettable-Fuse-/321412059105

1A SMD Polyswitch device:
http://www.ebay.com/itm/10Pcs-Polyswitch-SMT-SMD-Resettable-Fuses-1812-1A-/180640196871

Construction Notes

Front panel drilling template for PCB257

Rear panel drilling template

Forming the front panel

Drill all the panel holes before rounding the panel. This provides the required mounting holes, and helps assure better alignment.

The template below allows the construction of a wooden jig. Only one pivot point is necessary. When the first end of the panel is rounded, undo the mounting screws and flip the panel over end-for-end instead of rotating it. This helps provide better left-right symmetry in the finished panel. Then fasten the panel again with mounting screws and round the other end.

Print this template at 600dpi for correct scaling.

Dark grey window tint was applied to the front surface to improve LED contrast. This also hides any scratches on the polycarbonate. It may also be possible to use the adhesive window tint to hold button labelling in place. If acrylic sheet is used, never clean it with alcohol as this will discolour the material.

For commercially made PCBs:

The optional SMD protective resistor on Display PCB 2 (marked "R") is best ignored.

Attach Display PCB 1 to Display PCB 2, making sure the PCB edge pads are aligned. Display PCB 2 should be vertically aligned with Display PCB 1 so that the top row of the 7-segment LED pins is just clear of the top edge of Display PCB 1. Tack solder Display PCB 2 to Display PCB 1 at the corner pads first, in case realignment is needed. Be careful not to stress the PCB edge pads, or they may break off the board making reconnection very difficult. Avoid dropping the boards at this stage, or the edge pads can break off! Also use a small amount of solder to bond parts of the large top pads of Display PCB 2 to the Display PCB reverse side for mechanical strength.

Display PCB Alignment: PCB1 = Display PCB 1
A: Ideal alignment for 1.6 mm thick PCBs. 7-segment LED is mounted almost touching the box top, 3 mm LEDs have very little space. LED pins clear the PCB1 edge OK.
B: Actual alignment used on the first prototype. 7-segment LED is mounted almost centrally. PCB1 is mounted slightly higher. 3 mm LEDs have a little more space. LED pins clash with PCB edge, requiring a chamfer.
C: Ideal alignment for 1.0 mm thick PCBs. 7-segment LED is mounted almost touching the box top, 3 mm LEDs have a bit more space. LED pins clear the PCB1 edge OK. Use this for PCB-260.

Using 1.0 mm thick PCB material increases the space above and below the 3 mm LEDs. This means the same design will work for 1.0 and 1.6 mm PCBs, as the 3 mm LED centre line is the same.

The commercially produced PCBs have very carefully controlled LED placement. Using 1.0mm PCBs (PCB260) they should mount like "C" above. The rear of Display PCB 1 is required to be vertically higher than the front, to enable sufficient separation of the 3.5 mm Rx In socket and the mini-TOSLINK socket. Sloping Display PCB 1 upwards slightly towards the rear allows for this.

Header pins are used in two areas, for CPU electrical connections and mechanical fixing. Gold plated pins are recommended to provide most reliable contact. Nickel plated pins were used on the first prototype, and became unreliable.

Some 2.54 mm header pins and sockets connect the signal lines between Display PCB 1 and the Main PCB:
 

Simpler PCB pin mounting The best way is to mount some straight header pins on the main board, and the header socket strips on the Display board (shown below). This avoids the need for creating a customized header pin assembly from right angled pins. Instead, the mounting pins of the socket strip are bent flat to match the pad pattern for SMD mounting. This method has been tested, and works fine.

After the pins are installed, plug Display PCB 1 into the main PCB, taking care to align all the pins. Test how well the PCBs fit the enclosure. They should align well, with no conflicts. Display PCB 1 tilts up at the rear for correct front and rear alignment.

Once the alignment is good, solder the remaining edge contacts that connect Display PCB 1 and Display PCB 2. Also solder the mechanical bonding solder pads at the top of the boards.

Next, fit the other parts to the boards. Start with the main PCB, then move on to the Display PCB components. Most parts are marked on the overlay. More details below.

Some capacitors can be either through hole or SMD. SMD capacitors have better RF performance.

Be careful to install the SIP OpAmps around the right way - they are difficult to unsolder. Pin 1 is marked with a square pad and a square outline on the overlay. Vcc is pin 8.

Use machined pin socket strips as a socket for the dsPIC chip. Mount the socket strips before the other components around them.

The TDA2003 leads need to be straightened to fit the board. Use minimal solder on the IC pins, as the pads have close spacing. They may be a tight fit. The TDA2003 needs to be mounted with its epoxy body pressed down until it touches the PCB, so the heatsink tab will clear the top of the enclosure. It only just fits.

Solder in a copper heatsink tab after fitting the IC, keeping the solder away from the IC lead pads. The tab will need one mounting hole, aligned to the IC tab hole. The tab needs to be the same height or slightly lower than the IC tab, or it will hit the top of the enclosure. Use a copper tab. Do not attempt to solder an aluminium heatsink tab, it will damage the PCB (aluminium soldering requires special solder alloy with a high melting point).

The commercial PCBs do not require fixed wire jumpers, but there are four flying leads required. One is for the Rx ADC input, one for Headphone output, and two are for the speaker output. Mount the four flying leads last, after the other components. The speaker flying leads can both have a ferrite bead added to reduce RF effects, but this is optional.

The main DSP board can be constructed first, and powered up with Display PCB 1 disconnected. The unit will function OK with no display. Before inserting the dsPIC IC, power up the main board, and check the supply rails. They should be 3.3v at the dsPIC IC. Also test the 8.3V rail for the analog circuits. Then power down and insert the dsPIC. It can then be programmed.

IMPORTANT: Never remove or insert Display PCB 1 with power applied or LED damage can result, due to short circuits.

Pads have been provided for an SMD RF bypass capacitor to be fitted to the TDA2003 audio input, if required. Currently unused.

The Polyswitch device can be either a through-hole or SMD type. It is used to protect the PCB tracks from melting in case of a short on the 12V rail. A 1.35A 30V device has been used.

The electrolytic capacitors must be physically small enough to fit the available space. This is especially important for values over 100 uF.
The smallest electrolytic capacitors are 4 x 7 mm from CAPPRO on eBay.
Use the lowest voltage rating suitable. e.g.: 16V for 12V bypass, 10V for 8V bypass, 10V or 6V for 3.3V bypass, 10V for coupling, etc.

The circuit is DC isolated from the enclosure. There are some SMD pads on the tracks for capacitor placement. The main groundplane has a 0.5 mm spacing from the edge tracks.

The programming connector is a 4-pin PCB header socket strip, fitted to the PCB underside using surface mounting, and facing the front. This avoids the need to remove Display PCB 1 to access the connector. It can then be accessed from the front after removing the front panel, or after sliding the PCB assembly out from the rear of the box. Glue the socket strip to the PCB with a dab of hot glue, to prevent the pins flexing and breaking.
See below:

Note: The 4-pin header socket does NOT align to the four resistors nearby - the right hand pin connects to the PCB ground pad. The PCB overlay correctly shows the pin locations, numbered 4, 3, 2, and 1. The commercial PCB has this marking on the overlay, and also features plated through holes to reinforce the pads.
 

Mini-DIN Plugs

Avoid overheating the pins when soldering or the plastic can melt - destroying the plug. Solder fast.

The outer jacket from some RG-6 coax fits these connectors perfectly, and keeps the wiring tidy. A dab of hot glue inside the plug shell halves acts a good strain relief. It will melt when soldering the shell halves together, producing a good bond to the RG-6 jacket.

 

3 wire electret converson

Dec 31, 2015: C1 (was across R1) has now been removed from the circuit, as the DSP EQ has been modified to work better with a flat input.

If RF feedback persists, add an RF choke of at least 1mH in series with the 2-wire output "Audio" line, close to the mike. 1mH = 1 millihenry = 1000uH = Brown Black Red.
 

Dynamic microphone amplifier

The input to the DSP Speech Processor is designed for Electret Condenser microphones. Dynamic microphones need more gain, and a lower input impedance. This circuit uses the phantom power supplied by R1 on the DSP board to run a one transistor amplifier. If Rx is used, the 4k7 resistor may need to be increased to boost the gain. Because the input of the amp loads the microphone, Rx needs to be higher than the microphone impedance.

* The 2n2 capacitors must be ceramic types, to be effective at RF suppression.

* If RF feedback persists, add an RF choke of at least 1mH in series with the "To DSP Mike input" line. 1mH = 1 millihenry = 1000uH = Brown Black Red.

* If this circuit is not located at the microphone, an additional RF choke of at least 1mH may be needed in the "Mike in" line, to suppress RF feedback. Place it between Rx and the 1uF capacitor, or between the 1uF capacitor and the 2n2 capacitor.
 

Extra RF filtering

An example is shown below:

Both the capacitor and the RFC must be mounted close to the microphone.

1mH = 1 millihenry = 1000uH = Brown Black Red.
Small value RF chokes (much less than 1mH) will be of little use to reduce RF levels on the microphone. If the RFC resonates with the 2.2nF capacitor at the radio frequency, it will make the RF voltage at the mike higher instead of reducing it!

The microphone cable shield wire must connect to microphone ground. This is to ensure good shielding against AC hum and RF.
Avoid using data cable for microphone cable, as the microphone wire is not shielded from the other wires. This causes capacitive coupling.
Do not connect microphone shield wire to chassis ground. This can cause earth loop feedback, RF feedback and AC hum pickup.
Do not short microphone ground to chassis ground. This can cause earth loop feedback and RF feedback.

If RF feedback is still present after installing effective microphone RF filtering, there is a problem somewhere that must be fixed. See Troubleshooting.
 

Grounding

DSP MIKE INPUT to MICROPHONE
The microphone cable must be shielded, and the shield must connect to the DSP microphone input ground. DSP microphone input ground is connected directly to PCB DC ground. It is grounded for audio, RF and DC. Connecting the shield to microphone ground prevents RF pickup, and AC hum pickup.

If the microphone cable has a separate ground wire and shield wire, they must be connected together at the processor. They can also be connected together at the microphone.

Do not connect the microphone shield wire to chassis ground. The DSP chassis ground is not DC grounded or audio grounded, and is not guaranteed to be RF grounded due to aluminium oxide and powder coating on the enclosure. Lack of grounding at audio frequencies will cause AC hum pickup. If the chassis has RF flowing on it, connecting the shield to the chassis could feed the RF directly into the microphone input via capacitive coupling through the cable. This can cause RF feedback.

Do not short microphone ground to chassis ground. Chassis ground is intentionally isolated from DC ground to prevent ground loops. Ground loops at the processor input will cause ground loop feedback (sounds just like RF feedback) due to the high processor gain. When the microphone ground is shorted to the chassis, this could happen at any time if the DSP box were to touch other grounded metalwork.

Do not use the DSP PTT input ground for a microphone ground or shield connection. It is only grounded via a 1k resistor, and will not be an effective ground for DC, audio or RF.

DSP MIKE OUTPUT to TRANSCEIVER
The DSP microphone output feeds the audio into the transceiver microphone input. The DSP microphone output ground connection must only connect to the transceiver microphone socket audio ground pin. The DSP microphone output ground connection must remain isolated from the DSP PCB ground, and the chassis ground. Any connection between these ground points can cause ground loop feedback (sounds like RF feedback).

If the transceiver microphone socket has separate PTT grounds and audio grounds, use the audio ground only. The PTT ground pin may not share the same ground voltage as the transceiver microphone input stage, due to being connected to a different part of the transceiver PCB. This can cause ground loop feedback (sounds like RF feedback).

Do not short the DSP microphone output ground to DSP chassis ground. This can cause ground loop feedback if the DSP chassis becomes shorted to another ground.

Do not short the DSP microphone output ground to transceiver chassis ground. This can cause ground loop feedback if the chassis is DC or audio grounded via any other path.

Do not short the DSP microphone output ground to transceiver power ground. This will cause ground loop feedback by disabling the ground loop cancellation circuit.

Do not short the DSP microphone output ground to DSP PCB ground. This will cause ground loop feedback by disabling the ground loop cancellation circuit.

NOTES
These rules might seem excessive. Sometimes, these rules can be broken and everything seems fine... until something changes. For example, if the mike ground is connected to the chassis ground of the DSP box, this of itself doesn't cause a problem. But if the box touches or is connected to other grounded metalwork (such as a metal bench, a station ground bus or a transceiver casing) suddenly it will cause ground loop feedback. Having correct ground connections prevents this from ever happening.

Because the DSP Speech Processor can have as much as 40dB gain on the microphone input, any signals caused by ground loop current or RF pickup will be greatly amplified. This causes problems that can range from slight distortion effects through to 100% feedback. It is important that the microphone shielding is correctly grounded to keep RF out of the microphone wiring. Even though the microphone and microphone input stage have RF lowpass filtering, there is a limited amount of RF rejection. At low frequencies this rejection is less than at higher frequencies. Strong RF pickup due to incorrect cable shielding can still cause RF feedback.

Another thing to be aware of, is there are often different grounds used for different frequency ranges. For example, standard practice in radio transceivers is to have the antenna output ground connected to the chassis, and the negative power connection used as PCB ground. But in many cases, these two grounds are isolated at DC and connected at RF by only grounding the PCB via small capacitors. For mobile radios, this is often done to prevent short circuits in case the radio is used in a vehicle using positive chassis ground on the electrical system. If the radio is manufactured with the PCB grounded to the chassis at DC, this is usually only done in certain places on the circuit to prevent ground loops due to the very high currents being used (often over 10A for 100W transceivers). The high currents cause voltage drops in the wiring and PCB tracks (due to resistance) that have to be kept from affecting sensitive inputs, (such as microphone inputs) that are operating in the millivolt range. This is why transceiver microphone inputs are never grounded to the chassis at DC and audio frequencies - although they are often grounded at radio frequencies using RF bypass capacitors to help prevent RF from entering via the microphone wiring.
 

Microphone Wiring

Mini-DIN pinouts

INPUT

OUTPUT

EQ and output level modifications

To decrease DSP analog output EQ (to suit transceivers with wider filters, or to reduce bass level or bass distortion in some transceivers):
Change R31 to 10k to reduce EQ by 6dB.
Connect a 22pF capacitor from U1b pin 6 to 7 to improve stability. This prevents OpAmp oscillation.
The output level will increase after this mod. Reduce output level setting to compensate.

Increasing output level:
Scale R45 as needed to increase level.
Some transceivers need more audio drive. Change R45 to 22k for 6dB more output, or 10k for 12dB more output. Do not reduce R45 more than necessary. Overdriving the transceiver can cause distortion and splatter.
Keep R45 above 1k.

Changing the analog EQ corner frequency:
Some transceivers have a wider or flatter frequency response, and need a lower corner frequency for the DSP analog EQ boost. The following table should help to calculate the best value for C31. Supplied by Frank, G0GSR.
 

C31	f+3dB
68n	477Hz
100n	322Hz
330n	99Hz
560n	57Hz

Voltages

U2a pin 1: 6.03V
U2b pin 7: 6.60V Versions prior to v5.1 are 4.51V

U1a pin 1: 5.64V
U1b pin 7: 5.86V

These voltages are not critical, but should be near Vcc/2 (6V) for maximum OpAmp voltage range to avoid clipping.

Current Drain

If current consumption is much higher than this, test for audio amp oscillation, a short circuit, or a failed SMD regulator.
Audio amp oscillation can be caused by: Poor connections, failed or poor quality components or shorted output wiring.

Caution: The 8.3V rail is supplied by a 5V regulator with its adjust pin connected to the 3.3V rail, which may prevent the 5V regulator overcurrent protection from working. Avoid shorting the 8.3V rail!

Temperature

Construction Pictures

The LED Display Board(s) - Contains a low-cost 3.3V SMD serial LED driver IC, a TDA2822 used as a full bridge LED driver for the R/G LED, a TOSLINK digital audio output driver circuit, and assorted dimming logic.

The green bargraph LEDs are superbright water-clear types to match the brightness of the other LEDs. They were made diffused by roughening the front surface with fine sandpaper.

A 3.5 mm TOSLINK adaptor was built from a 3mm red LED and a 3.5 mm socket.

Normal resistors were used to make construction easier. Some surface mount devices were used to save space and money. The header pins are standard right-angle types, modified to be surface mounted. The CAD process makes the boards fit together perfectly.

The two PCBs side-by-side.

The two PCBs fit together well, like a sandwich. The back of Display PCB 1 is all groundplane, except for an isolation strip near the front edge.

Triangle modulated sinewave being used to perform a linearity test on a transceiver. A good result!

DSP Speech Processor transceiver setup

Connect a 4 to 8 ohm speaker to the DSP Speech Processor 3.5 mm speaker output (the socket further away from the DC Power socket). The TDA2003 audio amplifier IC can drive speakers down to 2 ohms, but the bass response will decrease due to the output coupling capacitor being 1000 uF.

Transceiver microphone fly leads will need to be made to suit the transceiver. Consult your transceiver schematic diagram for the microphone wiring. The transceiver microphone lead goes to the mini-DIN socket nearest the DC Power connector, and the input microphone lead goes to the mini-DIN socket furthest from the DC Power connector.

See Microphone Wiring Diagram.
See Grounding.

Receive

Adjust the radio volume control so the last orange LED lights on the loudest voice or noise peaks. This is the point at which the DSP Receive AGC activates. Signals above this level will be compressed to prevent them exceeding the set volume level. The volume can be reduced further to avoid this AGC action if desired.

The DSP Speech Processor Receive volume setting can be adjusted by the "r" Select Menu. This is the default at power-on. At the time of writing, this DSP adjustment will be lost at power-off. At power-on, the unit will start at the default setting again.

If NR is activated via Select Menu "n", the audio level may reduce. If needed, increase the radio volume slightly until the speaker volume is at a comfortable level again. NR decreases random noise on the signal, and enhances cyclic content.

The NR adjustments are:
d Delay: Adjusts the DSP delay between the audio input and the LMS filter. Increasing delay makes the noise reduction more aggressive, and increases audio artifacts.
G Gain: Adjusts how closely the LMS filter attempts to follow the signal. Increasing gain makes the signal louder and increases audio artifacts.
L Leak: Adjusts how fast the LMS filter response decays to zero. Increasing leak improves noise reduction and increases audio artifacts.
= EQ: Adjusts the EQ slope of the LMS audio input. Increasing EQ boosts treble and cuts bass. Decreasing EQ boosts bass and cuts treble.

If the red LED lights at any time, it indicates input ADC clipping. Reduce the radio volume until the red LED does not light on signal peaks.
 

Transmit

Audio can also be monitored via the S/PDIF output. If the S/PDIF is fed to a stereo amplifier, severe alias distortion will be heard, as the S/PDIF output does not currently feature antialias filtering. If S/PDIF audio is recorded using a sample rate of 11025 Hz, the alias frequencies above 5.5 kHz will be completely removed. The S/PDIF audio is the unprocessed (Right) output and processed (Left) output (no transceiver output EQ).

Transmit setup of the processor is best done while transmitting into a dummy load.

Switch PTT to transmit and speak normally into your microphone. The processor LEDs should light up to show each speech band being compressed. The first green LED (LED 0) will mostly light during sibilance ("S" sounds) or other high frequency sounds. The last orange LED (LED 7) will light most of the time, showing output compressor action. The red LED (LED 8) will light to show Hilbert clipper action.

Set the correct bandwidth to match the transceiver. This is set using the Select menu "U" option.
At the time of writing, six settings are available:

• LED 0 - Wide1 60 Hz - 4 kHz
• LED 1 - Wide2 125 Hz - 4 kHz
• LED 2 - Wide3 250 Hz - 4 kHz
• LED 3 - Narrow1 60 Hz - 3 kHz
• LED 4 - Narrow2 125 Hz - 3 kHz
• LED 5 - Narrow3 250 Hz - 3 kHz

Adjust the transceiver microphone gain for the desired ALC action. If the output EQ is well matched to the transceiver, it should be possible to obtain full PEP output at all voice frequencies with little or no ALC action.

If this is not possible, it likely indicates the DSP output EQ does not suit the transceiver. With the current design, this will usually mean the bass end of the audio spectrum is too low in level. Sometimes this can be improved by boosting the bass EQ by circuit adjustment, or DSP output EQ adjustment (limited selections are available at this time). Another method is to modify the transceiver audio response to be flatter towards the bass end of the spectrum, by increasing coupling capacitor values in the transceiver microphone amplifier circuit.

Transceiver SSB carrier offset frequencies should be checked for accuracy and corrected if errors are found. It is possible to improve the low frequency transmit response on SSB by increasing the audio path coupling capacitor values, and by careful tuning of the carrier offsets. This may degrade opposite sideband rejection slightly at low frequencies. It is not too difficult to obtain a reasonably flat SSB response down to 100Hz on a typical transceiver, using the SSB 2 EQ (default) and some careful carrier offset tuning (see image below). This is done using the white noise mode (main mode "d") while observing the signal on a low-cost SDR receiver. An RTL dongle works well on the higher bands (27/28MHz) for this purpose. The carrier is placed part way down the SSB filter curve, and finely adjusted for best passband flatness on white noise. Only a small vestige of opposite sideband energy is left close to the carrier (depends on filter response). If audio distortion is too high, increasing the transmit path RF gain after the balanced modulator can decrease distortion by lowering the audio drive requirement. This will reduce the carrier suppression by an equal amount, but with careful adjustments it can easily reach an acceptable limit. The best option for wideband audio on SSB might be to replace the SSB filter with a new unit having a wider response and/or steeper skirts, or even better still - use a DSP based SSB generator, or a DSP based transceiver.

The transceiver passband response can be checked by using the Mode "b" Swept tone flatness test, and an RF power meter or scope. This will display the combined response of the DSP output EQ, analog EQ, transceiver microphone amplifier and SSB filter. Ideally, the output power should be constant across the desired transmit passband. The white noise mode "d" can be used to check transmit passband flatness and opposite sideband rejection, by using an SDR receiver with a spectrum display. See image below.

If the DSP audio output level needs adjustment, it can be performed as follows:
With the PTT down, and the processing Mode number displayed, go to the Output Level setting in the Select Menu by pressing the Select button (button 2) three times. The 7-segment LED should display "o". The output level can then be increased by button 4 or decreased by button 3. This setting can now be saved using the "Programming Mode".

If the transceiver features an SSB RF clipper processor, the RF clipper can be used as a final limiting stage. Since an RF clipper has no methods to control in-passband distortion, it should be used at its lowest level for best quality (just barely clipping). The DSP then does the bulk of the processing, with the RF clipper providing final peak protection for the transmit stages by removing overshoots. Any other transceiver audio compression should be disabled, or reduced to minimum. If the transceiver does not feature an RF clipper, but instead only has an audio based processor/clipper, it is recommended to leave the transceiver processing disabled.

SSB white noise spectrum on a typical CB transceiver using a 4kHz wide monolithic crystal filter. The LSB passband is mostly flat with only a tiny amount of USB energy present.

Troubleshooting

Possible problems:

Audio break-up, echoing distortion, scratchy noise or feedback (SSB) or feedback squeal (AM).
Could be RF feedback, or a ground loop.

• Perform a test transmission on a dummy load, running the same power level.
• If feedback persists, problem is a ground loop.
• Ensure the DSP PCB "Mike GND" output is not shorted to main PCB ground, or chassis ground. Should only be grounded to PCB ground (negative power) through R44 (10k).
• Ensure DSP PCB Rx audio input ground is not shorted to PCB ground. Should only be grounded to PCB ground (negative power) through R22 (10k).
• Ensure the output "Mike GND" line is connected to the transceiver Audio Ground line on the microphone connector.
• Ensure that L3 and L4 have not been swapped by accident. L4 needs to have a very low resistance for the ground loop cancellation to work. L3 needs higher inductance (1mH = 1000uH).
• Ensure TDA2003 tab is not shorting to the top of the enclosure.
• Do not connect microphone ground to chassis ground. This can cause ground loop feedback.
• Do not connect microphone shield wire to chassis ground. This can cause RF pickup and ground loop feedback.
• If using microphone scan buttons, check for any ground interconnections that might be causing an earth loop. Isolate PTT input ground and microphone input ground from scan buttons.
• If feedback stops when using a dummy load, it is due to RF.
• Ensure microphone wire shielding is grounded to microphone ground. Microphone cable shielding must connect to microphone ground, not chassis ground. See Microphone Wiring.
• Ensure RF bypass capacitors are ceramic, and the correct value (2.2nF, 2200pF).
• In multi-wire cables, ensure microphone signal wire is the same wire that has the shielding around it (important!). Data cable is not recommended for microphone cable. Multi-shielded wire (1 shield on each wire) is OK.
• Ensure the microphone cable shield is connected to the DSP Mike input ground, and not PTT ground. PTT ground is only grounded through a 1k resistor, so it is unsuitable for shield grounding.
• Do not short microphone ground to chassis ground. This can cause ground loop feedback.
• Do not connect microphone shield wire to chassis ground. This can cause AC hum pickup, RF pickup and ground loop feedback.
• Electret microphones need a ceramic RF bypass capacitor added directly across their terminals. Use a 2n2 capacitor.
• Do not overheat the microphone element during soldering, or it can be damaged. Tin the capacitor leads first. Use a clamp-on soldering heatsink or pliers. Solder quickly with a temperature controlled iron.
• If RF feedback persists after the capacitor is added, add a 1mH RF choke in series with the microphone signal line at the back of the electret element. 1mH = 1 millihenry = 1000uH = Brown Black Red.
• A larger value RFC can be fitted if available. This can increase RF suppression at low frequencies.
• See Grounding.

• If using a separate power supply for the DSP unit, the negative power wire must be joined to the transceiver power supply negative.
• Check microphone shielding and grounding. Microphone shielding must connect to microphone ground, not chassis ground. Do not short chassis ground to microphone ground - this can cause ground loop feedback.
• Dynamic microphones are sensitive to AC magnetic fields. If using a dynamic microphone, ensure it is not near a mains transformer, filter choke or other source of magnetic hum field.
• Isolate the enclosure from external grounds. If this cures the problem, the enclosure or PCB edge tracks may be shorted to the DSP PCB ground.

• Output EQ doesn't match transceiver EQ adequately, or transceiver bandwidth is too narrow for processor bandwidth.
• Ensure Utility mode is set to match transceiver bandwidth.
• Use Mode "b" Swept Tone Flatness Test to check transmit passband flatness. Modify DSP output EQ setting, hardware EQ components or transceiver EQ to improve flatness.
• Some limited DSP output EQ settings are available in the firmware.
• Alternatively, reduce output level, reduce transceiver microphone gain control, reduce ALC setting, or enable the transceiver RF clipper if it has one.
• Transceiver built-in speech processors can generate high distortion levels.

• Microphone AGC could be frozen. Press button "MODE" (first pushbutton) to reset input AGC level.
• Audio is too low to reach limiting. Increase microphone level. Increase input gain setting in Select menu.
• Hilbert Clipping is disabled. Enable the Hilbert Clipper. The last Red LED should light on voice.

• Transmit output level may be set too high. Reduce it.
• Some old transceivers may have poor ALC. Reduce audio drive to transceiver, reduce ALC setting, or modify ALC circuits for better response.

• The peak-to-average ratio of the processed output in Mode 4 and Mode 5 is about 6dB. Don't attempt much more than 25 watts average power on a 100 watt PEP radio, or distortion and splatter can result. Most RF power meters only read average power, not PEP.
• Lower the output level or transceiver mike gain control.
• Disable the transceiver built-in processing as much as possible.
• Transceiver built-in speech processors can generate high distortion levels.

• Is the Red Tx LED constantly flashing yellow? This indicates input AGC action due to excessive microphone drive. Reduce microphone input level. Normally, this won't cause distortion due to good AGC action.
• It's OK if the input AGC activates on occasional peaks. Don't reduce the mike input too much, or the processor will lose effectiveness. Most LEDs should light up on speech.
• If the microphone input level is reduced too much, the noise gates can shut down, causing muffled audio.

• For accurate control of peak levels, it is important to have the flattest possible frequency response between the DSP clipper stage and the transmitter RF output (voice EQ is applied BEFORE the clipper stage).
• Getting good bass response on a typical SSB transceiver is difficult, as they are designed to suppress low voice frequencies.
• To compensate for this, a large amount of bass boost output EQ is used. Some radios cannot handle high bass levels and can distort badly.
• If this happens, there are three options:
• Reduce output level, or reduce/disable output EQ via the Select menu. * * * Note: Disabling the DSP EQ raises the output level!
• Select a narrower DSP bandwidth using the Utility modes.
• Modify transceiver transmit response to be flatter at low frequencies.
• Test for audio peak clipping in DSP OpAmp output stages and transceiver microphone amplifier stages using a scope.
• Normally, the DSP output can't clip even at maximum output setting and EQ. If it does happen, check for a component error around the OpAmp output stages. Also test the DC voltages at the output of the OpAmps.
• Transceiver mod: Increase audio coupling capacitor values in microphone preamp and balanced modulator circuits.
• Check transceiver SSB carrier offsets are correct. SSB carrier offsets can be tuned to improve bass response, but it may reduce opposite sideband and carrier rejection. See Setup.
• Test the transmit frequency response using the DSP white noise output (Mode d) and an SDR receiver, or the DSP swept tone (Mode b) and an RF power meter. Adjust settings to make it flatter. See Setup.
• If required, the analog low frequency EQ boost can be changed. See Modifications.

• Reduce the Hilbert clipping level.
• If the RED Hilbert clipping LED (last LED) is not lighting on voice, but the distortion is still occurring, the distortion is not from processing. Reduce the output level.
• If the RED Hilbert clipping LED (last LED) is not lighting on voice, the Hilbert clipper is not clipping. This will make the processor less effective.
• Using voice input, the audio on the default setup after power-on sounds processed, but GOOD QUALITY. The Red Hilbert clipping LED lights brightly. Distortion is only heard when the received signal is strong.
• If single tone mode (8) causes distortion, the radio is being overdriven. Reduce output level until RF power begins to drop slightly.

• Press button "MODE" (first pushbutton) to reset input AGC level.
• Installing adequate "pop" filtering on the microphone prevents this problem being caused by breath noise. More than one layer may be necessary. Don't eat the mike!
• If this happens repeatedly from only moderate input levels, it means the mike input capacitors may have high leakage current. Use 1uF tantalum (or monolithic ceramic) capacitors for C1 and C3. C1 is most critical as it has the highest voltage across it (maximum 8.3 volts - 10V capacitors are adequate).
• At high temperatures, the input AGC transistor gain increases. This may cause AGC malfunction on some units. The January 13, 2018 firmware reduces the AGC loop gain to improve stability.
• Analog input AGC Gain Freeze can now be disabled. This will remove the requirement to manually reset the input AGC by hitting button 1, but it won't fix audio problems caused by strong LF noise affecting the AGC.

• Ensure transceiver internal speech compression is disabled.
• Ensure output level is set correctly. Transceiver ALC action should be minimal.
• Enable the noise gate. The noise gate can reduce background noise by about 20 dB. Enabled by default.
• A compression gain setting has been added to the firmware. Reducing the gain will reduce background noise.
• Attenuate the microphone signal. Use the resistive shunt attenuator pot shown in the schematic, wired across the mike.
• Use a unidirectional or cardioid microphone. This greatly reduces noise to the sides and rear of the microphone.
• Disable the Output Compressor. This will reduce total compression, and reduce background noise. Hilbert clipping level may need to be adjusted. Distortion may increase.

• Caution! If U4 oscillates it can become a burn hazard as it will be VERY HOT (~90° C).
• An oscilliscope will show a large HF signal on the U4 speaker output terminal if oscillation is occurring.
• Ensure bypass capacitors C13 and C14 are properly connected, and of good quality.
• Start with a value of 220nF for C15.
• If oscillation persists, try increasing C15 further, or reducing R20 in value.
• Try increasing bypass capacitor C14 in value.
• Prototype 2 was unstable with 100n for C15, but became stable with 200n.

PCB Pictures

Commercially made PCB, underside:

Commercially made PCB, Display PCB 1:

Commercially made PCB, Display PCB 2 - front view (not to scale):
3mm or SMD (untested) LEDs can be used with this panel. Safety resistor "R" is optional, and has a shorting link. It can be ignored.

Overlay of main PCB - all resistors marked. Click for full-size image.

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