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Add CESSB

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boblark 2 years ago
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      docs/index.html
  2. BIN
      gui/DesignTool_F32.zip

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docs/index.html
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@ -417,6 +417,10 @@ span.mainfunction {color: #993300; font-weight: bolder}
{"type":"radioModulatedGenerator_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"Modulator","inputs":"2","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
{"type":"radioNoiseBlanker_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"NoiseBlank","inputs":"2","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
{"type":"radioCESSBtransmit_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"CESSB_Mod","inputs":"1","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
{"type":"RadioFMDiscriminator_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FMDiscrim","inputs":"1","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
{"type":"radioBFSKModulator_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"BFSKMod","inputs":"0","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
{"type":"UART_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"UART","inputs":"1","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
@ -3449,6 +3453,125 @@ The actual packets are taken
</div>
</script>
<script type="text/x-red" data-help-name="radioCESSBtransmit_F32">
<!-- ============ radioCESSBtransmit_F32 ========= -->
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Converts audio into Weaver SSB and then applies Dave Hershberger, W9GR,
CESSB controlled clipping and
filtering. This prevents the SSB peaks from exceeding a maximum level.
This increases the peak limited average power by 3 or more dB. The output
can be converted to conventional SSB by a RadioIQMixer_F32 object.
</p>
</div>
<h3>Audio Connections</h3>
<table class=doc align=center cellpadding=3>
<tr class=top><th>Port</th><th>Purpose</th></tr>
<tr class=odd><td align=center>In 0</td><td>Input Audio Signal</td></tr>
<tr class=odd><td align=center>Out 0</td><td>Weaver SSB I Signal</td></tr>
<tr class=odd><td align=center>Out 1</td><td>Weaver SSB Q Signal</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>setSampleRate_Hz</span>
(<strong>float32_t</strong> fs_Hz bitRate);</p>
<p class=desc>Specifically, this sets the sample rate, in samples per second,
that is used by CESSB. It also sets other parameters, such as
decimation ratios and filter cutoff frequencies. Thus this function
is <strong>required.</strong> At this time, the design is centered
on 48000 sps, but can be used with other close values such as
44100 or 50000. The plan is to eventually support 96000 sps if users
are needing it. There is no default value and the CESSB objects will not run if
this function is not called.</p>
<p class=func><span class=keyword>getLevels</span>(<strong>int</strong> what);</p>
<p class=desc>Returns a pointer to a structure of type levels. This allows
knowledge of the average and peak levels at both the input and output sides
of the SSB clipper and overshoot compensator. If what==0 the pointer is returned
but no updating is done. That is used to setup the process before data is
available. If what != 0, the contents of the structure are updated and measuring
is reset. The function levelDataCount() below can be used to set the time
between updates. The stucture is part of the object and is defined as:
<pre>
struct levels {
float32_t pwr0; // Average power at input
float32_t peak0; // Peak voltage at input
float32_t pwr1; // Average power at output
float32_t peak1; // Peak voltage at output
uint32_t countP; // Number of averaged samples for pwr0.
};
</pre></p>
<p class=func><span class=keyword>levelDataCount</span>();</p>
<p class=desc>Returns an uint32_t with the number of averaged samples
of the input power. See getLevels() above. The number of output
samples may differ by an integer factor because of decimation inside
the object.</p>
<p class=func><span class=keyword>setGains</span>(
<strong>float32_t</strong> gainIn,
<strong>float32_t</strong> gainCompensate,
<strong>float32_t</strong> gainOut);</p>
<p class=desc> These are the controls for the CESSB class. gainIn sets
the amount of clipping by setting the input level to the clipper.
gainCompensate sets the amount of correction applied to prevent
overshoot. A value of 2.0 or slightly less is normally used. gainOut is
for convenience and sets the drive level to the next block. </p>
<p class=func><span class=keyword>setSideband</span>(<strong>bool</strong> sbReverse);</p>
<p class=desc>The LSB/USB selection depends on the processing of the
IQ outputs of this class. But, what we can do here is to reverse the
selection by reversing the phase of one of the Weaver LO's. </p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; OpenAudio_ArduinoLibrary &gt; CESSB
</p>
<h3>Notes</h3>
<p>The technical description, implementation and test results are in references
listed in the include file for this class,
<a href="https://github.com/chipaudette/OpenAudio_ArduinoLibrary/blob/master/radioCESSBtransmit_F32.h"
target="_blank">available from Github</a>
These should be used to understand the details of CESSB. The notes at the top of
that include file has information relating to this Teensy Audio implementation, as well.
</p>
<p>The first activity for CESSB is to limit or clip the amplitude of the SSB signal. Internally
this always occurs when the envelope of the SSB signal exceeds 1.0. This is all done
with floating point arithmetic so values may exceed 1.0. The input level where this occurs
depends on the setting for gainIn, described above. The maximum level seen ahead of the clipper
is measured by getLevels() as described above. One way to control the input to the CESSB block
is with
<a href="http://www.janbob.com/electron/OpenAudio_Design_Tool/index.html?info=AudioEffectCompressor2_F32"
target="_blank">Compressor2 Library block.</a> Note that Compressor2
is not a clipper, but is rather an automatic gain control that uses look-ahead
processing to allow gradual gain changes.
</p>
<p>The output of the CESSB processing is two sampled data signals representing
Weaver SSB. This uses the in-phase and quadratuere components of the SSB signal
that has been converted to frequencies of -1350 to +1350 Hz. Wow, if that seems
confusing, take a look at the CESSB example and see how this can be translated
into either an Upper Sideband (USB) or a Lower Sideband (LSB) signal. Implemented
in DSP, the Weaver Method of SSB generation has some interesting and good features.
It is worth considering as an alternative to the phasing method, with or without CESSB.
By lowering the input level, this CESSB block can be used as a Weaver Method SSB
generator.</p>
<p>CESSB as implemented here is intended for voice input, and also filters the voice
to a communications bandwidth of around 2700 Hz.</p>
</script>
<script type="text/x-red" data-template-name="radioBFSKModulator_F32">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>
<script type="text/x-red" data-help-name="radioBFSKModulator_F32">
<h3>Summary</h3>
<div class=tooltipinfo>
@ -3811,11 +3934,12 @@ Times: T3.6 update() block of 128 is about 53 microseconds, AM Single output.
<!-- ============ RadioIQMixer_F32 ========= -->
<h3>Summary</h3>
<div class=tooltipinfo>
<p>This quadrature mixer block for both transmit and receive.
A basic building block is a pair of mixers with the
<p>This quadrature mixer block is for both transmit and receive.
It is a basic building block with a pair of mixers along with a sin/cose
LO going to the mixers at the same frequency, but differing in phase
by 90 degrees (programmable). The LO's are included
in the block, but there are no post-mixing filters. </p>
in the block, but there are no post-mixing filters. Hardware
phase and amplitude error correction is included. </p>
</div>
<h3>Boards Supported</h3>
<ul>
@ -3836,32 +3960,41 @@ Times: T3.6 update() block of 128 is about 53 microseconds, AM Single output.
<h3>Functions</h3>
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> fr);</p>
<p class=desc></p>
<p>Sets Mixer LO frequency in Hz</p>
<p class=desc>Sets Mixer LO frequency in Hz. The default is 1000 Hz.</p>
<p class=func><span class=keyword>iqmPhaseS</span>(<strong>float</strong> ps);</p>
<p class=desc>This phase comes in the range (0, 2PI) keeping with C math functions.
<p class=desc>This phase comes in the range (0, 2PI radians) keeping with C math functions.
This function allows multiple mixers to be phase coordinated (stop
interrupts when setting). </p>
interrupts when setting).</p>
<p class=func><span class=keyword>phaseS_C_r</span>(<strong>float</strong> pc);</p>
<p class=desc> Sets the number of radians that the cosine LO leads the
sine LO. The default is PI/2 radians. This is used to correct hardware phase unbalance.</p>
sine LO. The default is PI/2 radians. This is used to correct hardware phase unbalance.
Not changeable if doSimple==true. </p>
<p class=func><span class=keyword>amplitudeC</span>(<strong>float</strong> g);</p>
<p class=desc> Sets the gain, g, for the I channel.
The Q channel is always 1.0. This is used to correct hardware amplitude unbalance.</p>
The Q channel is always 1.0. This is used to correct hardware amplitude unbalance.
Not changeable if doSimple==true. The default is g=1.0.</p>
<p class=func><span class=keyword>void useTwoChannel</span>(<strong>bool</strong> twoCh);</p>
<p class=func><span class=keyword>void useTwoChannel</span>(<strong>float32_t</strong> gainOut);</p>
<p class=desc>
Channel 0 (left) is the in-phase input I for twoCh true of false. Channel 1 (right) is Q for
complex 2-channel input (twoCh==true) and not used for twoChannel==false. Caution, never
use twoCh=false with two inputs.
use twoCh=false with two inputs. The default is twoCh==false.
</p>
<p class=func><span class=keyword>setGainOut</span>(<strong>float</strong> g);</p>
<p class=desc> Sets the gain, g, for both the I and Q channels.
The default value is 1.0. It is available for either doSimple or not doSimple. The reason for this
function is that, is that the I and Q mixers have a gain of 0.5 for either sideband output. This function
can bring the net gain of the object to unity by setting gainOut to 2.0f. It can
also be used as a general gain control.
</p>
<p class=func><span class=keyword>useSimple</span>(<strong>bool</strong> simple);</p>
<p class=desc>Faster if true, but no phase/amplitude adjustment.</p>
<p class=desc>Faster if true, but no phase/amplitude adjustment. Default is doSimple = true.</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; OpenAudio_ArduinoLibrary &gt; ReceiverPart1
@ -3873,13 +4006,16 @@ Times: T3.6 update() block of 128 is about 53 microseconds, AM Single output.
<p class=exam>File &gt; Examples &gt; OpenAudio_ArduinoLibrary &gt; ReceiverSSB
</p>
<p class=exam>File &gt; Examples &gt; OpenAudio_ArduinoLibrary &gt; CESSB
</p>
<h3>Notes</h3>
<p>There is provision for varying
the phase between the sine and cosine oscillators. The relative gain in the
I and Q channels is also programmable. This allows for flaws in the
I and Q channels is also programmable. This can be used to correct for errors in the
response of real hardware.</p>
<P>The output levels are 0.5 times the input level </P>
<P>The output levels are 0.5 times the input level, for each sideband. </P>
<p>Time: T3.6, For an update of a 128 sample block, doSimple=1, 46 microseconds;
T4.0, For an update of a 128 sample block, doSimple=1, 20 microseconds</p>

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