pull/11/head
boblark
4 years ago
parent
ae99656d29
commit
0af24d2bef
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/*
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* AudioAnalyzePhase_F32.cpp |
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* |
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* 31 March 2020 Rev 8 April 2020 |
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* Bob Larkin, in support of the library: |
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* Chip Audette, OpenAudio, Apr 2017 |
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* |
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* * Copyright (c) 2020 Bob Larkin |
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* |
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* Permission is hereby granted, free of charge, to any person obtaining a copy |
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* of this software and associated documentation files (the "Software"), to deal |
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* in the Software without restriction, including without limitation the rights |
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
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* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
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* |
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* The above copyright notice and this permission notice shall be included in all |
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* copies or substantial portions of the Software. |
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* |
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
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* SOFTWARE. |
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*/ |
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/* There are two inputs, I and Q (Left and Right or 0 and 1)
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* There is one output, the phase angle between the two inputs expressed in |
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* radians (180 degrees is Pi radians). See AudioAnalyzePhase_F32.h |
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* for details. |
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*/ |
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#include "AudioAnalyzePhase_F32.h" |
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#include <math.h> |
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#include "mathDSP_F32.h" |
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void AudioAnalyzePhase_F32::update(void) { |
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audio_block_f32_t *block0, *block1; |
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uint16_t i; |
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float32_t mult0, mult1, min0, max0, minmax0, min1, max1, minmax1; |
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float32_t min_max = 1.0f; // Filled in correctly, later
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mathDSP_F32 mathDSP1; |
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#if TEST_TIME |
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if (iitt++ >1000000) iitt = -10; |
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uint32_t t1, t2;
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t1 = tElapse; |
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#endif |
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// Get first input, 0
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block0 = AudioStream_F32::receiveWritable_f32(0); |
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if (!block0) { |
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if(errorPrint) Serial.println("AN_PHASE ERR: No input memory"); |
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return; |
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} |
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// Get second input, 1
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block1 = AudioStream_F32::receiveWritable_f32(1); |
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if (!block1){ |
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if(errorPrint) Serial.println("AN_PHASE ERR: No input memory"); |
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AudioStream_F32::release(block0); |
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return; |
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} |
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// Limiter on both inputs; can be used with narrow IIR filter
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if (pdConfig & LIMITER_MASK) { |
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for (uint16_t j = 0; j<block0->length; j++) { |
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if (block0->data[j]>=0.0f) |
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block0->data[j] = 1.0f; |
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else |
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block0->data[j] = -1.0f; |
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if (block1->data[j]>=0.0f) |
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block1->data[j] = 1.0f; |
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else |
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block1->data[j] = -1.0f; |
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} |
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} |
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// If needed, find a scaling factor for the multiplier type phase det
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// This would not normally be used with a limiter---they are
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// both trying to solve the same problem.
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if(pdConfig & SCALE_MASK) { |
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min0 = 1.0E6f; max0 = -1.0E6f; |
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min1 = 1.0E6f; max1 = -1.0E6f; |
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for (i=0; i < block0->length; i++){ |
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if(block0->data[i] < min0) min0 = block0->data[i]; |
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if(block0->data[i] > max0) max0 = block0->data[i]; |
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if(block1->data[i] < min1) min1 = block1->data[i]; |
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if(block1->data[i] > max1) max1 = block1->data[i]; |
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} |
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minmax0 = max0 - min0; // 0 to 2
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minmax1 = max1 - min1; // 0 to 2
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min_max = minmax0 * minmax1; // 0 to 4
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} |
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// multiply the two inputs to get phase plus second harmonic. Low pass
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// filter and then apply ArcCos of the result to return the phase difference
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// in radians. Multiply and leave result in 1
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arm_mult_f32(block0->data, block1->data, block1->data, block0->length); |
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if (LPType == NO_LP_FILTER) { |
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for (i=0; i < block0->length; i++) |
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block0->data[i] = block1->data[i]; // Move back to block0
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} |
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else if(LPType == IIR_LP_FILTER) { |
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// Now filter 1, leaving result in 0. This is 4 BiQuads
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arm_biquad_cascade_df1_f32(&iir_inst, block1->data, block0->data, block0->length); |
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} |
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else { // Alternate FIR filter for FIR_LP_FILTER
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arm_fir_f32(&fir_inst, block1->data, block0->data, block0->length); |
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} |
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AudioStream_F32::release(block1); // Not needed further
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/* For variable, pdConfig:
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* LIMITER_MASK: 0=No Limiter 1=Use limiter |
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* ACOS_MASK: 00=Use no acos linearizer 01=undefined |
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* 10=Fast, math-continuous acos() (default) 11=Accurate acos() |
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* SCALE_MASK: 0=No scale of multiplier 1=scale to min-max (default) |
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* UNITS_MASK: 0=Output in degrees 1=Output in radians (default) |
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*/ |
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if(pdConfig & SCALE_MASK) |
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mult0 = 8.0f / min_max; |
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else |
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mult0 = 2.0f; |
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if(pdConfig & UNITS_MASK) |
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mult1 = 1.0; |
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else |
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mult1 = 57.295779f; |
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// Optionally, apply ArcCos() to linearize block0 output in radians * units to scale to degrees
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if (pdConfig & ACOS_MASK) { |
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if((pdConfig & ACOS_MASK) == 0b00110) { // Accurate acosf from C-library
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for (uint i = 0; i<block_size; i++) { |
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block0->data[i] = mult1 * acosf(mult0 * block0->data[i]);
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} |
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} |
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else if ((pdConfig & ACOS_MASK) == 0b00100) { // Fast acos from polynomial
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for (uint i = 0; i<block_size; i++) { |
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block0->data[i] = mult1 * mathDSP1.acos_f32(mult0 * block0->data[i]); |
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} |
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} |
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} |
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// Not applying linearization or scaling, at all. Works for multiplier outputs near 0 (90 deg difference)
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else { |
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float32_t mult = 8.0f * mult0 * mult1; |
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for (uint i = 0; i<block_size; i++) { |
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block0->data[i] = mult * block0->data[i]; |
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} |
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}
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AudioStream_F32::transmit(block0, 0); |
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AudioStream_F32::release(block0); |
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#if TEST_TIME |
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t2 = tElapse; |
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if(iitt++ < 0) {Serial.print("At AnalyzePhase end, microseconds = "); Serial.println (t2 - t1); }
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t1 = tElapse; |
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#endif |
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} // End update()
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@ -0,0 +1,271 @@ |
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/*
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* AudioAnalyzePhase_F32.h |
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* |
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* 31 March 2020, Rev 8 April 2020
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* Status Tested OK T3.6 and T4.0.
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* Bob Larkin, in support of the library: |
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* Chip Audette, OpenAudio, Apr 2017 |
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* ------------------- |
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* There are two inputs, 0 and 1 (Left and Right) |
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* There is one output, the phase angle between 0 & 1 expressed in |
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* radians (180 degrees is Pi radians) or degrees. This is a 180-degree |
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* type of phase detector. See RadioIQMixer_F32 for a 360 degree type. |
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* |
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* This block can be used to measure phase between two sinusoids, and the default IIR filter is suitable for this with a cut-off |
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* frequency of 100 Hz. The only IIR configuration is 4-cascaded satages of BiQuad. For this, 20 coefficients must be provided |
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* in 4 times (b0, b1, b2, -a1, -a2) order (example below). This IIR filter inherently does not have very good |
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* linearity in phase vs. frequency. This can be a problem for communications systems. |
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* As an alternative, a linear phase (as long as coefficients are symmetrical) |
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* FIR filter can be set up with the begin method. The built in FIR LP filter has a cutoff frequency of 4 kHz when used |
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* at a 44.1 kHz sample rate. This filter uses 53 coefficients (called taps). Any FIR filter with 4 to 200 coefficients can be used |
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* as set up by the begin method. |
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* |
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* DEFAULTS: 100 Hz IIR LP, output is in radians, and this does *NOT* need a call to begin(). This can be changed, including |
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* using a FIR LP where linear phase is needed, or NO_LP_FILTER that leaves harmonics of the input frequency. Method begin() |
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* changes the options. For instance, to use a 60 coefficient FIR the setup() in the .INO might do |
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* myAnalyzePhase.begin(FIR_LP_FILTER, &myFIRCoefficients[0], 60, DEGREES_PHASE); |
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* If _pcoefficients is NULL, the coefficients will be left default. For instance, to use the default 100 Hz IIR filter, with degree output |
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* myAnalyzePhase.begin(IIR_LP_FILTER, NULL, 20, DEGREES_PHASE); |
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* To provide a new set of IIR coefficients (note strange coefficient order and negation for a() that CMSIS needs) |
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* myAnalyzePhase.begin(IIR_LP_FILTER, &myIIRCoefficients[0], 20, RADIANS_PHASE); |
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* In begin() the pdConfig can be set (see #defines below). The default is to use no limiter, but to measure the input levels over the
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* block and use that to scale the multiplier output. This will cause successive blocks to change slightly in output level due to |
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* errors in level measurement, but is other wise fine. If the limiter is used, the narrow band IIR filter should also be used to |
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* prevent artifacts from "beats" between the sample rate and the input frequency. |
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*
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* Three different scaling routines are available following the LP filter. These deal with the issue that the multiplier type |
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* of phase detector produces an output proportional to the cosine of the phase angle between the two input sine waves. |
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* If the inputs both have a magnitude ranging from -1.0 to 1.0, the output will be cos(phase difference). Other values of |
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* sine wave will multiply this by the product of the two maximum levels. The selection of "fast" or "accurate" acos() will |
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* make the output approximately the angle, as scaled by UNITS_MASK. The ACOS_MASK bits in pdConfig, set by begin(), selects the |
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* acos used. Note that if acos function is used, the output range is 0 to pi radians, i.e., 0 to 180 degrees. "Units" have no |
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* effect when acos90 is not being used, as that would make little sense for the (-1,1) output. |
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*
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* Functions: |
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* setAnalyzePhaseConfig(const uint16_t LPType, float32_t *pCoeffs, uint16_t nCoeffs) |
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* setAnalyzePhaseConfig(const uint16_t LPType, float32_t *pCoeffs, uint16_t nCoeffs, uint16_t pdConfig) |
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* are used to chhange the output filter from the IIR default, where: |
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* LPType is NO_LP_FILTER, IIR_LP_FILTER, FIR_LP_FILTER to select the output filter |
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* pCoeffs is a pointer to filter coefficients, either IIR or FIR
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* nCoeffs is the number of filter coefficients |
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* pdConfig is bitwise selection (default 0b1100) of |
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* Bit 0: 0=No Limiter (default) 1=Use limiter |
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* Bit 2 and 1: 00=Use no acos linearizer 01=undefined |
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* 10=Fast, math-continuous acos() (default) 11=Accurate acosf() |
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* Bit 3: 0=No scale of multiplier 1=scale to min-max (default) |
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* Bit 4: 0=Output in degrees 1=Output in radians (default) |
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* showError(uint16_t e) sets whether error printing comes from update (e=1) or not (e=0). |
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*
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* Examples: AudioTestAnalyzePhase.ino and AudioTestSinCos.ino
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*
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* Some measured time data for a 128 size block, Teensy 3.6, parts of update(): |
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* Default settings, total time 123 microseconds |
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* Overhead of update(), loading arrays, handling blocks, less than 2 microseconds |
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* Min-max calculation, 23 microseconds |
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* Multiplier DBMixer 8 microseconds |
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* IIR LPF (default filter) 57 microseconds |
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* 53-term FIR filter 149 microseconds
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* Fast acos_32() linearizer 32 microseconds |
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* Accurate acosf(x) seems to vary (with x?), 150 to 350 microsecond range |
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*
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* Measured total update() time for the min-max scaling, fast acos(), and 53-term FIR filtering |
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* case is 214 microseconds for Teensy 3.6 and 45 microseconds for Teensy 4.0. |
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* |
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* Copyright (c) 2020 Bob Larkin |
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* |
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* Permission is hereby granted, free of charge, to any person obtaining a copy |
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* of this software and associated documentation files (the "Software"), to deal |
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* in the Software without restriction, including without limitation the rights |
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
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* copies of the Software, and to permit persons to whom the Software is |
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* furnished to do so, subject to the following conditions: |
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* |
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* The above copyright notice and this permission notice shall be included in all |
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* copies or substantial portions of the Software. |
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* |
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
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* SOFTWARE. |
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*/ |
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#ifndef _analyze_phase_f32_h |
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#define _analyze_phase_f32_h |
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#define N_STAGES 4 |
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#define NFIR_MAX 200 |
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#define NO_LP_FILTER 0 |
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#define IIR_LP_FILTER 1 |
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#define FIR_LP_FILTER 2 |
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#define RADIANS_PHASE 1.0 |
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#define DEGREES_PHASE 57.295779 |
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// Test the number of microseconds to execute update()
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#define TEST_TIME 1 |
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#define LIMITER_MASK 0b00001 |
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#define ACOS_MASK 0b00110 |
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#define SCALE_MASK 0b01000 |
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#define UNITS_MASK 0b10000 |
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#include "AudioStream_F32.h" |
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#include <math.h> |
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class AudioAnalyzePhase_F32 : public AudioStream_F32 { |
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//GUI: inputs:2, outputs:1 //this line used for automatic generation of GUI node
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//GUI: shortName: AnalyzePhase
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public: |
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// Option of AudioSettings_F32 change to block size or sample rate:
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AudioAnalyzePhase_F32(void) : AudioStream_F32(2, inputQueueArray_f32) { // default block_size and sampleRate_Hz
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// Initialize BiQuad IIR instance (ARM DSP Math Library)
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arm_biquad_cascade_df1_init_f32(&iir_inst, N_STAGES, &iir_coeffs[0], &IIRStateF32[0]); |
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} |
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// Constructor including new block_size and/or sampleRate_Hz
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AudioAnalyzePhase_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) { |
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block_size = settings.audio_block_samples; |
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sampleRate_Hz = settings.sample_rate_Hz;
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// Initialize BiQuad IIR instance (ARM DSP Math Library)
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arm_biquad_cascade_df1_init_f32(&iir_inst, N_STAGES, &iir_coeffs[0], &IIRStateF32[0]); |
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} |
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// Set AnalyzePhaseConfig while leaving pdConfig as is
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void setAnalyzePhaseConfig(const uint16_t _LPType, float32_t *_pCoeffs, uint16_t _nCoeffs) { |
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setAnalyzePhaseConfig( _LPType, _pCoeffs, _nCoeffs, pdConfig); |
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} |
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// Set AnalyzePhaseConfig in full generality
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void setAnalyzePhaseConfig(const uint16_t _LPType, float32_t *_pCoeffs, uint16_t _nCoeffs, uint16_t _pdConfig) { |
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AudioNoInterrupts(); // No interrupts while changing parameters
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LPType = _LPType; |
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if (LPType == NO_LP_FILTER) { |
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//Serial.println("Advice: in AnalyzePhase, for NO_LP_FILTER the output contains 2nd harmonics");
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//Serial.println(" that need external filtering.");
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} |
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else if (LPType == IIR_LP_FILTER) { |
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if(_pCoeffs != NULL){ |
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pIirCoeffs = _pCoeffs; |
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nIirCoeffs = _nCoeffs; |
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} |
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if (nIirCoeffs != 20){ |
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//Serial.println("Error, in AnalyzePhase, for IIR_LP_FILTER there must be 20 coefficients.");
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nIirCoeffs = 20; |
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} |
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arm_biquad_cascade_df1_init_f32(&iir_inst, N_STAGES, pIirCoeffs, &IIRStateF32[0]); |
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}
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else if (LPType==FIR_LP_FILTER) { |
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if(_pCoeffs != NULL){ |
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pFirCoeffs = _pCoeffs; |
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nFirCoeffs = _nCoeffs; |
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} |
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if (nFirCoeffs<4 || nFirCoeffs>NFIR_MAX) { // Too many or too few
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//Serial.print("Error, in AnalyzePhase, for FIR_LP_FILTER there must be >4 and <=");
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//Serial.print(NFIR_MAX);
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//Serial.println(" coefficients.");
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//Serial.println(" Restoring default IIR Filter.");
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LPType = IIR_LP_FILTER; |
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pIirCoeffs = &iir_coeffs[0]; |
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nIirCoeffs = 20; // Number of coefficients 20
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pdConfig = 0b11100; |
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LPType = IIR_LP_FILTER; // Variables were set in setup() above
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} |
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else { //Acceptable number, so initialize it
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arm_fir_init_f32(&fir_inst, nFirCoeffs, pFirCoeffs, &FIRStateF32[0], block_size); |
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} |
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} |
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pdConfig = _pdConfig; |
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AudioInterrupts();
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} |
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void showError(uint16_t e) { |
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errorPrint = e; |
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} |
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void update(void); |
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private: |
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float32_t sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; |
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uint16_t block_size = AUDIO_BLOCK_SAMPLES; |
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// Two input data pointers
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audio_block_f32_t *inputQueueArray_f32[2]; |
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// Variables controlling the configuration
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uint16_t LPType = IIR_LP_FILTER; // NO_LP_FILTER, IIR_LP_FILTER or FIR_LP_FILTER
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float32_t *pIirCoeffs = &iir_coeffs[0]; // Coefficients for IIR
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float32_t *pFirCoeffs = &fir_coeffs[0]; // Coefficients for FIR
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uint16_t nIirCoeffs = 20; // Number of coefficients 20
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uint16_t nFirCoeffs = 53; // Number of coefficients <=200
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uint16_t pdConfig = 0b11100; // No limiter, fast acos, scale multiplier, radians out;
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// Control error printing in update(). Should never be enabled
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// until all audio objects have been initialized.
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// Only used as 0 or 1 now, but 16 bits are available.
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uint16_t errorPrint = 0; |
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// *Temporary* - TEST_TIME allows measuring time in microseconds for each part of the update()
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#if TEST_TIME |
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elapsedMicros tElapse; |
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int32_t iitt = 998000; // count up to a million during startup
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#endif |
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/* FIR filter designed with http://t-filter.appspot.com
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* Sampling frequency: 44100 Hz |
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* 0 Hz - 4000 Hz gain = 1.0, ripple = 0.101 dB |
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* 7000 - 22000 Hz attenuation >= 81.8 dB |
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* Suitable for measuring phase in communications systems with linear phase. |
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*/ |
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float32_t fir_coeffs[53] = { |
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-0.000206064,-0.000525129,-0.00083518, -0.000774011, 2.5925E-05, |
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0.001614912, 0.003431897, 0.004335125, 0.003127158, -0.000566047, |
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-0.005566484,-0.009192163,-0.008417443,-0.001801824, 0.008839149, |
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0.018273049, 0.019879265, 0.009349346,-0.011696836, -0.034389317, |
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-0.045008839,-0.030706279, 0.013824834, 0.082060266, 0.156328996, |
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0.213799940, 0.235420817, 0.213799940, 0.156328996, 0.082060266, |
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0.013824834,-0.030706279,-0.045008839,-0.034389317, -0.011696836, |
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0.009349346, 0.019879265, 0.018273049, 0.008839149, -0.001801824, |
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-0.008417443,-0.009192163,-0.005566484,-0.000566047, 0.003127158, |
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0.004335125, 0.003431897, 0.001614912, 2.5925E-05, -0.000774011, |
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-0.000835180,-0.000525129,-0.000206064 }; |
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// 8-pole Biquad fc=0.0025fs, -80 dB Iowa Hills
|
||||
// This is roughly the narrowest that doesn't have
|
||||
// artifacts from numerical errors more than about
|
||||
// 0.001 radians (0.06 deg), per experiments using F32.
|
||||
// b0,b1,b2,a1,a2 for each BiQuad. Start with stage 0
|
||||
float32_t iir_coeffs[5 * N_STAGES]={ |
||||
0.08686551007982608, |
||||
-0.1737214710369926, |
||||
0.08686551007982608, |
||||
1.9951804375779567, |
||||
-0.9951899867006161, |
||||
// and stage 1
|
||||
0.20909791845765324, |
||||
-0.4181667739705088, |
||||
0.20909791845765324, |
||||
1.9965910753714984, |
||||
-0.9966201383162961, |
||||
// stage 2
|
||||
0.18360046797931723, |
||||
-0.3671514768697197, |
||||
0.18360046797931723, |
||||
1.9981966389027592, |
||||
-0.998246097991674, |
||||
// stage 3
|
||||
0.03079484444321144, |
||||
-0.061529427044071175, |
||||
0.03079484444321144, |
||||
1.999421284937329, |
||||
-0.9994815467796806}; |
||||
|
||||
// ARM DSP Math library IIR filter instance
|
||||
arm_biquad_casd_df1_inst_f32 iir_inst; |
||||
|
||||
// And a FIR type, as either can be used via begin()
|
||||
arm_fir_instance_f32 fir_inst; |
||||
|
||||
// Delay line space for the FIR
|
||||
float32_t FIRStateF32[AUDIO_BLOCK_SAMPLES + NFIR_MAX]; |
||||
|
||||
// Delay line space for the Biquad, each arranged as {x[n-1], x[n-2], y[n-1], y[n-2]}
|
||||
float32_t IIRStateF32[4 * N_STAGES]; |
||||
}; |
||||
#endif |
||||
@ -0,0 +1,192 @@ |
||||
/* AudioFilterEqualizer_F32.cpp
|
||||
* |
||||
* Bob Larkin, W7PUA 8 May 2020 |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
#include "AudioFilterEqualizer_F32.h" |
||||
|
||||
void AudioFilterEqualizer_F32::update(void) { |
||||
audio_block_f32_t *block, *block_new; |
||||
|
||||
#if TEST_TIME_EQ |
||||
if (iitt++ >1000000) iitt = -10; |
||||
uint32_t t1, t2;
|
||||
t1 = tElapse; |
||||
#endif |
||||
|
||||
block = AudioStream_F32::receiveReadOnly_f32(); |
||||
if (!block) return; |
||||
|
||||
// If there's no coefficient table, give up.
|
||||
if (cf32f == NULL) { |
||||
AudioStream_F32::release(block); |
||||
return; |
||||
} |
||||
|
||||
block_new = AudioStream_F32::allocate_f32(); // get a block for the FIR output
|
||||
if (block_new) { |
||||
//apply the FIR
|
||||
arm_fir_f32(&fir_inst, block->data, block_new->data, block->length); |
||||
AudioStream_F32::transmit(block_new); // send the FIR output
|
||||
AudioStream_F32::release(block_new); |
||||
} |
||||
AudioStream_F32::release(block); |
||||
|
||||
#if TEST_TIME_EQ |
||||
t2 = tElapse; |
||||
if(iitt++ < 0) {Serial.print("At AnalyzePhase end, microseconds = "); Serial.println (t2 - t1); }
|
||||
t1 = tElapse; |
||||
#endif |
||||
} |
||||
|
||||
/* equalizerNew() calculates the Equalizer FIR filter coefficients. Works from:
|
||||
* uint16_t equalizerNew(uint16_t _nBands, float32_t *feq, float32_t *adb, |
||||
uint16_t _nFIR, float32_t *_cf32f, float32_t kdb) |
||||
* nBands Number of equalizer bands |
||||
* feq Pointer to array feq[] of nBands breakpoint frequencies, fractions of sample rate, Hz |
||||
* adb Pointer to array aeq[] of nBands levels, in dB, for the feq[] defined frequency bands |
||||
* nFIR The number of FIR coefficients (taps) used in the equalzer |
||||
* cf32f Pointer to an array of float to hold FIR coefficients |
||||
* kdb A parameter that trades off sidelobe levels for sharpness of band transition. |
||||
* kdb=30 sharp cutoff, poor sidelobes |
||||
* kdb=60 slow cutoff, low sidelobes |
||||
*
|
||||
* The arrays, feq[], aeq[] and cf32f[] are supplied by the calling .INO |
||||
* |
||||
* Returns: 0 if successful, or an error code if not. |
||||
* Errors: 1 = Too many bands, 50 max |
||||
* 2 = sidelobe level out of range, must be > 0 |
||||
* 3 = nFIR out of range |
||||
* |
||||
* Note - This function runs at setup time, and there is no need to fret about |
||||
* processor speed. Likewise, local arrays are created on the stack and are |
||||
* available for other use when this function closes. |
||||
*/ |
||||
uint16_t AudioFilterEqualizer_F32::equalizerNew(uint16_t _nBands, float32_t *feq, float32_t *adb, |
||||
uint16_t _nFIR, float32_t *_cf32f, float32_t kdb) { |
||||
uint16_t i, j; |
||||
uint16_t nHalfFIR; |
||||
float32_t beta, kbes; |
||||
float32_t q, xj2, scaleXj2, WindowWt; |
||||
float32_t fNorm[50]; // Normalized to the sampling frequency
|
||||
float32_t aVolts[50]; // Convert from dB to "quasi-Volts"
|
||||
mathDSP_F32 mathEqualizer; // For Bessel function
|
||||
|
||||
// Make private copies
|
||||
cf32f = _cf32f; |
||||
nFIR = _nFIR; |
||||
nBands = _nBands; |
||||
|
||||
// Check range of nFIR
|
||||
if (nFIR<5 || nFIR>EQUALIZER_MAX_COEFFS) |
||||
return ERR_EQ_NFIR; |
||||
|
||||
// The number of FIR coefficients needs to be odd
|
||||
if (2*(nFIR/2) == nFIR) |
||||
nFIR -= 1; // We just won't use the last element of the array
|
||||
nHalfFIR = (nFIR - 1)/2; // If nFIR=199, nHalfFIR=99
|
||||
|
||||
for (int kk = 0; kk<nFIR; kk++) // To be sure, zero the coefficients
|
||||
cf32f[kk] = 0.0f; |
||||
|
||||
// Convert dB to Voltage ratios, frequencies to fractions of sampling freq
|
||||
if(nBands <2 || nBands>50) return ERR_EQ_BANDS; |
||||
for (i=0; i<nBands; i++) { |
||||
aVolts[i]=powf(10.0, (0.05*adb[i])); |
||||
fNorm[i]=feq[i]/sample_rate_Hz; |
||||
} |
||||
|
||||
/* Find FIR coefficients, the Fourier transform of the frequency
|
||||
* response. This is done by dividing the response into a sequence |
||||
* of nBands rectangular frequency blocks, each of a different level. |
||||
* We can precalculate the Fourier transform for each rectangular band. |
||||
* The linearity of the Fourier transform allows us to sum the transforms |
||||
* of the individual blocks to get pre-windowed coefficients. As follows |
||||
* |
||||
* Numbering example for nFIR==199: |
||||
* Subscript 0 to 98 is 99 taps; 100 to 198 is 99 taps; 99+1+99=199 taps |
||||
* The center coef ( for nFIR=199 taps, nHalfFIR=99 ) is a
|
||||
* special case that comes from sin(0)/0 and treated first: |
||||
*/ |
||||
cf32f[nHalfFIR] = 2.0f*(aVolts[0]*fNorm[0]); // Coefficient "99"
|
||||
for(i=1; i<nBands; i++) { |
||||
cf32f[nHalfFIR] += 2.0f*aVolts[i]*(fNorm[i]-fNorm[i-1]); |
||||
} |
||||
for (j=1; j<=nHalfFIR; j++) { // Coefficients "100 to 198"
|
||||
q = MF_PI*(float32_t)j; |
||||
// First, deal with the zero frequency end band that is "low-pass."
|
||||
cf32f[j+nHalfFIR] = aVolts[0]*sinf(fNorm[0]*2.0*q)/q; |
||||
// and then the rest of the bands that have low and high frequencies
|
||||
for(i=1; i<nBands; i++) |
||||
cf32f[j+nHalfFIR] += aVolts[i]*( (sinf(fNorm[i]*2.0*q)/q) - (sinf(fNorm[i-1]*2.0*q)/q) ); |
||||
} |
||||
|
||||
/* At this point, the cf32f[] coefficients are simply truncated sin(x)/x shapes, creating
|
||||
* very high sidelobe responses. To reduce the sidelobes, a windowing function is applied. |
||||
* This has the side affect of increasing the rate of cutoff for sharp frequency changes. |
||||
* The only windowing function available here is that of James Kaiser. This has a number |
||||
* of desirable features. The tradeoff of sidelobe level versus cutoff rate is variable.
|
||||
* We specify it in terms of kdb, the highest sidelobe, in dB, next to a sharp cutoff. For |
||||
* calculating the windowing vector, we need a parameter beta, found as follows: |
||||
*/ |
||||
if (kdb<0) return ERR_EQ_SIDELOBES; |
||||
if (kdb>50) |
||||
beta = 0.1102*(kdb-8.7); |
||||
else if (kdb>20.96 && kdb<=50.0) |
||||
beta = 0.58417*powf((kdb-20.96), 0.4) + 0.07886*(kdb-20.96); |
||||
else |
||||
beta=0.0; |
||||
// Note: i0f is the floating point in & out zero'th order Bessel function (see mathDSP_F32.h)
|
||||
kbes = 1.0f / mathEqualizer.i0f(beta); // An additional derived parameter used in loop
|
||||
|
||||
// Apply the Kaiser window
|
||||
scaleXj2 = 2.0f/(float32_t)nFIR; |
||||
scaleXj2 *= scaleXj2; |
||||
for (j=0; j<=nHalfFIR; j++) { // For 199 Taps, this is 0 to 99
|
||||
xj2 = (int16_t)(0.5f+(float32_t)j); |
||||
xj2 = scaleXj2*xj2*xj2; |
||||
WindowWt=kbes*(mathEqualizer.i0f(beta*sqrt(1.0-xj2))); |
||||
cf32f[nHalfFIR + j] *= WindowWt; // Apply the Kaiser window to upper half
|
||||
cf32f[nHalfFIR - j] = cf32f[nHalfFIR +j]; // and create the lower half
|
||||
} |
||||
// And fill in the members of fir_inst
|
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size); |
||||
return 0; |
||||
} |
||||
|
||||
/* Calculate response in dB. Leave nFreq point result in array rdb[] supplied
|
||||
* by the calling .INO See Parks and Burris, "Digital Filter Design," p27 (Type 1). |
||||
*/ |
||||
void AudioFilterEqualizer_F32::getResponse(uint16_t nFreq, float32_t *rdb) { |
||||
uint16_t i, j; |
||||
float32_t bt; |
||||
float32_t piOnNfreq; |
||||
uint16_t nHalfFIR; |
||||
|
||||
nHalfFIR = (nFIR - 1)/2; |
||||
piOnNfreq = MF_PI / (float32_t)nFreq; |
||||
for (i=0; i<nFreq; i++) { |
||||
bt = cf32f[nHalfFIR];//bt = 0.5f*cf32f[nHalfFIR]; // Center coefficient
|
||||
for (j=0; j<nHalfFIR; j++) // Add in the others twice, as they are symmetric
|
||||
bt += 2.0f*cf32f[j]*cosf(piOnNfreq*(float32_t)((nHalfFIR-j)*i)); |
||||
rdb[i] = 20.0f*log10f(fabsf(bt)); // Convert to dB
|
||||
} |
||||
} |
||||
@ -0,0 +1,178 @@ |
||||
/*
|
||||
* AudioFilterEqualizer_F32 |
||||
* |
||||
* Created: Bob Larkin W7PUA 8 May 2020 |
||||
*
|
||||
* This is a direct translation of the receiver audio equalizer written |
||||
* by this author for the open-source DSP-10 radio in 1999. See |
||||
* http://www.janbob.com/electron/dsp10/dsp10.htm and
|
||||
* http://www.janbob.com/electron/dsp10/uhf3_35a.zip
|
||||
*
|
||||
* Credit and thanks to PJRC, Paul Stoffregen and Teensy products for the audio |
||||
* system and library that this is built upon as well as the float32 |
||||
* work of Chip Audette embodied in the OpenAudio_ArduinoLibrary. Many thanks |
||||
* for the library structures and wonderful Teensy products. |
||||
* |
||||
* This equalizer is specified by an array of 'nBands' frequency bands |
||||
* each of of arbitrary frequency span. The first band always starts at |
||||
* 0.0 Hz, and that value is not entered. Each band is specified by the upper |
||||
* frequency limit to the band. |
||||
* The last band always ends at half of the sample frequency, which for 44117 Hz |
||||
* sample frequency would be 22058.5. Each band is specified by its upper |
||||
* frequency in an .INO supplied array feq[]. The dB level of that band is |
||||
* specified by a value, in dB, arranged in an .INO supplied array |
||||
* aeq[]. Thus a trivial bass/treble control might look like: |
||||
* nBands = 3; |
||||
* feq[] = {300.0, 1500.0, 22058.5}; |
||||
* float32_t bass = -2.5; // in dB, relative to anything
|
||||
* float32_t treble = 6.0; |
||||
* aeq[] = {bass, 0.0, treble}; |
||||
* |
||||
* It may be obvious that this equalizer is a more general case of the common |
||||
* functions such as low-pass, band-pass, notch, etc. For instance, a pair |
||||
* of band pass filters would look like: |
||||
* nBands = 5; |
||||
* feq[] = {500.0, 700.0, 2000.0, 2200.0, 22058.5}; |
||||
* aeq[] = {-100.0, 0.0, -100.0, 2.0, -100.0}; |
||||
* where we added 2 dB of gain to the 2200 to 2400 Hz filter, relative to the 500 |
||||
* to 700 Hz band. |
||||
* |
||||
* An octave band equalizer is made by starting at some low frequency, say 40 Hz for the |
||||
* first band. The lowest frequency band will be from 0.0 Hz up to that first frequency. |
||||
* Next multiply the first frequency by 2, creating in our example, a band from 40.0 |
||||
* to 80 Hz. This is continued until the last frequency is about 22058 Hz. |
||||
* This works out to require 10 bands, as follows: |
||||
* nBands = 10; |
||||
* feq[] = { 40.0, 80.0, 160.0, 320.0, 640.0, 1280.0, 2560.0, 5120.0, 10240.0, 22058.5}; |
||||
* aeq[] = { 5.0, 4.0, 2.0, -3.0, -4.0, -1.0, 3.0, 6.0, 3.0, 0.5 }; |
||||
* |
||||
* For a "half octave" equalizer, multiply each upper band limit by the square root of 2 = 1.414 |
||||
* to get the next band limit. For that case, feq[] would start with a sequence |
||||
* like 40, 56.56, 80.00, 113.1, 160.0, ... for a total of about 20 bands. |
||||
* |
||||
* How well all of this is achieved depends on the number of FIR coefficients |
||||
* being used. In the Teensy 3.6 / 4.0 the resourses allow a hefty number, |
||||
* say 201, of coefficients to be used without stealing all the processor time |
||||
* (see Timing, below). The coefficient and FIR memory is sized for a maximum of |
||||
* 250 coefficients, but can be recompiled for bigger with the define FIR_MAX_COEFFS. |
||||
* To simplify calculations, the number of FIR coefficients should be odd. If not |
||||
* odd, the number will be reduced by one, quietly. |
||||
* |
||||
* If you try to make the bands too narrow for the number of FIR coeffficients, |
||||
* the approximation to the desired curve becomes poor. This can all be evaluated |
||||
* by the function getResponse(nPoints, pResponse) which fills an .INO-supplied array |
||||
* pResponse[nPoints] with the frequency response of the equalizer in dB. The nPoints |
||||
* are spread evenly between 0.0 and half of the sample frequency. |
||||
* |
||||
* Initialization is a 2-step process. This makes it practical to change equalizer |
||||
* levels on-the-fly. The constructor starts up with a 4-tap FIR setup for direct |
||||
* pass through. Then the setup() in the .INO can specify the equalizer. |
||||
* The newEqualizer() function has several parameters, the number of equalizer bands, |
||||
* the frequencies of the bands, and the sidelobe level. All of these can be changed |
||||
* dynamically. This function can be changed dynamically, but it may be desireable to |
||||
* mute the audio during the change to prevent clicks. |
||||
*
|
||||
* This 16-bit integer version adjusts the maximum coefficient size to scale16 in the calls |
||||
* to both equalizerNew() and getResponse(). Broadband equalizers can work with full-scale |
||||
* 32767.0f sorts of levels, where narrow band filtering may need smaller values to |
||||
* prevent overload. Experiment and check carefully. Use lower values if there are doubts. |
||||
*
|
||||
* For a pass-through function, something like this (which can be intermixed with fancy equalizers): |
||||
* float32_t fBand[] = {10000.0f, 22058.5f};
|
||||
* float32_t dbBand[] = {0.0f, 0.0f}; |
||||
* equalize1.equalizerNew(2, &fBand[0], &dbBand[0], 4, &equalizeCoeffs[0], 30.0f, 32767.0f); |
||||
*
|
||||
* Measured timing of update() for a 128 sample block, Teensy 3.6: |
||||
* Fixed time 13 microseconds |
||||
* Per FIR Coefficient time 2.5 microseconds |
||||
* Total for 199 FIR Coefficients = 505 microseconds (17.4% of 44117 Hz available time) |
||||
*
|
||||
* Per FIR Coefficient, Teensy 4.0, 0.44 microseconds |
||||
*
|
||||
* Copyright (c) 2020 Bob Larkin |
||||
* Any snippets of code from PJRC or Chip Audette used here brings with it |
||||
* the associated license. |
||||
*
|
||||
* In addition, work here is covered by MIT LIcense: |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
#ifndef _filter_equalizer_f32_h |
||||
#define _filter_equalizer_f32_h |
||||
|
||||
#include "Arduino.h" |
||||
#include "AudioStream_F32.h" |
||||
#include "arm_math.h" |
||||
#include "mathDSP_F32.h" |
||||
|
||||
#ifndef MF_PI |
||||
#define MF_PI 3.1415926f |
||||
#endif |
||||
|
||||
// Temporary timing test
|
||||
#define TEST_TIME_EQ 0 |
||||
|
||||
#define EQUALIZER_MAX_COEFFS 251 |
||||
|
||||
#define ERR_EQ_BANDS 1 |
||||
#define ERR_EQ_SIDELOBES 2 |
||||
#define ERR_EQ_NFIR 3 |
||||
|
||||
class AudioFilterEqualizer_F32 : public AudioStream_F32 |
||||
{ |
||||
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
|
||||
//GUI: shortName:filter_Equalizer
|
||||
public: |
||||
AudioFilterEqualizer_F32(void): AudioStream_F32(1,inputQueueArray) { |
||||
// Initialize FIR instance (ARM DSP Math Library) with default simple passthrough FIR
|
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size); |
||||
} |
||||
AudioFilterEqualizer_F32(const AudioSettings_F32 &settings): AudioStream_F32(1,inputQueueArray) { |
||||
block_size = settings.audio_block_samples; |
||||
sample_rate_Hz = settings.sample_rate_Hz; |
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size); |
||||
} |
||||
|
||||
uint16_t equalizerNew(uint16_t _nBands, float32_t *feq, float32_t *adb, |
||||
uint16_t _nFIR, float32_t *_cf32f, float32_t kdb); |
||||
void getResponse(uint16_t nFreq, float32_t *rdb); |
||||
void update(void); |
||||
|
||||
private: |
||||
audio_block_f32_t *inputQueueArray[1]; |
||||
uint16_t block_size = AUDIO_BLOCK_SAMPLES; |
||||
float32_t firStart[4] = {0.0, 1.0, 0.0, 0.0}; // Initialize to passthrough
|
||||
float32_t* cf32f = firStart; // pointer to current coefficients
|
||||
uint16_t nFIR = 4; // Number of coefficients
|
||||
uint16_t nBands = 2; |
||||
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE; |
||||
|
||||
// *Temporary* - TEST_TIME allows measuring time in microseconds for each part of the update()
|
||||
#if TEST_TIME_EQ |
||||
elapsedMicros tElapse; |
||||
int32_t iitt = 999000; // count up to a million during startup
|
||||
#endif |
||||
// ARM DSP Math library filter instance
|
||||
arm_fir_instance_f32 fir_inst; |
||||
float32_t StateF32[AUDIO_BLOCK_SAMPLES + EQUALIZER_MAX_COEFFS]; // max, max
|
||||
}; |
||||
#endif |
||||
|
||||
|
||||
@ -0,0 +1,259 @@ |
||||
/* AudioFilterFIRGeneralr_F32.cpp
|
||||
* |
||||
* Bob Larkin, W7PUA 20 May 2020 (c) |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
/* A math note - The design of the FIR filter from the frequency response requires
|
||||
* taking a discrete Fourier transform. The calculation of the achieved |
||||
* frequency response requires another Fourier transform. Good news is |
||||
* that this is arithmetically simple, because of |
||||
* working with real (not complex) inputs and requiring the FIR be linear |
||||
* phase. One of a number of references is C.S. Burris and T. W. Parks, |
||||
* "Digital Filter Design." That book also has excellent sections on |
||||
* Chebychev error (Parks-McClellan-Rabiner) FIR filters and IIR Filters. |
||||
*/ |
||||
|
||||
#include "AudioFilterFIRGeneral_F32.h" |
||||
|
||||
void AudioFilterFIRGeneral_F32::update(void) { |
||||
audio_block_f32_t *blockIn, *blockOut; |
||||
|
||||
#if TEST_TIME_FIRG |
||||
if (iitt++ >1000000) iitt = -10; |
||||
uint32_t t1, t2; |
||||
t1 = tElapse; |
||||
#endif |
||||
|
||||
blockIn = AudioStream_F32::receiveReadOnly_f32(); |
||||
if (!blockIn) return; |
||||
|
||||
// If there's no coefficient table, give up.
|
||||
if (cf32f == NULL) { |
||||
AudioStream_F32::release(blockIn); |
||||
return; |
||||
} |
||||
|
||||
blockOut = AudioStream_F32::allocate_f32(); // get a block for the FIR output
|
||||
if (blockOut) { |
||||
// The FIR update
|
||||
arm_fir_f32(&fir_inst, blockIn->data, blockOut->data, blockIn->length); |
||||
AudioStream_F32::transmit(blockOut); // send the FIR output
|
||||
AudioStream_F32::release(blockOut); |
||||
} |
||||
AudioStream_F32::release(blockIn); |
||||
|
||||
#if TEST_TIME_FIRG |
||||
t2 = tElapse; |
||||
if(iitt++ < 0) {Serial.print("At FIRGeneral end, microseconds = "); Serial.println (t2 - t1); // }
|
||||
Serial.print("numtaps = "); Serial.println (fir_inst.numTaps);
|
||||
} |
||||
t1 = tElapse; |
||||
#endif |
||||
} |
||||
|
||||
/* FIRGeneralNew() calculates the generalFIR filter coefficients. Works from:
|
||||
* adb Pointer to nFIR/2 array adb[] of levels, in dB |
||||
* nFIR The number of FIR coefficients (taps) used |
||||
* cf32f Pointer to an array of float to hold nFIR FIR coefficients |
||||
* kdb A parameter that trades off sidelobe levels for sharpness of band transition. |
||||
* kdb=30 sharp cutoff, poor sidelobes |
||||
* kdb=60 slow cutoff, low sidelobes |
||||
* pStateArray Pointer to 128+nFIR array of floats, used here briefly and then |
||||
* passed to the FIR routine as working storage |
||||
* |
||||
* The arrays, adb[], cf32f[] and pStateArray[] are supplied by the calling .INO |
||||
* |
||||
* Returns: 0 if successful, or an error code if not. |
||||
* Errors: 1 = NU |
||||
* 2 = sidelobe level out of range, must be > 0 |
||||
* 3 = nFIR out of range |
||||
* |
||||
* Note - This function runs at setup time, so slowness is not an issue |
||||
*/ |
||||
uint16_t AudioFilterFIRGeneral_F32::FIRGeneralNew( |
||||
float32_t *adb, uint16_t _nFIR, float32_t *_cf32f, float32_t kdb, |
||||
float32_t *pStateArray) { |
||||
|
||||
uint16_t i, k, n; |
||||
uint16_t nHalfFIR; |
||||
float32_t beta, kbes; |
||||
float32_t num, xn2, scaleXn2, WindowWt; |
||||
float32_t M; // Burris and Parks, (2.16)
|
||||
mathDSP_F32 mathEqualizer; // For Bessel function
|
||||
bool even; |
||||
|
||||
cf32f = _cf32f; // Make pivate copies
|
||||
nFIR = _nFIR; |
||||
|
||||
// Check range of nFIR
|
||||
if (nFIR<4) |
||||
return ERR_FIRGEN_NFIR; |
||||
|
||||
M = 0.5f*(float32_t)(nFIR - 1); |
||||
// The number of FIR coefficients is even or odd
|
||||
if (2*(nFIR/2) == nFIR) { |
||||
even = true;
|
||||
nHalfFIR = nFIR/2; |
||||
} |
||||
else { |
||||
even = false;
|
||||
nHalfFIR = (nFIR - 1)/2; |
||||
} |
||||
|
||||
for (i=0; i<nFIR; i++) // To be sure, zero the coefficients
|
||||
cf32f[i] = 0.0f; |
||||
for(i=0; i<(nFIR+AUDIO_BLOCK_SAMPLES); i++) // And the storage
|
||||
pStateArray[i] = 0.0f; |
||||
|
||||
// Convert dB to "Voltage" ratios, frequencies to fractions of sampling freq
|
||||
// Borrow pStateArray, as it has not yet gone to ARM math
|
||||
for(i=0; i<=nHalfFIR; i++) |
||||
pStateArray[i] = powf(10.0, 0.05f*adb[i]); |
||||
|
||||
/* Find FIR coefficients, the Fourier transform of the frequency
|
||||
* response. This general frequency description has half as many |
||||
* frequency dB points as FIR coefficients. If nFIR is odd, |
||||
* the number of frequency points is nHalfFIR = (nFIR - 1)/2. |
||||
* |
||||
* Numbering example for nFIR==21: |
||||
* Odd nFIR: Subscript 0 to 9 is 10 taps; 11 to 20 is 10 taps; 10+1+10=21 taps |
||||
* |
||||
* Numbering example for nFIR==20:
|
||||
* Even nFIR: Subscript 0 to 9 is 10 taps; 10 to 19 is 10 taps; 10+10=20 taps |
||||
*/ |
||||
float32_t oneOnNFIR = 1.0f / (float32_t)nFIR; |
||||
if(even) { |
||||
for(n=0; n<nHalfFIR; n++) { // Over half of even nFIR FIR coeffs
|
||||
cf32f[n] = pStateArray[0]; // DC term
|
||||
for(k=1; k<nHalfFIR; k++) { // Over all frequencies, half of nFIR
|
||||
num = (M - (float32_t)n) * (float32_t)k;
|
||||
cf32f[n] += 2.0f*pStateArray[k]*cosf(MF_TWOPI*num*oneOnNFIR); |
||||
} |
||||
cf32f[n] *= oneOnNFIR; // Divide by nFIR
|
||||
}
|
||||
} |
||||
else { // nFIR is odd
|
||||
for(n=0; n<=nHalfFIR; n++) { // Over half of FIR coeffs, nFIR odd
|
||||
cf32f[n] = pStateArray[0]; |
||||
for(k=1; k<=nHalfFIR; k++) { |
||||
num = (float32_t)((nHalfFIR - n)*k); |
||||
cf32f[n] += 2.0f*pStateArray[k]*cosf(MF_TWOPI*num*oneOnNFIR); |
||||
} |
||||
cf32f[n] *= oneOnNFIR; |
||||
}
|
||||
} |
||||
/* At this point, the cf32f[] coefficients are simply truncated, creating
|
||||
* high sidelobe responses. To reduce the sidelobes, a windowing function is applied. |
||||
* This has the side affect of increasing the rate of cutoff for sharp frequency transition. |
||||
* The only windowing function available here is that of James Kaiser. This has a number |
||||
* of desirable features. The sidelobes drop off as the frequency away from a transition. |
||||
* Also, the tradeoff of sidelobe level versus cutoff rate is variable. |
||||
* Here we specify it in terms of kdb, the highest sidelobe, in dB, next to a sharp cutoff. For |
||||
* calculating the windowing vector, we need a parameter beta, found as follows: |
||||
*/ |
||||
if (kdb<0.0f) return ERR_FIRGEN_SIDELOBES; |
||||
if (kdb < 20.0f) |
||||
beta = 0.0; |
||||
else |
||||
beta = -2.17+0.17153*kdb-0.0002841*kdb*kdb; // Within a dB or so
|
||||
// Note: i0f is the fp zero'th order modified Bessel function (see mathDSP_F32.h)
|
||||
kbes = 1.0f / mathEqualizer.i0f(beta); // An additional derived parameter used in loop
|
||||
scaleXn2 = 4.0f / ( ((float32_t)nFIR - 1.0f)*((float32_t)nFIR - 1.0f) ); // Needed for even & odd
|
||||
if (even) { |
||||
// nHalfFIR = nFIR/2; for nFIR even
|
||||
for (n=0; n<nHalfFIR; n++) { // For 20 Taps, this is 0 to 9
|
||||
xn2 = 0.5f+(float32_t)n; |
||||
xn2 = scaleXn2*xn2*xn2; |
||||
WindowWt=kbes*(mathEqualizer.i0f(beta*sqrt(1.0-xn2))); |
||||
if(kdb > 0.1f) // kdb==0.0 means no window
|
||||
cf32f[nHalfFIR - n - 1] *= WindowWt; // Apply window, reverse subscripts
|
||||
cf32f[nHalfFIR + n] = cf32f[nHalfFIR - n - 1]; // and create the upper half
|
||||
} |
||||
} |
||||
else { // nFIR is odd, nHalfFIR = (nFIR - 1)/2
|
||||
for (n=0; n<=nHalfFIR; n++) { // For 21 Taps, this is 0 to 10, including center
|
||||
xn2 = (int16_t)(0.5f+(float32_t)n); |
||||
xn2 = scaleXn2*xn2*xn2; |
||||
WindowWt=kbes*(mathEqualizer.i0f(beta*sqrt(1.0-xn2))); |
||||
if(kdb > 0.1f) |
||||
cf32f[nHalfFIR - n] *= WindowWt; |
||||
// 21 taps, for n=0, nHalfFIR-n = 10, for n=1 nHalfFIR-n=9, for n=nHalfFIR, nHalfFIR-n=0
|
||||
cf32f[nHalfFIR + n] = cf32f[nHalfFIR - n]; // and create upper half (rewrite center is OK)
|
||||
} |
||||
} |
||||
// And finally, fill in the members of fir_inst given in update() to the ARM FIR routine.
|
||||
AudioNoInterrupts(); |
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &pStateArray[0], (uint32_t)block_size); |
||||
AudioInterrupts();
|
||||
return 0; |
||||
} |
||||
|
||||
// FIRGeneralLoad() allows an array of nFIR FIR coefficients to be loaded. They come from an .INO
|
||||
// supplied array. Also, pStateArray[] is .INO supplied and must be (block_size + nFIR) in size.
|
||||
uint16_t AudioFilterFIRGeneral_F32::LoadCoeffs(uint16_t _nFIR, float32_t *_cf32f, float32_t *pStateArray) { |
||||
nFIR = _nFIR; |
||||
cf32f = _cf32f; |
||||
if (nFIR<4) // Check range of nFIR
|
||||
return ERR_FIRGEN_NFIR; |
||||
for(int i=0; i<(nFIR+AUDIO_BLOCK_SAMPLES); i++) // Zero, to be sure
|
||||
pStateArray[i] = 0.0f; |
||||
AudioNoInterrupts();
|
||||
arm_fir_init_f32(&fir_inst, nFIR, &cf32f[0], &pStateArray[0], (uint32_t)block_size); |
||||
AudioInterrupts();
|
||||
return 0; |
||||
} |
||||
|
||||
/* Calculate frequency response in dB. Leave nFreq point result in array rdb[] supplied
|
||||
* by the calling .INO See B&P p27 (Type 1 and 2). Be aware that if nFIR*nFreq is big, |
||||
* like 100,000 or more, this will take a while to calcuulate all the cosf(). Normally, |
||||
* this is not an issue as this is an infrequent calculation. |
||||
* This function assumes that the phase of the FIR is linear with frequency, i.e., |
||||
* the coefficients are symmetrical about the middle. Otherwise, doubling of values, |
||||
* as is done here, is not valid. |
||||
*/ |
||||
void AudioFilterFIRGeneral_F32::getResponse(uint16_t nFreq, float32_t *rdb) { |
||||
uint16_t i, n; |
||||
float32_t bt; |
||||
float32_t piOnNfreq; |
||||
uint16_t nHalfFIR; |
||||
float32_t M; |
||||
|
||||
piOnNfreq = MF_PI / (float32_t)nFreq; |
||||
// The number of FIR coefficients, even or odd?
|
||||
if (2*(nFIR/2) == nFIR) { // it is even
|
||||
nHalfFIR = nFIR/2; |
||||
M = 0.5f*(float32_t)(nFIR - 1); |
||||
for (i=0; i<nFreq; i++) { |
||||
bt = 0.0; |
||||
for (n=0; n<nHalfFIR; n++) // Add in terms twice, as they are symmetric
|
||||
bt += 2.0f*cf32f[n]*cosf(piOnNfreq*((M-(float32_t)n)*(float32_t)i)); |
||||
rdb[i] = 20.0f*log10f(fabsf(bt)); // Convert to dB
|
||||
} |
||||
} |
||||
else { // it is odd
|
||||
nHalfFIR = (nFIR - 1)/2; |
||||
for (i=0; i<nFreq; i++) { |
||||
bt = cf32f[nHalfFIR]; // Center coefficient
|
||||
for (n=0; n<nHalfFIR; n++) // Add in the others twice, as they are symmetric
|
||||
bt += 2.0f*cf32f[n]*cosf(piOnNfreq*(float32_t)((nHalfFIR-n)*i)); |
||||
rdb[i] = 20.0f*log10f(fabsf(bt)); |
||||
} |
||||
}
|
||||
} |
||||
@ -0,0 +1,188 @@ |
||||
/*
|
||||
* AudioFilterFIRGeneral_F32 |
||||
* |
||||
* Created: Bob Larkin W7PUA 20 May 2020 |
||||
* |
||||
* Credit and thanks to PJRC, Paul Stoffregen and Teensy products for the audio |
||||
* system and library that this is built upon as well as the float32 |
||||
* work of Chip Audette embodied in the OpenAudio_ArduinoLibrary. Many thanks |
||||
* for the library structures and wonderful Teensy products. |
||||
* |
||||
* There are enough different FIR filter Audio blocks to need a summary. Here goes: |
||||
*
|
||||
* AudioFilterFIR (Teensy Audio Library by PJRC) handles 16-bit integer data, |
||||
* and a maximum of 200 FIR coefficients, even only. (taps). For Teensy Audio. |
||||
* AudioFilterFIR_F32 (OpenAudio_ArduinoLibrary by Chip Audette) handles 32-bit floating point |
||||
* data and a maximum of 200 taps. Can be used with a mix of Teensy Audio |
||||
* and 32-bit OpenAudio_Arduino blocks. |
||||
* AudioFilterFIRGeneral_F32 (This block) handles 32-bit floating point |
||||
* data and very large number of taps, even or odd. Can be used with a mix of Teensy Audio |
||||
* and 32-bit OpenAudio_Arduino blocks. This includes the design of an |
||||
* arbitrary frequency response using Kaiser windows. |
||||
* AudioFilterFIRGeneral_I16 Same as this block, but data is 16-bit integer and |
||||
* the number of taps must be even. |
||||
* AudioFilterEqualizer_F32 handles 32-bit floating point data and 250 maximum taps |
||||
* even or odd. Can be used with a mix of Teensy Audio |
||||
* and 32-bit OpenAudio_Arduino blocks. This includes the design of an |
||||
* arbitrary frequency response using multiple flat response steps. A Kaiser windows |
||||
* is used. |
||||
* AudioFilterEqualizer_I16 handles 16-bit integer data and 250 maximum taps, |
||||
* even only. |
||||
* |
||||
* FIR filters suffer from needing considerable computation of the multiply-and-add |
||||
* sort. This limits the number of taps that can be used, but less so as time goes by. |
||||
* In particular, the Teensy 4.x, if it *did nothing but* FIR calculations, could |
||||
* use about 6000 taps inmonaural, which is a huge number. But, this also
|
||||
* suggests that if the filtering task is an important function of a project, |
||||
* using, say 2000 taps is practical. |
||||
*
|
||||
* FIR filters can be (and are here) implemented to have symmetrical coefficients. This |
||||
* results in constant delay at all frequencies (linear phase). For some applications this can |
||||
* be an important feature. Sometimes it is suggested that the FIR should not be |
||||
* used because of the latency it creates. Note that if constant delay is needed, the FIR |
||||
* implementation does this with minimum latency. |
||||
*
|
||||
* For this block, AudioFilterFIRGeneral_F32, memory storage for the FIR |
||||
* coefficiients as well as working storage for the ARM FIR routine is provided |
||||
* by the calling .INO. This allows large FIR sizes without always tying up a |
||||
* big memory block. |
||||
*
|
||||
* This block normally calculates the FIR coefficients using a Fourier transform |
||||
* of the desired amplitude response and a Kaiser window. This flexability requires |
||||
* the calling .INO to provide an array of response levels, in relative dB, |
||||
* that is half the length of the number of FIR taps. An example of this entry is a |
||||
* 300 point low-pass filter with a cutoff at 10% of the 44.1 kHz sample frequency: |
||||
* for(int i=0; i<150; i++) { |
||||
* if (i<30) dbA[i] = 0.0f; |
||||
* else dbA[i] = -140.0f; |
||||
* } |
||||
* firg1.FIRGeneralNew(&dbA[0], 300, &equalizeCoeffs[0], 50.0f, &workSpace[0]); |
||||
*
|
||||
* As an alternate to inputting the response function, the FIR coefficients can be |
||||
* entered directly using LoadCoeffs(nFIR, cf32f, *pStateArray). This is a very quick |
||||
* operation as only pointers to coefficients are involved. Several filters can be |
||||
* stored in arrays and switched quickly this way. If this is done, pStateArray[] |
||||
* as initially setup should be large enough for all filters. There will be "clicks" |
||||
* associated with filter changes and these may need to be muted.
|
||||
* |
||||
* How well the desired response is achieved depends on the number of FIR coefficients |
||||
* being used. As noted above, for some applications it may be desired to use |
||||
* large numbers of taps. The achieved response can be evaluated
|
||||
* by the function getResponse(nPoints, pResponse) which fills an .INO-supplied array |
||||
* pResponse[nPoints] with the frequency response of the equalizer in dB. The nPoints |
||||
* are spread evenly between 0.0 and half of the sample frequency. |
||||
* |
||||
* Initialization is a 2-step process. This makes it practical to change equalizer |
||||
* levels on-the-fly. The constructor starts up with a 4-tap FIR setup for direct |
||||
* pass through. Then the setup() in the .INO can specify the equalizer. |
||||
* The newEqualizer() function has several parameters, the number of equalizer bands, |
||||
* the frequencies of the bands, and the sidelobe level. All of these can be changed |
||||
* dynamically. This function can be changed dynamically, but it may be desireable to |
||||
* mute the audio during the change to prevent clicks. |
||||
* |
||||
* Measured timing of update() for L or R 128 sample block, Teensy 3.6: |
||||
* Fixed time 13 microseconds |
||||
* Per FIR Coefficient time 2.5 microseconds (about 51E6 multiply and accumulates/sec) |
||||
* Total for 199 FIR Coefficients = 505 microseconds (17.4% of 44117 Hz available time) |
||||
* Total for stereo is twice those numbers. |
||||
* Measured timing of update() for L or R 128 sample block, Teensy 4.0: |
||||
* Fixed time 1.4 microseconds |
||||
* Per FIR Coefficient time 0.44 microseconds (about 290E6 multiply and accumulate/sec) |
||||
* Total for 199 FIR Coefficients = 90 microseconds (3.1% of 44117 Hz available time) |
||||
* Total for stereo is twice those numbers |
||||
* Measured for FIRGeneralNew(), T4.0, to design a 4000 tap FIR is 14 sec. This goes |
||||
* with the square of the number of taps. |
||||
* Measured for getResponse() for nFIR=4000 and nFreq=5000, T4.0, is about a minute. |
||||
* |
||||
* Functions for the AudioFilterFIRGeneral_F32 object are |
||||
* FIRGeneralNew(*adb, nFIR, cf32f, kdb, *pStateArray); // to design and use an adb[]
|
||||
* frequency response. |
||||
* LoadCoeffs(nFIR, cf32f, *pStateArray); // To directly load FIR coefficients cf32f[]
|
||||
* getResponse(nFreq, *rdb); // To obtain the amplitude response in dB, rdb[]
|
||||
*
|
||||
* Status: Tested T3.6, T4.0 No known bugs |
||||
*
|
||||
* Examples: TestFIRGeneralLarge4.ino TestFIRGeneralLarge5.ino |
||||
*
|
||||
* Copyright (c) 2020 Bob Larkin |
||||
* Any snippets of code from PJRC or Chip Audette used here brings with it |
||||
* the associated license. |
||||
* |
||||
* In addition, work here is covered by MIT LIcense: |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
#ifndef _filter_FIR_general_f32_h |
||||
#define _filter_FIR_general_f32_h |
||||
|
||||
#include "Arduino.h" |
||||
#include "AudioStream_F32.h" |
||||
#include "arm_math.h" |
||||
#include "mathDSP_F32.h" |
||||
|
||||
#ifndef MF_PI |
||||
#define MF_PI 3.1415926f |
||||
#endif |
||||
|
||||
// Temporary timing test
|
||||
#define TEST_TIME_FIRG 0 |
||||
|
||||
#define ERR_FIRGEN_BANDS 1 |
||||
#define ERR_FIRGEN_SIDELOBES 2 |
||||
#define ERR_FIRGEN_NFIR 3 |
||||
|
||||
class AudioFilterFIRGeneral_F32 : public AudioStream_F32 |
||||
{ |
||||
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
|
||||
//GUI: shortName:filter_Equalizer
|
||||
public: |
||||
AudioFilterFIRGeneral_F32(void): AudioStream_F32(1,inputQueueArray) { |
||||
// Initialize FIR instance (ARM DSP Math Library) with default simple passthrough FIR
|
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size); |
||||
} |
||||
AudioFilterFIRGeneral_F32(const AudioSettings_F32 &settings): AudioStream_F32(1,inputQueueArray) { |
||||
block_size = settings.audio_block_samples; |
||||
sample_rate_Hz = settings.sample_rate_Hz; |
||||
arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size); |
||||
} |
||||
|
||||
uint16_t FIRGeneralNew(float32_t *adb, uint16_t _nFIR, float32_t *_cf32f, float32_t kdb, float32_t *pStateArray); |
||||
uint16_t LoadCoeffs(uint16_t _nFIR, float32_t *_cf32f, float32_t *pStateArray); |
||||
void getResponse(uint16_t nFreq, float32_t *rdb); |
||||
void update(void); |
||||
|
||||
private: |
||||
audio_block_f32_t *inputQueueArray[1]; |
||||
uint16_t block_size = AUDIO_BLOCK_SAMPLES; |
||||
float32_t firStart[4] = {0.0, 1.0, 0.0, 0.0}; // Initialize to passthrough
|
||||
float32_t* cf32f = firStart; // pointer to current coefficients
|
||||
uint16_t nFIR = 4; // Number of coefficients
|
||||
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE; |
||||
|
||||
// *Temporary* - TEST_TIME allows measuring time in microseconds for each part of the update()
|
||||
#if TEST_TIME_FIRG |
||||
elapsedMicros tElapse; |
||||
int32_t iitt = 999000; // count up to a million during startup
|
||||
#endif |
||||
// ARM DSP Math library filter instance
|
||||
arm_fir_instance_f32 fir_inst; |
||||
float32_t StateF32[AUDIO_BLOCK_SAMPLES + 4]; // FIR_GENERAL_MAX_COEFFS]; // max, max
|
||||
}; |
||||
#endif |
||||
@ -0,0 +1,136 @@ |
||||
/*
|
||||
* RadioFMDetector_F32.cpp |
||||
* |
||||
* 22 March 2020 |
||||
* Bob Larkin, in support of the library: |
||||
* Chip Audette, OpenAudio, Apr 2017 |
||||
* ------------------- |
||||
* |
||||
* Copyright (c) 2020 Bob Larkin |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*
|
||||
* See RadioFMDetector_F32.h for usage details |
||||
*/ |
||||
#include "RadioFMDetector_F32.h" |
||||
|
||||
// 513 values of the sine wave in a float array:
|
||||
#include "sinTable512_f32.h" |
||||
|
||||
// ==== UPDATE ====
|
||||
void RadioFMDetector_F32::update(void) { |
||||
audio_block_f32_t *blockIn, *blockOut=NULL; |
||||
|
||||
uint16_t i, index_sine; |
||||
float32_t deltaPhase, a, b, dtemp1, dtemp2; |
||||
float32_t v_i[128]; // max size
|
||||
float32_t v_q[128]; |
||||
mathDSP_F32 mathDSP1; // Math support functions
|
||||
|
||||
#if TEST_TIME_FM |
||||
if (iitt++ >1000000) iitt = -10; |
||||
uint32_t t1, t2;
|
||||
t1 = tElapse; |
||||
#endif |
||||
|
||||
// Get input to FM Detector block
|
||||
blockIn = AudioStream_F32::receiveWritable_f32(0); |
||||
if (!blockIn) { |
||||
if(errorPrintFM) Serial.println("FMDET-ERR: No input memory"); |
||||
return; |
||||
} |
||||
|
||||
// If there's no coefficient table, give up.
|
||||
if (fir_IQ_Coeffs == NULL) { |
||||
if(errorPrintFM) Serial.println("FMDET-ERR: No IQ FIR Coefficients"); |
||||
AudioStream_F32::release(blockIn); |
||||
return; |
||||
} |
||||
|
||||
if (fir_Out_Coeffs == NULL) { |
||||
if(errorPrintFM) Serial.println("FMDET-ERR: No Out FIR Coefficients"); |
||||
AudioStream_F32::release(blockIn); |
||||
return; |
||||
} |
||||
|
||||
// Try to get a block for the FM output
|
||||
blockOut = AudioStream_F32::allocate_f32(); |
||||
if (!blockOut){ // Didn't have any
|
||||
if(errorPrintFM) Serial.println("FMDET-ERR: No Output Memory"); |
||||
AudioStream_F32::release(blockIn); |
||||
return; |
||||
} |
||||
|
||||
// Generate sine and cosine of center frequency and double-balance mix
|
||||
// these with the input signal to produce an intermediate result
|
||||
// saved as v_i[] and v_q[]
|
||||
for (i=0; i < block_size; i++) { |
||||
phaseS += phaseIncrement; |
||||
if (phaseS > 512.0f) |
||||
phaseS -= 512.0f; |
||||
index_sine = (uint16_t) phaseS; |
||||
deltaPhase = phaseS -(float32_t) index_sine; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index_sine]; |
||||
b = sinTable512_f32[index_sine+1]; |
||||
// Linear interpolation and multiplying (DBMixer) with input
|
||||
v_i[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase); |
||||
|
||||
/* Repeat for cosine by adding 90 degrees phase */ |
||||
index_sine = (index_sine + 128) & 0x01ff; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index_sine]; |
||||
b = sinTable512_f32[index_sine+1]; |
||||
/* deltaPhase will be the same as used for sin */ |
||||
v_q[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase); |
||||
} |
||||
|
||||
// Do I FIR and Q FIR. We can borrow blockIn and blockOut at this point
|
||||
//void arm_fir_f32( const arm_fir_instance_f32* S, float32_t* pSrc, float32_t* pDst, uint32_t blockSize)
|
||||
arm_fir_f32(&FMDet_I_inst, v_i, blockIn->data, (uint32_t)blockIn->length); |
||||
arm_fir_f32(&FMDet_Q_inst, v_q, blockOut->data, (uint32_t)blockOut->length); |
||||
// Do ATAN2, differentiation and de-emphasis in single loop
|
||||
for(i=0; i<block_size; i++) { // y x
|
||||
dtemp1 = mathDSP1.fastAtan2((float)blockOut->data[i], (float)blockIn->data[i]); |
||||
// Apply differentiator by subtracting last value of atan2
|
||||
if(dtemp1>MF_PI_2 && diffLast<-MF_PI_2) // Probably a wrap around
|
||||
dtemp2 = dtemp1 - diffLast - MF_TWOPI; |
||||
else if(dtemp1<-MF_PI_2 && diffLast >MF_PI_2) // Probably a reverse wrap around
|
||||
dtemp2 = dtemp1 - diffLast + MF_TWOPI; |
||||
else |
||||
dtemp2 = dtemp1 - diffLast; // Differentiate
|
||||
diffLast = dtemp1; // Ready for next time through loop
|
||||
// Data point is now dtemp2. Apply single pole de-emphasis LPF, in place
|
||||
dLast = Kdem * dtemp2 + OneMinusKdem * dLast; |
||||
blockIn->data[i] = dLast; // and save to an array
|
||||
} |
||||
|
||||
// Do output FIR filter. Data now in blockIn.
|
||||
arm_fir_f32(&FMDet_Out_inst, blockIn->data, blockOut->data, (uint32_t)blockIn->length); |
||||
AudioStream_F32::release(blockIn); |
||||
|
||||
// Transmit the data
|
||||
AudioStream_F32::transmit(blockOut, 0); |
||||
AudioStream_F32::release(blockOut); |
||||
#if TEST_TIME_FM |
||||
t2 = tElapse; |
||||
if(iitt++ < 0) {Serial.print("At end of FM Det "); Serial.println (t2 - t1); }
|
||||
t1 = tElapse; |
||||
#endif |
||||
} |
||||
@ -0,0 +1,277 @@ |
||||
/*
|
||||
* RadioFMDetector_F32 |
||||
* 22 March 2020 Bob Larkin |
||||
* With much credit to: |
||||
* Chip Audette (OpenAudio) Feb 2017 |
||||
* Building from AudioFilterFIR from Teensy Audio Library |
||||
* (AudioFilterFIR credited to Pete (El Supremo)) |
||||
* and of course, to PJRC for the Teensy and Teensy Audio Library |
||||
* |
||||
* Copyright (c) 2020 Bob Larkin |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
/* This consists of a single input at some frequency, such as 10 to 20 kHz and
|
||||
* an output, such as 0 to 5 kHz. The output level is linearly dependent on the |
||||
* frequency of the input sine wave frequency, i.e., an it is an FM detector. |
||||
* The input needs to be band limited below the lower frequency side of the |
||||
* input, typically 10 kHz. This is not part of this block. |
||||
*
|
||||
* NOTE: Due to the sample frequencies we are working with, like 44.1 kHz, this |
||||
* detector cannot handle full FM broadcast bandwidths. It is suitable for |
||||
* NBFM as used in communications, marine radio, ham radio, etc. |
||||
* |
||||
* The output can be FIR filtered using default parameters, |
||||
* or using coefficients from an array. A separate single pole de-emphasis filer |
||||
* is included that again can be programmed. |
||||
* |
||||
* Internally, the detector uses a pair of mixers (multipliers) that generate the |
||||
* in-phase and quadrature inputs to a atan2 type of phase detector. These |
||||
* mixers have two output signals at the difference (desired) and sum (undesired) |
||||
* frequencies. The high frequency sum signal can be filtered (For a 15 kHz center, |
||||
* with an input band of 10 to 20 kHz the sum signal will be from 25 to 35 kHz that |
||||
* wraps around the 22 kHz half-sample point to produce 19 to 9 kHz. This needs to |
||||
* be removed before the atan2. A pair of FIR filters, using FIR_IQ_Coeffs |
||||
* are used. These are again programmable and default to a 29-tap LPF with |
||||
* a 5 kHz cutoff. |
||||
*
|
||||
* Status: Tested static, tested with FM modulated Fluke 6061B. |
||||
* An input of about 60 microvolts to the SGTL5000 gave 12 dB SINAD. |
||||
* The output sounded good. Tested T3.6 and T4.0. No known bugs |
||||
*
|
||||
* Output: Float, sensitivity is 2*pi*(f - fCenter)*sample_rate_Hz |
||||
* For 44117Hz samplerate, this is 0.000142421 per Hz |
||||
*
|
||||
* Accuracy: The function used is precise. However, the approximations, such |
||||
* fastAtan2, slightly limit the accuracy. A 200 point sample of a |
||||
* 14 kHz input had an average error of 0.03 Hz
|
||||
* and a standard deviation of 0.81 Hz. |
||||
* The largest errors in this sample were about +/- 1.7 Hz. This is |
||||
* with the default filters. |
||||
* |
||||
* Functions: |
||||
* frequency(float fCenter ) sets the center frequency in Hz, default 15000. |
||||
* |
||||
* filterOut(float *firCoeffs, uint nFIR, float Kdem) sets output filtering where: |
||||
* float32_t* firCoeffs is an array of coefficients |
||||
* uint nFIR is the number of coefficients |
||||
* float32_t Kdem is the de-emphasis frequency factor, where |
||||
* Kdem = 1/(0.5+(tau*fsample)) and tau is the de-emphasis |
||||
* time constant, typically 0.0005 second and fsample is |
||||
* the sample frequency, typically 44117. |
||||
* |
||||
* filterIQ(float *fir_IQ_Coeffs, uint nFIR_IQ) sets output filtering where: |
||||
* float32_t* fir_IQ_Coeffs is an array of coefficients |
||||
* uint nFIR_IQ is the number of coefficients, max 60 |
||||
*
|
||||
* setSampleRate_Hz(float32_t _sampleRate_Hz) allows dynamic changing of |
||||
* the sample rate (experimental as of May 2020). |
||||
*
|
||||
* returnInitializeFMError() Returns the initialization errors. |
||||
* B0001 (value 1) is an error in the IQ FIR Coefficients or quantity. |
||||
* B0010 (value 2) is an error in the Output FIR Coefficients or quantity. |
||||
* B0100 (value 4) is an error in the de-emphasis constant |
||||
* B1000 (value 8) is center frequency above half-sample frequency. |
||||
* All for debug. |
||||
*
|
||||
* showError(uint16_t e) Turns error printing in the update function on (e=1) |
||||
* or off (e=0). For debug only. |
||||
* |
||||
* Time: For T3.6, an update of a 128 sample block, 370 microseconds, or |
||||
* 2.9 microseconds per data point. |
||||
* For T4.0, 87 microseconds, or 0.68 microseconds per data point. |
||||
*
|
||||
* Error checking: See functions setSampleRate_Hz() and returnInitializeFMError() |
||||
* above. |
||||
*/ |
||||
|
||||
#ifndef _radioFMDetector_f32_h |
||||
#define _radioFMDetector_f32_h |
||||
|
||||
#include "mathDSP_F32.h" |
||||
#include "AudioStream_F32.h" |
||||
#include "arm_math.h" |
||||
|
||||
#define MAX_FIR_IQ_COEFFS 100 |
||||
#define MAX_FIR_OUT_COEFFS 120 |
||||
|
||||
#define TEST_TIME_FM 0 |
||||
|
||||
class RadioFMDetector_F32 : public AudioStream_F32 { |
||||
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
|
||||
//GUI: shortName: FMDetector
|
||||
public: |
||||
// Default block size and sample rate:
|
||||
RadioFMDetector_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { |
||||
initializeFM(); |
||||
} |
||||
// Option of AudioSettings_F32 change to block size and/or sample rate:
|
||||
RadioFMDetector_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) { |
||||
sampleRate_Hz = settings.sample_rate_Hz; |
||||
block_size = settings.audio_block_samples; |
||||
initializeFM(); |
||||
} |
||||
|
||||
// Provide for changing input center frequency, in Hz
|
||||
void frequency(float32_t _fCenter) { |
||||
fCenter = _fCenter; |
||||
phaseIncrement = 512.0f * fCenter / sampleRate_Hz; |
||||
} |
||||
|
||||
// Provide for user FIR for I and Q signals to user supplied array
|
||||
void filterIQ(float32_t* _fir_IQ_Coeffs, int _nFIR_IQ) { |
||||
if( fir_IQ_Coeffs==NULL || nFIR_IQ<4 || nFIR_IQ>MAX_FIR_IQ_COEFFS ) { |
||||
initializeFMErrors |= 1; |
||||
return; |
||||
} |
||||
fir_IQ_Coeffs = _fir_IQ_Coeffs; |
||||
nFIR_IQ = _nFIR_IQ; |
||||
initializeFM(); |
||||
} |
||||
|
||||
// Provide for changing to user FIR for detector output, (and user de-emphasis)
|
||||
void filterOut(float32_t *_fir_Out_Coeffs, int _nFIR_Out, float32_t _Kdem) { |
||||
if( _fir_Out_Coeffs==NULL || _nFIR_Out<4 || _nFIR_Out>MAX_FIR_OUT_COEFFS) { |
||||
initializeFMErrors |= 2; |
||||
return; |
||||
} |
||||
if( _Kdem<0.0001 || _Kdem>1.0 ) { |
||||
initializeFMErrors |= 4; |
||||
return; |
||||
} |
||||
fir_Out_Coeffs = _fir_Out_Coeffs; |
||||
nFIR_Out = _nFIR_Out; |
||||
Kdem = _Kdem; |
||||
OneMinusKdem = 1.0f - Kdem; |
||||
initializeFM(); |
||||
} |
||||
|
||||
void setSampleRate_Hz(float32_t _sampleRate_Hz) { |
||||
if (fCenter > _sampleRate_Hz/2.0f) { // Check freq range
|
||||
initializeFMErrors |= 8; |
||||
return; |
||||
} |
||||
sampleRate_Hz = _sampleRate_Hz; |
||||
// update phase increment for new frequency
|
||||
phaseIncrement = 512.0f * fCenter / sampleRate_Hz; |
||||
} |
||||
|
||||
void showError(uint16_t e) { |
||||
errorPrintFM = e; |
||||
} |
||||
|
||||
uint16_t returnInitializeFMError(void) { |
||||
return initializeFMErrors; |
||||
} |
||||
|
||||
void update(void); |
||||
|
||||
private: |
||||
// One input data pointer
|
||||
audio_block_f32_t *inputQueueArray_f32[1]; |
||||
float32_t fCenter = 15000.0f; |
||||
float32_t phaseS = 0.0f; |
||||
float32_t phaseS_C = 128.00f; |
||||
float32_t phaseIncrement = 512.0f*15000.0f/AUDIO_SAMPLE_RATE_EXACT; |
||||
float32_t sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; |
||||
uint16_t block_size = AUDIO_BLOCK_SAMPLES; |
||||
// De-emphasis constant
|
||||
float32_t Kdem = 0.045334f; |
||||
float32_t OneMinusKdem = 0.954666f; |
||||
// Save last data point of atan2 for differentiator
|
||||
float32_t diffLast = 0.0f; |
||||
// Save last data point for next update of de-emphasis filter
|
||||
float32_t dLast = 0.0f; |
||||
// Control error printing in update(), normally off
|
||||
uint16_t errorPrintFM = 0; |
||||
// Monitor constructor errors
|
||||
uint16_t initializeFMErrors = 0; |
||||
uint16_t nFIR_IQ = 29; |
||||
float32_t* fir_IQ_Coeffs = fir_IQ29; |
||||
uint16_t nFIR_Out = 39; |
||||
float32_t* fir_Out_Coeffs = fir_Out39; |
||||
#if TEST_TIME_FM |
||||
elapsedMicros tElapse; |
||||
int32_t iitt = 999000; // count up to a million during startup
|
||||
#endif |
||||
// ARM CMSIS FIR filter instances and State vectors, sized for max, max
|
||||
arm_fir_instance_f32 FMDet_I_inst; |
||||
float32_t State_I_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_IQ_COEFFS]; // 228
|
||||
|
||||
arm_fir_instance_f32 FMDet_Q_inst; |
||||
float32_t State_Q_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_IQ_COEFFS]; // 248
|
||||
|
||||
arm_fir_instance_f32 FMDet_Out_inst; |
||||
float32_t State_Out_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_OUT_COEFFS]; |
||||
|
||||
// Initialize the FM Detector, part of setting up and changing parameters
|
||||
void initializeFM(void) { |
||||
if (fir_IQ_Coeffs && nFIR_IQ <= MAX_FIR_IQ_COEFFS) { |
||||
/* the instance setup call
|
||||
* void arm_fir_init_f32(
|
||||
* arm_fir_instance_f32* S, points to instance of floating-point FIR filter structure. |
||||
* uint16_t numTaps, Number of filter coefficients in the filter. |
||||
* float32_t* pCoeffs, points to the filter coefficients buffer. |
||||
* float32_t* pState, points to the state buffer. |
||||
* uint32_t blockSize) Number of samples that are processed per call. |
||||
*/ |
||||
arm_fir_init_f32(&FMDet_I_inst, nFIR_IQ, (float32_t*)fir_IQ_Coeffs, &State_I_F32[0], (uint32_t)block_size);
|
||||
arm_fir_init_f32(&FMDet_Q_inst, nFIR_IQ, (float32_t*)fir_IQ_Coeffs, &State_Q_F32[0], (uint32_t)block_size);
|
||||
} |
||||
else initializeFMErrors |= B0001;
|
||||
|
||||
if (fir_Out_Coeffs && nFIR_Out <= MAX_FIR_OUT_COEFFS) { |
||||
arm_fir_init_f32(&FMDet_Out_inst, nFIR_Out, (float32_t*)fir_Out_Coeffs, &State_Out_F32[0], (uint32_t)block_size); |
||||
} |
||||
else initializeFMErrors |= B0010; |
||||
dLast = 0.0; |
||||
} |
||||
|
||||
/* FIR filter designed with http://t-filter.appspot.com
|
||||
* fs = 44100 Hz, < 5kHz ripple 0.29 dB, >9 kHz, -62 dB, 29 taps |
||||
*/ |
||||
float32_t fir_IQ29[29] = { |
||||
-0.000970689f, -0.004690292f, -0.008256345f, -0.007565650f, |
||||
0.001524420f, 0.015435011f, 0.021920240f, 0.008211937f, |
||||
-0.024286413f, -0.052184700f, -0.040532507f, 0.031248107f, |
||||
0.146902412f, 0.255179564f, 0.299445269f, 0.255179564f, |
||||
0.146902412f, 0.031248107f, -0.040532507f, -0.052184700f, |
||||
-0.024286413f, 0.008211937f, 0.021920240f, 0.015435011f, |
||||
0.001524420f, -0.007565650f, -0.008256345f, -0.004690292f, |
||||
-0.000970689f}; |
||||
|
||||
/* FIR filter designed with http://t-filter.appspot.com
|
||||
* fs = 44100 Hz, < 3kHz ripple 0.36 dB, >6 kHz, -60 dB, 39 taps |
||||
* Corrected to give DC gain = 1.00
|
||||
*/ |
||||
float32_t fir_Out39[39] = { |
||||
-0.0008908477f, -0.0008401274f, -0.0001837353f, 0.0017556005f,
|
||||
0.0049353322f, 0.0084952916f, 0.0107668722f, 0.0097441685f,
|
||||
0.0039877576f, -0.0063455016f, -0.0188069300f, -0.0287453055f,
|
||||
-0.0303831521f, -0.0186809770f, 0.0085931270f, 0.0493875744f,
|
||||
0.0971742012f, 0.1423015880f, 0.1745838382f, 0.1863024485f,
|
||||
0.1745838382f, 0.1423015880f, 0.0971742012f, 0.0493875744f,
|
||||
0.0085931270f, -0.0186809770f, -0.0303831521f, -0.0287453055f,
|
||||
-0.0188069300f, -0.0063455016f, 0.0039877576f, 0.0097441685f,
|
||||
0.0107668722f, 0.0084952916f, 0.0049353322f, 0.0017556005f,
|
||||
-0.0001837353f, -0.0008401274f, -0.0008908477f }; |
||||
|
||||
}; |
||||
#endif |
||||
@ -0,0 +1,114 @@ |
||||
/* Test AudioAnalyzePhase_F32.cpp RSL 7 April 2020
|
||||
* Generates 2 sine waves of different phase, but same frequency |
||||
* and measures the phase difference. |
||||
*/ |
||||
|
||||
//Include files
|
||||
#include <AudioAnalyzePhase_F32.h> |
||||
//#include <AudioStream_F32.h>
|
||||
//#include <AudioStream.h>
|
||||
//#include <Arduino.h>
|
||||
#include <Audio.h> |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
//#include <math.h>
|
||||
|
||||
//Create audio objects
|
||||
// Input object creates stream, even though not used. I16 object to allow T4.x.
|
||||
AudioInputI2S audioInI2S1; |
||||
AudioConvert_I16toF32 cnvt1; |
||||
// And the objects for the phase measurement:
|
||||
AudioSynthWaveformSine_F32 sine1; |
||||
AudioSynthWaveformSine_F32 sine2; |
||||
AudioAnalyzePhase_F32 phase1; |
||||
AudioRecordQueue_F32 queue1_F;
|
||||
AudioConnection patchCord1(audioInI2S1, 0, cnvt1, 0); |
||||
AudioConnection_F32 patchCord3(sine1, 0, phase1, 0); |
||||
AudioConnection_F32 patchCord4(sine2, 0, phase1, 1); |
||||
AudioConnection_F32 patchCord5(phase1, 0, queue1_F, 0); |
||||
AudioControlSGTL5000 sgtl5000_1;
|
||||
|
||||
#define NBLOCKS 8 |
||||
float dt1[128*NBLOCKS]; |
||||
float *pq, *pd; |
||||
int k, i; |
||||
|
||||
// ==================== SETUP() ============================================
|
||||
void setup(void) { |
||||
//Start the USB serial link (to enable debugging)
|
||||
Serial.begin(300); delay(500); |
||||
|
||||
AudioMemory(2); // Allocate Int16 audio data blocks
|
||||
AudioMemory_F32(15); // Allocate Float32 audio data blocks
|
||||
sgtl5000_1.enable(); |
||||
|
||||
AudioNoInterrupts();
|
||||
sine1.amplitude(1.0); |
||||
sine1.frequency(12345); |
||||
sine1.phase(0.0); |
||||
|
||||
sine2.amplitude(1.0); |
||||
sine2.frequency(12345); |
||||
// The next call sets the phase difference
|
||||
sine2.phase(60.0); // This phase reationship will measure +60 degrees
|
||||
|
||||
AudioInterrupts(); |
||||
|
||||
// The next argument can be set to 1 to show update() errors. But, it will show false
|
||||
// errors before the audiostream is up and running.
|
||||
phase1.showError(0); // For diagnostics
|
||||
|
||||
/* For variable pdConfig (default 0b1100):
|
||||
* Bit 0: 0=No Limiter (default) 1=Use limiter |
||||
* Bit 2 and 1: 00=Use no acos linearizer 01=undefined |
||||
* 10=Fast, math-continuous acos() (default) 11=Accurate acos() |
||||
* Bit 3: 0=No scale of multiplier 1=scale to min-max (default) |
||||
* Bit 4: 0=Output in degrees 1=Output in radians (default) |
||||
*
|
||||
* Uncomment one of the next 4 examples, or leave all 4 commented |
||||
* out and get the default settings for begin(), using about |
||||
* 123 microseconds for 128 block size. |
||||
* Times tu are time spent in update() on T3.6 for full 128 point block. |
||||
*/ |
||||
//#1 - This uses min-max scaling, fast acos(), and FIR filtering
|
||||
phase1.setAnalyzePhaseConfig(FIR_LP_FILTER, NULL, 53, 0b01100); // tu = 213 microseconds
|
||||
|
||||
//#2 - This uses min-max scaling and the Accurate acos()
|
||||
// phase1.setAnalyzePhaseConfig(FIR_LP_FILTER, NULL, 53, 0b01110); // tu <= 531 microseconds
|
||||
|
||||
//#3 - This uses min-max scaling, IIR Filter and the no acos() linearization
|
||||
// phase1.setAnalyzePhaseConfig(IIR_LP_FILTER, NULL, 53, 0b01000); // tu = 96 microseconds
|
||||
|
||||
//#4 - This uses no scaling (use two magnitude 1.0 sine waves),
|
||||
// and the no acos() linearization. No LP filtering
|
||||
// phase1.setAnalyzePhaseConfig(NO_LP_FILTER, NULL, 20, 0b10000); // tu = 28 uSec
|
||||
|
||||
i = 0; k=0; |
||||
queue1_F.begin(); |
||||
Serial.println("Setup complete."); |
||||
}; |
||||
|
||||
void loop(void) {
|
||||
// Collect 128xNBLOCKS samples and output to Serial
|
||||
// This "if" will be active for i on (0, NBLOCKS-1)
|
||||
if (queue1_F.available() >= 1 && i>=0 && i<NBLOCKS) // See if 128 words are available
|
||||
{ |
||||
pq = queue1_F.readBuffer(); |
||||
pd = &dt1[128*i++]; |
||||
for(k=0; k<128; k++) |
||||
{ if(i==2 && k==5) Serial.println(dt1[259]); |
||||
*pd++ = *pq++; // Save 128 words in dt1[]
|
||||
} |
||||
queue1_F.freeBuffer(); |
||||
|
||||
// If 128*NBLOCKS words are saved, send them out Serial
|
||||
if(i == NBLOCKS) |
||||
{ |
||||
i = NBLOCKS + 1; // Should stop all this
|
||||
queue1_F.end(); // No more data to queue1
|
||||
Serial.println("128 x NBLOCKS Data Points:"); |
||||
for (k=0; k<128*NBLOCKS; k++) |
||||
Serial.println (dt1[k],6); |
||||
Serial.println(); |
||||
} |
||||
} |
||||
} |
||||
@ -0,0 +1,58 @@ |
||||
/* AudioTestPeakRMS.ino Bob Larkin 2 May 2020
|
||||
*
|
||||
* Generates sine wave and measures RMS, Peak and Peak |
||||
* to Peak values. |
||||
* NOTE: This is for the floating point _F32 versions of |
||||
* AnalyzePeak and AnalyzeRMS *NOT* the Teensy Audio Library |
||||
* versions with 16-bit Fixed point data blocks. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
#include "analyze_peak_f32.h" |
||||
#include "analyze_rms_f32.h" |
||||
|
||||
AudioInputI2S i2s1;
|
||||
AudioSynthWaveformSine_F32 sine1; |
||||
AudioAnalyzeRMS_F32 rms1; |
||||
AudioAnalyzePeak_F32 peak1; |
||||
AudioConnection_F32 patchCord1(sine1, 0, rms1, 0); |
||||
AudioConnection_F32 patchCord2(sine1, 0, peak1, 0); |
||||
|
||||
uint16_t n = 0; |
||||
|
||||
void setup(void) { |
||||
AudioMemory(5); //allocate Int16 audio data blocks
|
||||
AudioMemory_F32(5); //allocate Float32 audio data blocks
|
||||
Serial.begin(300); delay(1000); |
||||
|
||||
// Default amlitude +/- 1.0
|
||||
sine1.frequency(1000.0); |
||||
|
||||
// Set next to 0 to suppress print errors in update()
|
||||
rms1.showError(1); |
||||
peak1.showError(1); |
||||
} |
||||
|
||||
void loop(void) { |
||||
if (n & 1) { |
||||
while(!rms1.available() ) ; //Wait
|
||||
Serial.print("RMS value = "); |
||||
Serial.println(rms1.read(), 7 ); |
||||
while(!peak1.available() ) ; //Wait
|
||||
Serial.print("Peak value = "); |
||||
Serial.println(peak1.read(), 7 ); |
||||
} |
||||
else { |
||||
while(!rms1.available() ) ; //Wait
|
||||
Serial.print("RMS value = "); |
||||
Serial.println(rms1.read(), 7 ); |
||||
while(!peak1.available() ) ; //Wait
|
||||
Serial.print("Peak to peak value = "); |
||||
Serial.println(peak1.readPeakToPeak(), 7 ); |
||||
} |
||||
n++; |
||||
// The RMS and Peak data collection runs during
|
||||
// delay() because of hardware interrupts
|
||||
delay(1000);
|
||||
} |
||||
@ -0,0 +1,96 @@ |
||||
/* AudioTestSinCos.ino Bob Larkin 19 April 2020
|
||||
* Generates 1024 samples of Sin and Cos from the |
||||
* AudioSynthSinCos_F32. Samples sent to the USB
|
||||
* Serial.print. Also, the sine and cosine signals are |
||||
* connected to the AudioAnalyzePhase_32 and a third |
||||
* Queue to Serial.print the, close to 90 degree, |
||||
* phase difference.
|
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
#define NBLOCKS 8 |
||||
// Necessary foraudio stream
|
||||
AudioInputI2S i2s1; |
||||
AudioConvert_I16toF32 convert;
|
||||
AudioConnection pcI16(i2s1, 0, convert, 0); |
||||
// And the experiment
|
||||
AudioSynthSineCosine_F32 sincos1; |
||||
AudioRecordQueue_F32 queue1; |
||||
AudioRecordQueue_F32 queue2; |
||||
AudioRecordQueue_F32 queue3; |
||||
AudioAnalyzePhase_F32 phase1; |
||||
AudioConnection_F32 patchCord1(sincos1, 0, queue1, 0); |
||||
AudioConnection_F32 patchCord2(sincos1, 1, queue2, 0); |
||||
AudioConnection_F32 patchCord3(sincos1, 0, phase1, 0); |
||||
AudioConnection_F32 patchCord4(sincos1, 1, phase1, 1); |
||||
AudioConnection_F32 patchCord5(phase1, 0, queue3, 0); |
||||
|
||||
float dt1[128*NBLOCKS]; // Place to save sin
|
||||
float dt2[128*NBLOCKS]; // and cos
|
||||
float dt3[128*NBLOCKS]; // and phase angle
|
||||
float *pq1, *pd1, *pq2, *pd2, *pd3, *pq3; |
||||
int i, k; |
||||
|
||||
void setup(void) { |
||||
AudioMemory(5); //allocate Int16 audio data blocks
|
||||
AudioMemory_F32(20); //allocate Float32 audio data blocks
|
||||
Serial.begin(300); delay(1000); |
||||
|
||||
// simple() is not needed here, as it is default unless amplitude or
|
||||
// phaseS_C_r is changed. But, it can be used to restore simple after
|
||||
// involking changes. Here it is just a sample of usage.
|
||||
sincos1.simple(true); |
||||
// Default amlitude +/- 1.0
|
||||
sincos1.frequency(1212.345); |
||||
// If either or both of the following two are uncommented, the
|
||||
// detailed, slower, sincos will be used:
|
||||
// sincos1.amplitude(0.9);
|
||||
// sincos1.phaseS_C_r(120.00*M_PI/180.0); // Test non-90 degree, like 120 deg
|
||||
|
||||
// Set next to 1 to print errors in update(), or 0 for no print. For debug only
|
||||
phase1.showError(0); |
||||
|
||||
queue1.begin(); |
||||
queue2.begin(); |
||||
queue3.begin(); |
||||
i = 0; k=0; |
||||
} |
||||
|
||||
void loop(void) { |
||||
// Collect 128xNBLOCKS samples and output to Serial
|
||||
// This "if" will be active for i on (0, NBLOCKS-1)
|
||||
if (queue1.available() >= 1 && queue2.available() |
||||
&& queue3.available() && i>=0 && i<NBLOCKS) { |
||||
pq1 = queue1.readBuffer(); |
||||
pd1 = &dt1[128*i]; |
||||
pq2 = queue2.readBuffer(); |
||||
pd2 = &dt2[128*i]; |
||||
pq3 = queue3.readBuffer(); |
||||
pd3 = &dt3[128*i++]; |
||||
for(k=0; k<128; k++) { |
||||
*pd1++ = *pq1++; // Save 128 words in dt1[]
|
||||
*pd2++ = *pq2++;
|
||||
*pd3++ = *pq3++;
|
||||
} |
||||
queue1.freeBuffer(); |
||||
queue2.freeBuffer(); |
||||
queue3.freeBuffer(); |
||||
} |
||||
if(i == NBLOCKS) { // Wait for all NBLOCKS
|
||||
i = NBLOCKS + 1; // Should stop data collection
|
||||
queue1.end(); // No more data to queue1
|
||||
queue2.end(); // No more data to queue2
|
||||
|
||||
Serial.println("1024 Sine, Cosine, Phase Data Points:"); |
||||
for (k=0; k<128*NBLOCKS; k++) { |
||||
Serial.print (dt1[k],7); |
||||
Serial.print (","); |
||||
Serial.print (dt2[k],7); |
||||
Serial.print (","); |
||||
Serial.println(dt3[k],7); |
||||
} |
||||
} |
||||
} |
||||
|
After Width: | Height: | Size: 69 KiB |
@ -0,0 +1,107 @@ |
||||
/* ReceiverFM.ino Bob Larkin 26 April 2020
|
||||
* This is a simple test of introducing a sine wave to the |
||||
* FM Detector and taking 512 samples of the output. It is |
||||
* a static test with a fixed frequency for test and so |
||||
* the output DC value and noise can be tested. Note that the 512 |
||||
* samples include the startup transient, so the first 300,
|
||||
* or so, points should be ignored in seeing the DC value. |
||||
*
|
||||
* Change the value of sine1.frequency to see the DC output change. |
||||
* See FMReceiver2.ino for testing with real AC modulation. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
|
||||
// Uncomment the lines "SINE" for internally generated sine wave.
|
||||
// Uncomment the lines "ADC" to use the SGTL5000 Teensy audio adaptor
|
||||
// AudioInputI2S_F32 i2sIn; // ADC
|
||||
AudioSynthWaveformSine_F32 sine1; // SINE
|
||||
RadioFMDetector_F32 fmDet1; |
||||
AudioRecordQueue_F32 queue1;
|
||||
AudioOutputI2S_F32 i2sOut; // Leave in for timing
|
||||
// AudioControlSGTL5000 sgtl5000_1; // ADC
|
||||
|
||||
// AudioConnection_F32 connect0(i2sIn, 0, fmDet1, 0); // ADC
|
||||
AudioConnection_F32 connect0(sine1, 0, fmDet1, 0); // SINE
|
||||
// AudioConnection_F32 connect1(sine1, 0, queue1, 0);
|
||||
AudioConnection_F32 connect3(fmDet1, 0, i2sOut, 0); |
||||
AudioConnection_F32 connect5(fmDet1, 0, queue1, 0); |
||||
|
||||
float dt1[512]; // Place to save output
|
||||
float *pq1, *pd1; |
||||
uint16_t k; |
||||
int i; |
||||
|
||||
void setup(void) { |
||||
AudioMemory(5); |
||||
AudioMemory_F32(5); |
||||
Serial.begin(300); delay(1000); // Any rate is OK
|
||||
Serial.println("Serial Started"); |
||||
|
||||
// sgtl5000_1.enable(); // ADC
|
||||
// sgtl5000_1.inputSelect(AUDIO_INPUT_LINEIN); //ADC
|
||||
|
||||
sine1.frequency(14000.0); // SINE
|
||||
|
||||
// The FM detector has error checking during object construction
|
||||
// when Serial.print is not available. See RadioFMDetector_F32.h:
|
||||
Serial.print("FM Initialization errors: "); |
||||
Serial.println( fmDet1.returnInitializeFMError() ); |
||||
|
||||
// The following enables error checking inside of the "ubdate()"
|
||||
// Output goes to the Serial (USB) Monitor. Normally, this is quiet.
|
||||
fmDet1.showError(1); |
||||
|
||||
queue1.begin();
|
||||
i = 0; k=0; |
||||
} |
||||
|
||||
void loop(void) { |
||||
// Collect 512 samples and output to Serial
|
||||
// This "if" will be active for i = 0,1,2,3
|
||||
if (queue1.available() >= 1) { |
||||
if( i>=0 && i<4) { |
||||
pq1 = queue1.readBuffer(); |
||||
pd1 = &dt1[i*128]; |
||||
for(k=0; k<128; k++) { |
||||
*pd1++ = *pq1++;
|
||||
} |
||||
queue1.freeBuffer(); |
||||
if(i++==3) { |
||||
i=4; // Only collect 4 blocks
|
||||
queue1.end(); // No more data to queue1
|
||||
} |
||||
} |
||||
else { |
||||
queue1.freeBuffer(); |
||||
} |
||||
} |
||||
// We have 512 data samples. Serial.print them
|
||||
if(i == 4) {
|
||||
Serial.println("512 Time in seconds and FM Output samples:"); |
||||
for (k=0; k<512; k++) { |
||||
Serial.print (0.000022667*(float32_t)k, 6); Serial.print (","); |
||||
Serial.println (dt1[k],7); |
||||
} |
||||
i = 5; |
||||
} |
||||
if(i==5) { |
||||
i = 6; |
||||
Serial.print("CPU: Percent Usage, Max: "); |
||||
Serial.print(AudioProcessorUsage()); |
||||
Serial.print(", "); |
||||
Serial.println(AudioProcessorUsageMax()); |
||||
|
||||
Serial.print("Int16 Memory Usage, Max: "); |
||||
Serial.print(AudioMemoryUsage()); |
||||
Serial.print(", "); |
||||
Serial.println(AudioMemoryUsageMax()); |
||||
|
||||
Serial.print("Float Memory Usage, Max: "); |
||||
Serial.print(AudioMemoryUsage_F32()); |
||||
Serial.print(", "); |
||||
Serial.println(AudioMemoryUsageMax_F32()); |
||||
Serial.println(); |
||||
} |
||||
} |
||||
@ -0,0 +1,123 @@ |
||||
// Following is 121 term Hilbert FIR filter
|
||||
float32_t hilbert121A[121] = { |
||||
0.000000000000000000, |
||||
0.000773378567767513, |
||||
0.000000000000000000, |
||||
0.001046207887980644, |
||||
0.000000000000000000, |
||||
0.001368896533613985, |
||||
0.000000000000000000, |
||||
0.001746769975247667, |
||||
0.000000000000000000, |
||||
0.002185555845922462, |
||||
0.000000000000000000, |
||||
0.002691457154069645, |
||||
0.000000000000000000, |
||||
0.003271251311125927, |
||||
0.000000000000000000, |
||||
0.003932423233774751, |
||||
0.000000000000000000, |
||||
0.004683343721596901, |
||||
0.000000000000000000, |
||||
0.005533508538632429, |
||||
0.000000000000000000, |
||||
0.006493859804516438, |
||||
0.000000000000000000, |
||||
0.007577220484233372, |
||||
0.000000000000000000, |
||||
0.008798886675905997, |
||||
0.000000000000000000, |
||||
0.010177443901536392, |
||||
0.000000000000000000, |
||||
0.011735907609641917, |
||||
0.000000000000000000, |
||||
0.013503343224246872, |
||||
0.000000000000000000, |
||||
0.015517212970554440, |
||||
0.000000000000000000, |
||||
0.017826854793349920, |
||||
0.000000000000000000, |
||||
0.020498780519188083, |
||||
0.000000000000000000, |
||||
0.023625003856774591, |
||||
0.000000000000000000, |
||||
0.027336628208641155, |
||||
0.000000000000000000, |
||||
0.031827023036304102, |
||||
0.000000000000000000, |
||||
0.037393534868609392, |
||||
0.000000000000000000, |
||||
0.044517689704988733, |
||||
0.000000000000000000, |
||||
0.054032871748808158, |
||||
0.000000000000000000, |
||||
0.067515548043274365, |
||||
0.000000000000000000, |
||||
0.088347125250410385, |
||||
0.000000000000000000, |
||||
0.125324201622410869, |
||||
0.000000000000000000, |
||||
0.210709715079613419, |
||||
0.000000000000000000, |
||||
0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.210709715079613419, |
||||
0.000000000000000000, |
||||
-0.125324201622410869, |
||||
0.000000000000000000, |
||||
-0.088347125250410385, |
||||
0.000000000000000000, |
||||
-0.067515548043274365, |
||||
0.000000000000000000, |
||||
-0.054032871748808158, |
||||
0.000000000000000000, |
||||
-0.044517689704988733, |
||||
0.000000000000000000, |
||||
-0.037393534868609392, |
||||
0.000000000000000000, |
||||
-0.031827023036304102, |
||||
0.000000000000000000, |
||||
-0.027336628208641155, |
||||
0.000000000000000000, |
||||
-0.023625003856774591, |
||||
0.000000000000000000, |
||||
-0.020498780519188083, |
||||
0.000000000000000000, |
||||
-0.017826854793349920, |
||||
0.000000000000000000, |
||||
-0.015517212970554440, |
||||
0.000000000000000000, |
||||
-0.013503343224246872, |
||||
0.000000000000000000, |
||||
-0.011735907609641917, |
||||
0.000000000000000000, |
||||
-0.010177443901536392, |
||||
0.000000000000000000, |
||||
-0.008798886675905997, |
||||
0.000000000000000000, |
||||
-0.007577220484233372, |
||||
0.000000000000000000, |
||||
-0.006493859804516438, |
||||
0.000000000000000000, |
||||
-0.005533508538632429, |
||||
0.000000000000000000, |
||||
-0.004683343721596901, |
||||
0.000000000000000000, |
||||
-0.003932423233774751, |
||||
0.000000000000000000, |
||||
-0.003271251311125927, |
||||
0.000000000000000000, |
||||
-0.002691457154069645, |
||||
0.000000000000000000, |
||||
-0.002185555845922462, |
||||
0.000000000000000000, |
||||
-0.001746769975247667, |
||||
0.000000000000000000, |
||||
-0.001368896533613985, |
||||
0.000000000000000000, |
||||
-0.001046207887980644, |
||||
0.000000000000000000, |
||||
-0.000773378567767513, |
||||
0.000000000000000000}; |
||||
@ -0,0 +1,21 @@ |
||||
// Following is a 19 term Hilbert FIR filter
|
||||
float hilbert19A[]={ |
||||
0.003006666677728199, |
||||
0.000000000000000000, |
||||
0.017439390086760032, |
||||
0.000000000000000000, |
||||
0.058235158985840196, |
||||
0.000000000000000000, |
||||
0.161905323559397157, |
||||
0.000000000000000000, |
||||
0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.161905323559397157, |
||||
0.000000000000000000, |
||||
-0.058235158985840196, |
||||
0.000000000000000000, |
||||
-0.017439390086760032, |
||||
0.000000000000000000, |
||||
-0.003006666677728199}; |
||||
@ -0,0 +1,253 @@ |
||||
// Following is 251 term Hilbert FIR filter
|
||||
float32_t hilbert251A[]={ |
||||
0.0000003255, |
||||
0.0000000000, |
||||
0.0000030702, |
||||
0.0000000000, |
||||
0.0000089286, |
||||
0.0000000000, |
||||
0.0000183061, |
||||
0.0000000000, |
||||
0.0000316287, |
||||
0.0000000000, |
||||
0.0000493436, |
||||
0.0000000000, |
||||
0.0000719193, |
||||
0.0000000000, |
||||
0.0000998451, |
||||
0.0000000000, |
||||
0.0001336320, |
||||
0.0000000000, |
||||
0.0001738120, |
||||
0.0000000000, |
||||
0.0002209393, |
||||
0.0000000000, |
||||
0.0002755899, |
||||
0.0000000000, |
||||
0.0003383625, |
||||
0.0000000000, |
||||
0.0004098790, |
||||
0.0000000000, |
||||
0.0004907853, |
||||
0.0000000000, |
||||
0.0005817525, |
||||
0.0000000000, |
||||
0.0006834782, |
||||
0.0000000000, |
||||
0.0007966881, |
||||
0.0000000000, |
||||
0.0009221383, |
||||
0.0000000000, |
||||
0.0010606178, |
||||
0.0000000000, |
||||
0.0012129515, |
||||
0.0000000000, |
||||
0.0013800041, |
||||
0.0000000000, |
||||
0.0015626848, |
||||
0.0000000000, |
||||
0.0017619529, |
||||
0.0000000000, |
||||
0.0019788241, |
||||
0.0000000000, |
||||
0.0022143787, |
||||
0.0000000000, |
||||
0.0024697715, |
||||
0.0000000000, |
||||
0.0027462425, |
||||
0.0000000000, |
||||
0.0030451312, |
||||
0.0000000000, |
||||
0.0033678928, |
||||
0.0000000000, |
||||
0.0037161183, |
||||
0.0000000000, |
||||
0.0040915578, |
||||
0.0000000000, |
||||
0.0044961498, |
||||
0.0000000000, |
||||
0.0049320558, |
||||
0.0000000000, |
||||
0.0054017033, |
||||
0.0000000000, |
||||
0.0059078375, |
||||
0.0000000000, |
||||
0.0064535860, |
||||
0.0000000000, |
||||
0.0070425380, |
||||
0.0000000000, |
||||
0.0076788436, |
||||
0.0000000000, |
||||
0.0083673390, |
||||
0.0000000000, |
||||
0.0091137048, |
||||
0.0000000000, |
||||
0.0099246683, |
||||
0.0000000000, |
||||
0.0108082660, |
||||
0.0000000000, |
||||
0.0117741868, |
||||
0.0000000000, |
||||
0.0128342256, |
||||
0.0000000000, |
||||
0.0140028938, |
||||
0.0000000000, |
||||
0.0152982506, |
||||
0.0000000000, |
||||
0.0167430570, |
||||
0.0000000000, |
||||
0.0183664064, |
||||
0.0000000000, |
||||
0.0202060801, |
||||
0.0000000000, |
||||
0.0223120327, |
||||
0.0000000000, |
||||
0.0247516963, |
||||
0.0000000000, |
||||
0.0276183140, |
||||
0.0000000000, |
||||
0.0310445375, |
||||
0.0000000000, |
||||
0.0352256211, |
||||
0.0000000000, |
||||
0.0404611696, |
||||
0.0000000000, |
||||
0.0472354231, |
||||
0.0000000000, |
||||
0.0563851215, |
||||
0.0000000000, |
||||
0.0694911881, |
||||
0.0000000000, |
||||
0.0899418673, |
||||
0.0000000000, |
||||
0.1265473875, |
||||
0.0000000000, |
||||
0.2116132716, |
||||
0.0000000000, |
||||
0.6358933477, |
||||
0.0000000000, |
||||
-0.6358933478, |
||||
0.0000000000, |
||||
-0.2116132717, |
||||
0.0000000000, |
||||
-0.1265473876, |
||||
0.0000000000, |
||||
-0.0899418674, |
||||
0.0000000000, |
||||
-0.0694911882, |
||||
0.0000000000, |
||||
-0.0563851216, |
||||
0.0000000000, |
||||
-0.0472354232, |
||||
0.0000000000, |
||||
-0.0404611697, |
||||
0.0000000000, |
||||
-0.0352256212, |
||||
0.0000000000, |
||||
-0.0310445376, |
||||
0.0000000000, |
||||
-0.0276183141, |
||||
0.0000000000, |
||||
-0.0247516964, |
||||
0.0000000000, |
||||
-0.0223120328, |
||||
0.0000000000, |
||||
-0.0202060802, |
||||
0.0000000000, |
||||
-0.0183664065, |
||||
0.0000000000, |
||||
-0.0167430571, |
||||
0.0000000000, |
||||
-0.0152982507, |
||||
0.0000000000, |
||||
-0.0140028939, |
||||
0.0000000000, |
||||
-0.0128342257, |
||||
0.0000000000, |
||||
-0.0117741869, |
||||
0.0000000000, |
||||
-0.0108082661, |
||||
0.0000000000, |
||||
-0.0099246684, |
||||
0.0000000000, |
||||
-0.0091137049, |
||||
0.0000000000, |
||||
-0.0083673391, |
||||
0.0000000000, |
||||
-0.0076788437, |
||||
0.0000000000, |
||||
-0.0070425381, |
||||
0.0000000000, |
||||
-0.0064535861, |
||||
0.0000000000, |
||||
-0.0059078376, |
||||
0.0000000000, |
||||
-0.0054017034, |
||||
0.0000000000, |
||||
-0.0049320559, |
||||
0.0000000000, |
||||
-0.0044961499, |
||||
0.0000000000, |
||||
-0.0040915579, |
||||
0.0000000000, |
||||
-0.0037161184, |
||||
0.0000000000, |
||||
-0.0033678929, |
||||
0.0000000000, |
||||
-0.0030451313, |
||||
0.0000000000, |
||||
-0.0027462426, |
||||
0.0000000000, |
||||
-0.0024697716, |
||||
0.0000000000, |
||||
-0.0022143788, |
||||
0.0000000000, |
||||
-0.0019788242, |
||||
0.0000000000, |
||||
-0.0017619530, |
||||
0.0000000000, |
||||
-0.0015626849, |
||||
0.0000000000, |
||||
-0.0013800042, |
||||
0.0000000000, |
||||
-0.0012129516, |
||||
0.0000000000, |
||||
-0.0010606179, |
||||
0.0000000000, |
||||
-0.0009221384, |
||||
0.0000000000, |
||||
-0.0007966882, |
||||
0.0000000000, |
||||
-0.0006834783, |
||||
0.0000000000, |
||||
-0.0005817526, |
||||
0.0000000000, |
||||
-0.0004907854, |
||||
0.0000000000, |
||||
-0.0004098791, |
||||
0.0000000000, |
||||
-0.0003383626, |
||||
0.0000000000, |
||||
-0.0002755900, |
||||
0.0000000000, |
||||
-0.0002209394, |
||||
0.0000000000, |
||||
-0.0001738121, |
||||
0.0000000000, |
||||
-0.0001336321, |
||||
0.0000000000, |
||||
-0.0000998452, |
||||
0.0000000000, |
||||
-0.0000719194, |
||||
0.0000000000, |
||||
-0.0000493437, |
||||
0.0000000000, |
||||
-0.0000316288, |
||||
0.0000000000, |
||||
-0.0000183062, |
||||
0.0000000000, |
||||
-0.0000089287, |
||||
0.0000000000, |
||||
-0.0000030703, |
||||
0.0000000000, |
||||
-0.0000003256}; |
||||
Binary file not shown.
@ -0,0 +1,90 @@ |
||||
/* ReceiverPart1.ino Bob Larkin 20 April 2020
|
||||
* This is "Hello World" of Radio design. It can |
||||
* receive a Lower Sideband signal, but that is it.
|
||||
* To test the floating point radio receiver blocks: |
||||
* AudioFilter90Deg_F32 |
||||
* RadioIQMixer_F32 |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
AudioInputI2S i2s1; |
||||
AudioSynthWaveformSine_F32 sine1; // Test signal
|
||||
RadioIQMixer_F32 iqmixer1; |
||||
AudioFilter90Deg_F32 hilbert1; |
||||
// NOTE: Use AudioMixer4_F32 *from Tympan*
|
||||
AudioMixer4_F32 sum1; |
||||
AudioRecordQueue_F32 queue1; // The LSB output
|
||||
AudioRecordQueue_F32 queue2; // The test sine wave
|
||||
|
||||
AudioConnection_F32 patchCord0(sine1, 0, iqmixer1, 0); |
||||
AudioConnection_F32 patchCord2(sine1, 0, iqmixer1, 1); |
||||
AudioConnection_F32 patchCord4(iqmixer1, 0, hilbert1, 0); |
||||
AudioConnection_F32 patchCord5(iqmixer1, 1, hilbert1, 1); |
||||
AudioConnection_F32 patchCord6(hilbert1, 0, sum1, 0); |
||||
AudioConnection_F32 patchCord7(hilbert1, 1, sum1, 1); |
||||
AudioConnection_F32 patchCord8(sum1, 0, queue1, 0); |
||||
AudioConnection_F32 patchCord9(sine1, 0, queue2, 0); |
||||
|
||||
// Pick one of these
|
||||
#include "hilbert19A.h" |
||||
#include "hilbert121A.h" |
||||
#include "hilbert251A.h" |
||||
|
||||
float dt1[128]; |
||||
float dt2[128]; |
||||
float *pq1, *pd1, *pq2, *pd2; |
||||
int i, k; |
||||
|
||||
void setup(void) { |
||||
AudioMemory(5);
|
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
sine1.frequency(14000.0); |
||||
iqmixer1.frequency(16000.0); |
||||
// Pick one of the three, the 251 does not have
|
||||
// enough samples here to show the full build-up.
|
||||
// hilbert1.begin(hilbert19A, 19);
|
||||
// hilbert1.begin(hilbert121A, 121);
|
||||
hilbert1.begin(hilbert121A, 121); |
||||
sum1.gain(0, 1.0); // Leave at 1.0
|
||||
sum1.gain(1, -1.0); // -1 for LSB out
|
||||
iqmixer1.showError(1); // Prints update() errors
|
||||
hilbert1.showError(1); |
||||
queue1.begin(); |
||||
queue2.begin(); |
||||
i = 0; k=0; |
||||
} |
||||
|
||||
void loop(void) { |
||||
// Collect 128 samples and output to Serial
|
||||
// This "if" will be active for i == 0
|
||||
if (queue1.available() >= 1 && i==0) { |
||||
pq1 = queue1.readBuffer(); |
||||
pd1 = &dt1[0]; |
||||
pq2 = queue2.readBuffer(); |
||||
pd2 = &dt2[0]; |
||||
for(k=0; k<128; k++) { |
||||
*pd1++ = *pq1++; |
||||
*pd2++ = *pq2++;
|
||||
} |
||||
i=1; // Only collect 1 block
|
||||
Serial.print("Maximum F32 memory usage at loop:"); |
||||
Serial.println( AudioMemoryUsageMax_F32() ); |
||||
queue1.freeBuffer(); |
||||
queue2.freeBuffer(); |
||||
queue1.end(); // No more data to queue1
|
||||
queue2.end(); // No more data to queue2
|
||||
} |
||||
if(i == 1) {
|
||||
Serial.println("128 Samples: Time (sec), LSB Out, Sine Wave In"); |
||||
for (k=0; k<128; k++) { |
||||
Serial.print (0.000022667*(float32_t)k, 6); Serial.print (","); |
||||
Serial.print (dt1[k],7); Serial.print (","); |
||||
Serial.println (dt2[k],7); |
||||
} |
||||
i = 2; |
||||
} |
||||
} |
||||
@ -0,0 +1,123 @@ |
||||
// Following is 121 term Hilbert FIR filter
|
||||
float32_t hilbert121A[121] = { |
||||
0.000000000000000000, |
||||
0.000773378567767513, |
||||
0.000000000000000000, |
||||
0.001046207887980644, |
||||
0.000000000000000000, |
||||
0.001368896533613985, |
||||
0.000000000000000000, |
||||
0.001746769975247667, |
||||
0.000000000000000000, |
||||
0.002185555845922462, |
||||
0.000000000000000000, |
||||
0.002691457154069645, |
||||
0.000000000000000000, |
||||
0.003271251311125927, |
||||
0.000000000000000000, |
||||
0.003932423233774751, |
||||
0.000000000000000000, |
||||
0.004683343721596901, |
||||
0.000000000000000000, |
||||
0.005533508538632429, |
||||
0.000000000000000000, |
||||
0.006493859804516438, |
||||
0.000000000000000000, |
||||
0.007577220484233372, |
||||
0.000000000000000000, |
||||
0.008798886675905997, |
||||
0.000000000000000000, |
||||
0.010177443901536392, |
||||
0.000000000000000000, |
||||
0.011735907609641917, |
||||
0.000000000000000000, |
||||
0.013503343224246872, |
||||
0.000000000000000000, |
||||
0.015517212970554440, |
||||
0.000000000000000000, |
||||
0.017826854793349920, |
||||
0.000000000000000000, |
||||
0.020498780519188083, |
||||
0.000000000000000000, |
||||
0.023625003856774591, |
||||
0.000000000000000000, |
||||
0.027336628208641155, |
||||
0.000000000000000000, |
||||
0.031827023036304102, |
||||
0.000000000000000000, |
||||
0.037393534868609392, |
||||
0.000000000000000000, |
||||
0.044517689704988733, |
||||
0.000000000000000000, |
||||
0.054032871748808158, |
||||
0.000000000000000000, |
||||
0.067515548043274365, |
||||
0.000000000000000000, |
||||
0.088347125250410385, |
||||
0.000000000000000000, |
||||
0.125324201622410869, |
||||
0.000000000000000000, |
||||
0.210709715079613419, |
||||
0.000000000000000000, |
||||
0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.210709715079613419, |
||||
0.000000000000000000, |
||||
-0.125324201622410869, |
||||
0.000000000000000000, |
||||
-0.088347125250410385, |
||||
0.000000000000000000, |
||||
-0.067515548043274365, |
||||
0.000000000000000000, |
||||
-0.054032871748808158, |
||||
0.000000000000000000, |
||||
-0.044517689704988733, |
||||
0.000000000000000000, |
||||
-0.037393534868609392, |
||||
0.000000000000000000, |
||||
-0.031827023036304102, |
||||
0.000000000000000000, |
||||
-0.027336628208641155, |
||||
0.000000000000000000, |
||||
-0.023625003856774591, |
||||
0.000000000000000000, |
||||
-0.020498780519188083, |
||||
0.000000000000000000, |
||||
-0.017826854793349920, |
||||
0.000000000000000000, |
||||
-0.015517212970554440, |
||||
0.000000000000000000, |
||||
-0.013503343224246872, |
||||
0.000000000000000000, |
||||
-0.011735907609641917, |
||||
0.000000000000000000, |
||||
-0.010177443901536392, |
||||
0.000000000000000000, |
||||
-0.008798886675905997, |
||||
0.000000000000000000, |
||||
-0.007577220484233372, |
||||
0.000000000000000000, |
||||
-0.006493859804516438, |
||||
0.000000000000000000, |
||||
-0.005533508538632429, |
||||
0.000000000000000000, |
||||
-0.004683343721596901, |
||||
0.000000000000000000, |
||||
-0.003932423233774751, |
||||
0.000000000000000000, |
||||
-0.003271251311125927, |
||||
0.000000000000000000, |
||||
-0.002691457154069645, |
||||
0.000000000000000000, |
||||
-0.002185555845922462, |
||||
0.000000000000000000, |
||||
-0.001746769975247667, |
||||
0.000000000000000000, |
||||
-0.001368896533613985, |
||||
0.000000000000000000, |
||||
-0.001046207887980644, |
||||
0.000000000000000000, |
||||
-0.000773378567767513, |
||||
0.000000000000000000}; |
||||
@ -0,0 +1,21 @@ |
||||
// Following is a 19 term Hilbert FIR filter
|
||||
float hilbert19A[]={ |
||||
0.003006666677728199, |
||||
0.000000000000000000, |
||||
0.017439390086760032, |
||||
0.000000000000000000, |
||||
0.058235158985840196, |
||||
0.000000000000000000, |
||||
0.161905323559397157, |
||||
0.000000000000000000, |
||||
0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.161905323559397157, |
||||
0.000000000000000000, |
||||
-0.058235158985840196, |
||||
0.000000000000000000, |
||||
-0.017439390086760032, |
||||
0.000000000000000000, |
||||
-0.003006666677728199}; |
||||
@ -0,0 +1,253 @@ |
||||
// Following is 251 term Hilbert FIR filter
|
||||
float32_t hilbert251A[]={ |
||||
0.0000003255, |
||||
0.0000000000, |
||||
0.0000030702, |
||||
0.0000000000, |
||||
0.0000089286, |
||||
0.0000000000, |
||||
0.0000183061, |
||||
0.0000000000, |
||||
0.0000316287, |
||||
0.0000000000, |
||||
0.0000493436, |
||||
0.0000000000, |
||||
0.0000719193, |
||||
0.0000000000, |
||||
0.0000998451, |
||||
0.0000000000, |
||||
0.0001336320, |
||||
0.0000000000, |
||||
0.0001738120, |
||||
0.0000000000, |
||||
0.0002209393, |
||||
0.0000000000, |
||||
0.0002755899, |
||||
0.0000000000, |
||||
0.0003383625, |
||||
0.0000000000, |
||||
0.0004098790, |
||||
0.0000000000, |
||||
0.0004907853, |
||||
0.0000000000, |
||||
0.0005817525, |
||||
0.0000000000, |
||||
0.0006834782, |
||||
0.0000000000, |
||||
0.0007966881, |
||||
0.0000000000, |
||||
0.0009221383, |
||||
0.0000000000, |
||||
0.0010606178, |
||||
0.0000000000, |
||||
0.0012129515, |
||||
0.0000000000, |
||||
0.0013800041, |
||||
0.0000000000, |
||||
0.0015626848, |
||||
0.0000000000, |
||||
0.0017619529, |
||||
0.0000000000, |
||||
0.0019788241, |
||||
0.0000000000, |
||||
0.0022143787, |
||||
0.0000000000, |
||||
0.0024697715, |
||||
0.0000000000, |
||||
0.0027462425, |
||||
0.0000000000, |
||||
0.0030451312, |
||||
0.0000000000, |
||||
0.0033678928, |
||||
0.0000000000, |
||||
0.0037161183, |
||||
0.0000000000, |
||||
0.0040915578, |
||||
0.0000000000, |
||||
0.0044961498, |
||||
0.0000000000, |
||||
0.0049320558, |
||||
0.0000000000, |
||||
0.0054017033, |
||||
0.0000000000, |
||||
0.0059078375, |
||||
0.0000000000, |
||||
0.0064535860, |
||||
0.0000000000, |
||||
0.0070425380, |
||||
0.0000000000, |
||||
0.0076788436, |
||||
0.0000000000, |
||||
0.0083673390, |
||||
0.0000000000, |
||||
0.0091137048, |
||||
0.0000000000, |
||||
0.0099246683, |
||||
0.0000000000, |
||||
0.0108082660, |
||||
0.0000000000, |
||||
0.0117741868, |
||||
0.0000000000, |
||||
0.0128342256, |
||||
0.0000000000, |
||||
0.0140028938, |
||||
0.0000000000, |
||||
0.0152982506, |
||||
0.0000000000, |
||||
0.0167430570, |
||||
0.0000000000, |
||||
0.0183664064, |
||||
0.0000000000, |
||||
0.0202060801, |
||||
0.0000000000, |
||||
0.0223120327, |
||||
0.0000000000, |
||||
0.0247516963, |
||||
0.0000000000, |
||||
0.0276183140, |
||||
0.0000000000, |
||||
0.0310445375, |
||||
0.0000000000, |
||||
0.0352256211, |
||||
0.0000000000, |
||||
0.0404611696, |
||||
0.0000000000, |
||||
0.0472354231, |
||||
0.0000000000, |
||||
0.0563851215, |
||||
0.0000000000, |
||||
0.0694911881, |
||||
0.0000000000, |
||||
0.0899418673, |
||||
0.0000000000, |
||||
0.1265473875, |
||||
0.0000000000, |
||||
0.2116132716, |
||||
0.0000000000, |
||||
0.6358933477, |
||||
0.0000000000, |
||||
-0.6358933478, |
||||
0.0000000000, |
||||
-0.2116132717, |
||||
0.0000000000, |
||||
-0.1265473876, |
||||
0.0000000000, |
||||
-0.0899418674, |
||||
0.0000000000, |
||||
-0.0694911882, |
||||
0.0000000000, |
||||
-0.0563851216, |
||||
0.0000000000, |
||||
-0.0472354232, |
||||
0.0000000000, |
||||
-0.0404611697, |
||||
0.0000000000, |
||||
-0.0352256212, |
||||
0.0000000000, |
||||
-0.0310445376, |
||||
0.0000000000, |
||||
-0.0276183141, |
||||
0.0000000000, |
||||
-0.0247516964, |
||||
0.0000000000, |
||||
-0.0223120328, |
||||
0.0000000000, |
||||
-0.0202060802, |
||||
0.0000000000, |
||||
-0.0183664065, |
||||
0.0000000000, |
||||
-0.0167430571, |
||||
0.0000000000, |
||||
-0.0152982507, |
||||
0.0000000000, |
||||
-0.0140028939, |
||||
0.0000000000, |
||||
-0.0128342257, |
||||
0.0000000000, |
||||
-0.0117741869, |
||||
0.0000000000, |
||||
-0.0108082661, |
||||
0.0000000000, |
||||
-0.0099246684, |
||||
0.0000000000, |
||||
-0.0091137049, |
||||
0.0000000000, |
||||
-0.0083673391, |
||||
0.0000000000, |
||||
-0.0076788437, |
||||
0.0000000000, |
||||
-0.0070425381, |
||||
0.0000000000, |
||||
-0.0064535861, |
||||
0.0000000000, |
||||
-0.0059078376, |
||||
0.0000000000, |
||||
-0.0054017034, |
||||
0.0000000000, |
||||
-0.0049320559, |
||||
0.0000000000, |
||||
-0.0044961499, |
||||
0.0000000000, |
||||
-0.0040915579, |
||||
0.0000000000, |
||||
-0.0037161184, |
||||
0.0000000000, |
||||
-0.0033678929, |
||||
0.0000000000, |
||||
-0.0030451313, |
||||
0.0000000000, |
||||
-0.0027462426, |
||||
0.0000000000, |
||||
-0.0024697716, |
||||
0.0000000000, |
||||
-0.0022143788, |
||||
0.0000000000, |
||||
-0.0019788242, |
||||
0.0000000000, |
||||
-0.0017619530, |
||||
0.0000000000, |
||||
-0.0015626849, |
||||
0.0000000000, |
||||
-0.0013800042, |
||||
0.0000000000, |
||||
-0.0012129516, |
||||
0.0000000000, |
||||
-0.0010606179, |
||||
0.0000000000, |
||||
-0.0009221384, |
||||
0.0000000000, |
||||
-0.0007966882, |
||||
0.0000000000, |
||||
-0.0006834783, |
||||
0.0000000000, |
||||
-0.0005817526, |
||||
0.0000000000, |
||||
-0.0004907854, |
||||
0.0000000000, |
||||
-0.0004098791, |
||||
0.0000000000, |
||||
-0.0003383626, |
||||
0.0000000000, |
||||
-0.0002755900, |
||||
0.0000000000, |
||||
-0.0002209394, |
||||
0.0000000000, |
||||
-0.0001738121, |
||||
0.0000000000, |
||||
-0.0001336321, |
||||
0.0000000000, |
||||
-0.0000998452, |
||||
0.0000000000, |
||||
-0.0000719194, |
||||
0.0000000000, |
||||
-0.0000493437, |
||||
0.0000000000, |
||||
-0.0000316288, |
||||
0.0000000000, |
||||
-0.0000183062, |
||||
0.0000000000, |
||||
-0.0000089287, |
||||
0.0000000000, |
||||
-0.0000030703, |
||||
0.0000000000, |
||||
-0.0000003256}; |
||||
@ -0,0 +1,177 @@ |
||||
/* ReceiverPart2.ino Bob Larkin 29 April 2020
|
||||
* This is a simple SP radio design. It can receive 2 modes, |
||||
* Single Sideband (SSB) and Narrow Band FM (NBFM). SSB |
||||
* breaks into Lower Sidband (LSB) and Upper Sideband (USB). |
||||
* It gets even better in that AM can be received on either |
||||
* LSB or USB by tuning to the AM carrier frequency. |
||||
*
|
||||
* The signal path is switched between SSB and FM by a Chip |
||||
* Audette AudioSwitch4_F32 switch block. This keeps resources |
||||
* from being used in blocks that are not needed. |
||||
*
|
||||
* We are restricted to receiving in the 8 to 22 kHz range, |
||||
* so to use this on the air would require frequency conversion |
||||
* hardware. |
||||
* |
||||
* Details on the blocks are part of the include .h files for the |
||||
* blocks and in the INO files |
||||
* TestFM.ino |
||||
* ReceiverPart1.ino |
||||
*
|
||||
* Input and Output is via the Teensy Audio Adaptor that uses |
||||
* a SGTL5000 CODEC, |
||||
* |
||||
* See the individual .h files for each block for more details. |
||||
*
|
||||
* Measured peak-to-peak levels, using AudioAnalyzePeak_F32, |
||||
* all done at overload point for ADC: |
||||
* At ADC (i2sIn), max, 2.0 |
||||
* At IQ Mixer out, max, 2.0 |
||||
* At 90 deg Phase out, 2.0 |
||||
* At Sum out 2.0 |
||||
* At LPF FIR Out 2.0 |
||||
*
|
||||
* FM Det gives 0.50 out for about 5.6 kHz p-p deviation |
||||
*
|
||||
* T3.6 Processor load, measured: 16% for NBFM |
||||
* 30% for LSB or USB 29 tap LPF |
||||
* T4.0 Processor load, measured: 4.3% for NBFM |
||||
* 6.5% for LSB or USB 29 tap LPF |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include <OpenAudio_ArduinoLibrary.h> |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
// ********* Mini Control Panel *********
|
||||
// Set mode and gain here and re-compile
|
||||
|
||||
// Here is the mode switch
|
||||
#define LSB 1 |
||||
#define USB 2 |
||||
#define NBFM 3 |
||||
uint16_t mode = NBFM; // <--Select mode
|
||||
|
||||
int gainControlDB = 0; // Set SSB gain in dB. 0 dB is a gain of 1.0
|
||||
|
||||
// *****************************************
|
||||
// To work with T4.0 the I2S routine outputs 16-bit integer (I16). Then
|
||||
// use Audette I16 to F32 convert. Same below for output, in reverse.
|
||||
AudioInputI2S i2sIn; |
||||
AudioConvert_I16toF32 cnvrt1; |
||||
AudioSwitch4_F32 switch1; // Select SSB or FM
|
||||
RadioIQMixer_F32 iqmixer1; |
||||
AudioFilter90Deg_F32 hilbert1; |
||||
AudioMixer4_F32 sum1; // Summing node for the SSB receiver
|
||||
AudioFilterFIR_F32 fir1; // Low Pass Filter to frequency limit the SSB
|
||||
RadioFMDetector_F32 fmdet1; // NBFM from 10 to 20 kHz
|
||||
AudioMixer4_F32 sum2; // SSB and NBFM rejoin here
|
||||
AudioConvert_F32toI16 cnvrt2; // Left
|
||||
AudioConvert_F32toI16 cnvrt3; // Right
|
||||
AudioOutputI2S i2sOut; |
||||
AudioAnalyzePeak_F32 peak1; |
||||
AudioControlSGTL5000 sgtl5000_1; |
||||
|
||||
AudioConnection conI16_1(i2sIn, 0, cnvrt1, 0); // ADC
|
||||
AudioConnection_F32 connect0(cnvrt1, 0, switch1, 0); // Analog to Digital
|
||||
AudioConnection_F32 connect1(switch1, 0, iqmixer1, 0); // SSB Input
|
||||
AudioConnection_F32 connect2(switch1, 0, iqmixer1, 1); // SSB Input
|
||||
AudioConnection_F32 connect3(switch1, 1, fmdet1, 0); // FM input
|
||||
AudioConnection_F32 connect4(iqmixer1, 0, hilbert1, 0); // Broadband 90 deg phase
|
||||
AudioConnection_F32 connect5(iqmixer1, 1, hilbert1, 1); |
||||
AudioConnection_F32 connect6(hilbert1, 0, sum1, 0); // Sideband select
|
||||
AudioConnection_F32 connect7(hilbert1, 1, sum1, 1); |
||||
AudioConnection_F32 connect8(sum1, 0, fir1, 0); // Limit audio BW
|
||||
AudioConnection_F32 connect9(fir1, 0, sum2, 0); // Output of SSB
|
||||
AudioConnection_F32 connectA(fmdet1, 0, sum2, 1); // Output of FM
|
||||
AudioConnection_F32 connectC(sum2, 0, cnvrt2, 0); // Out to the CODEC left
|
||||
AudioConnection_F32 connectD(sum2, 0, cnvrt3, 0); // and right
|
||||
AudioConnection_F32 connectE(sum2, 0, peak1, 0); |
||||
AudioConnection conI16_2(cnvrt2, 0, i2sOut, 0); // DAC
|
||||
AudioConnection conI16_3(cnvrt3, 0, i2sOut, 1); // DAC
|
||||
|
||||
// Filter for AudioFilter90Deg_F32 hilbert1
|
||||
#include "hilbert251A.h" |
||||
|
||||
/* FIR filter designed with http://t-filter.appspot.com
|
||||
* fs = 44100 Hz, < 5kHz ripple 0.29 dB, >9 kHz, -62 dB, 29 taps |
||||
*/ |
||||
float32_t fir_IQ29[29] = { |
||||
-0.000970689f, -0.004690292f, -0.008256345f, -0.007565650f, |
||||
0.001524420f, 0.015435011f, 0.021920240f, 0.008211937f, |
||||
-0.024286413f, -0.052184700f, -0.040532507f, 0.031248107f, |
||||
0.146902412f, 0.255179564f, 0.299445269f, 0.255179564f, |
||||
0.146902412f, 0.031248107f, -0.040532507f, -0.052184700f, |
||||
-0.024286413f, 0.008211937f, 0.021920240f, 0.015435011f, |
||||
0.001524420f, -0.007565650f, -0.008256345f, -0.004690292f, |
||||
-0.000970689f}; |
||||
|
||||
void setup(void) { |
||||
float32_t vGain; |
||||
AudioMemory(5); |
||||
AudioMemory_F32(5); |
||||
Serial.begin(300); delay(1000); |
||||
|
||||
// Enable the audio shield, select input, and enable output
|
||||
sgtl5000_1.enable(); // ADC
|
||||
sgtl5000_1.inputSelect(AUDIO_INPUT_LINEIN); |
||||
|
||||
// The switch is single pole 4 position, numbered (0, 3) 0=SSB, 1 = FM
|
||||
if(mode==LSB || mode==USB){ switch1.setChannel(0); Serial.println("SSB"); } |
||||
else if(mode==NBFM) { switch1.setChannel(1); Serial.println("NBFM"); } |
||||
|
||||
iqmixer1.frequency(15000.0); // LO Frequency, typically 10 to 15 kHz
|
||||
|
||||
hilbert1.begin(hilbert251A, 251); // Set the Hilbert transform FIR filter
|
||||
|
||||
sum1.gain(0, 1.0f); // Leave set at 1.0
|
||||
if(mode==LSB) sum1.gain(1, -1.0f); // -1 for LSB out
|
||||
else if(mode==USB) sum1.gain(1, 1.0f); // +1 for USB and for NBFM, we don't care
|
||||
|
||||
// The FM detector has error checking during object construction
|
||||
// when Serial.print is not available. See RadioFMDetector_F32.h:
|
||||
Serial.print("FM Initialization errors: "); |
||||
Serial.println( fmdet1.returnInitializeFMError() ); |
||||
|
||||
// See RadioFMDetector_F32.h for information on functions for modifying the
|
||||
// FM Detector. Default values are used here, starting with a 15 kHz center frequency.
|
||||
|
||||
fir1.begin(fir_IQ29, 29); // 5 kHz bandwidth set for radio in SSB
|
||||
|
||||
// The gainControlDB goes in 1 dB steps. Convert here to a voltage ratio
|
||||
vGain = powf(10.0f, ((float32_t)gainControlDB)/20.0 ); |
||||
// And apply that ratio to the output summing block. Gain here for SSB only
|
||||
sum2.gain(0, vGain); Serial.print("vGain = "); Serial.println(vGain, 4); |
||||
sum2.gain(1, 1.0f); // FM gain
|
||||
|
||||
// The following enable error checking inside of the blocks indicated.
|
||||
// Output goes to the Serial (USB) Monitor. Use for debug.
|
||||
//hilbert1.showError(1); // Should show input error when in FM
|
||||
//fmdet1.showError(1); // Should show input error when in LSB or USB
|
||||
} |
||||
|
||||
void loop(void) { |
||||
// Here is where the adjustment of the volume control could go.
|
||||
// And anything else that needs regular attention, other
|
||||
// than the audio stream.
|
||||
|
||||
if (peak1.available() ) {Serial.print("P-P ="); Serial.println(peak1.readPeakToPeak(), 6);} |
||||
else Serial.println("Peak-Peak not available"); |
||||
Serial.print("CPU: Percent Usage, Max: "); |
||||
Serial.print(AudioProcessorUsage()); |
||||
Serial.print(", "); |
||||
Serial.print(AudioProcessorUsageMax()); |
||||
Serial.print(" "); |
||||
Serial.print("Int16 Memory: "); |
||||
Serial.print(AudioMemoryUsage()); |
||||
Serial.print(", "); |
||||
Serial.print(AudioMemoryUsageMax()); |
||||
Serial.print(" "); |
||||
Serial.print("Float Memory: "); |
||||
Serial.print(AudioMemoryUsage_F32()); |
||||
Serial.print(", "); |
||||
Serial.println(AudioMemoryUsageMax_F32()); |
||||
Serial.println(); |
||||
|
||||
delay(1000); |
||||
}
|
||||
@ -0,0 +1,123 @@ |
||||
// Following is 121 term Hilbert FIR filter
|
||||
float32_t hilbert121A[121] = { |
||||
0.000000000000000000, |
||||
0.000773378567767513, |
||||
0.000000000000000000, |
||||
0.001046207887980644, |
||||
0.000000000000000000, |
||||
0.001368896533613985, |
||||
0.000000000000000000, |
||||
0.001746769975247667, |
||||
0.000000000000000000, |
||||
0.002185555845922462, |
||||
0.000000000000000000, |
||||
0.002691457154069645, |
||||
0.000000000000000000, |
||||
0.003271251311125927, |
||||
0.000000000000000000, |
||||
0.003932423233774751, |
||||
0.000000000000000000, |
||||
0.004683343721596901, |
||||
0.000000000000000000, |
||||
0.005533508538632429, |
||||
0.000000000000000000, |
||||
0.006493859804516438, |
||||
0.000000000000000000, |
||||
0.007577220484233372, |
||||
0.000000000000000000, |
||||
0.008798886675905997, |
||||
0.000000000000000000, |
||||
0.010177443901536392, |
||||
0.000000000000000000, |
||||
0.011735907609641917, |
||||
0.000000000000000000, |
||||
0.013503343224246872, |
||||
0.000000000000000000, |
||||
0.015517212970554440, |
||||
0.000000000000000000, |
||||
0.017826854793349920, |
||||
0.000000000000000000, |
||||
0.020498780519188083, |
||||
0.000000000000000000, |
||||
0.023625003856774591, |
||||
0.000000000000000000, |
||||
0.027336628208641155, |
||||
0.000000000000000000, |
||||
0.031827023036304102, |
||||
0.000000000000000000, |
||||
0.037393534868609392, |
||||
0.000000000000000000, |
||||
0.044517689704988733, |
||||
0.000000000000000000, |
||||
0.054032871748808158, |
||||
0.000000000000000000, |
||||
0.067515548043274365, |
||||
0.000000000000000000, |
||||
0.088347125250410385, |
||||
0.000000000000000000, |
||||
0.125324201622410869, |
||||
0.000000000000000000, |
||||
0.210709715079613419, |
||||
0.000000000000000000, |
||||
0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.210709715079613419, |
||||
0.000000000000000000, |
||||
-0.125324201622410869, |
||||
0.000000000000000000, |
||||
-0.088347125250410385, |
||||
0.000000000000000000, |
||||
-0.067515548043274365, |
||||
0.000000000000000000, |
||||
-0.054032871748808158, |
||||
0.000000000000000000, |
||||
-0.044517689704988733, |
||||
0.000000000000000000, |
||||
-0.037393534868609392, |
||||
0.000000000000000000, |
||||
-0.031827023036304102, |
||||
0.000000000000000000, |
||||
-0.027336628208641155, |
||||
0.000000000000000000, |
||||
-0.023625003856774591, |
||||
0.000000000000000000, |
||||
-0.020498780519188083, |
||||
0.000000000000000000, |
||||
-0.017826854793349920, |
||||
0.000000000000000000, |
||||
-0.015517212970554440, |
||||
0.000000000000000000, |
||||
-0.013503343224246872, |
||||
0.000000000000000000, |
||||
-0.011735907609641917, |
||||
0.000000000000000000, |
||||
-0.010177443901536392, |
||||
0.000000000000000000, |
||||
-0.008798886675905997, |
||||
0.000000000000000000, |
||||
-0.007577220484233372, |
||||
0.000000000000000000, |
||||
-0.006493859804516438, |
||||
0.000000000000000000, |
||||
-0.005533508538632429, |
||||
0.000000000000000000, |
||||
-0.004683343721596901, |
||||
0.000000000000000000, |
||||
-0.003932423233774751, |
||||
0.000000000000000000, |
||||
-0.003271251311125927, |
||||
0.000000000000000000, |
||||
-0.002691457154069645, |
||||
0.000000000000000000, |
||||
-0.002185555845922462, |
||||
0.000000000000000000, |
||||
-0.001746769975247667, |
||||
0.000000000000000000, |
||||
-0.001368896533613985, |
||||
0.000000000000000000, |
||||
-0.001046207887980644, |
||||
0.000000000000000000, |
||||
-0.000773378567767513, |
||||
0.000000000000000000}; |
||||
@ -0,0 +1,21 @@ |
||||
// Following is a 19 term Hilbert FIR filter
|
||||
float hilbert19A[]={ |
||||
0.003006666677728199, |
||||
0.000000000000000000, |
||||
0.017439390086760032, |
||||
0.000000000000000000, |
||||
0.058235158985840196, |
||||
0.000000000000000000, |
||||
0.161905323559397157, |
||||
0.000000000000000000, |
||||
0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.161905323559397157, |
||||
0.000000000000000000, |
||||
-0.058235158985840196, |
||||
0.000000000000000000, |
||||
-0.017439390086760032, |
||||
0.000000000000000000, |
||||
-0.003006666677728199}; |
||||
@ -0,0 +1,253 @@ |
||||
// Following is 251 term Hilbert FIR filter
|
||||
float32_t hilbert251A[]={ |
||||
0.0000003255, |
||||
0.0000000000, |
||||
0.0000030702, |
||||
0.0000000000, |
||||
0.0000089286, |
||||
0.0000000000, |
||||
0.0000183061, |
||||
0.0000000000, |
||||
0.0000316287, |
||||
0.0000000000, |
||||
0.0000493436, |
||||
0.0000000000, |
||||
0.0000719193, |
||||
0.0000000000, |
||||
0.0000998451, |
||||
0.0000000000, |
||||
0.0001336320, |
||||
0.0000000000, |
||||
0.0001738120, |
||||
0.0000000000, |
||||
0.0002209393, |
||||
0.0000000000, |
||||
0.0002755899, |
||||
0.0000000000, |
||||
0.0003383625, |
||||
0.0000000000, |
||||
0.0004098790, |
||||
0.0000000000, |
||||
0.0004907853, |
||||
0.0000000000, |
||||
0.0005817525, |
||||
0.0000000000, |
||||
0.0006834782, |
||||
0.0000000000, |
||||
0.0007966881, |
||||
0.0000000000, |
||||
0.0009221383, |
||||
0.0000000000, |
||||
0.0010606178, |
||||
0.0000000000, |
||||
0.0012129515, |
||||
0.0000000000, |
||||
0.0013800041, |
||||
0.0000000000, |
||||
0.0015626848, |
||||
0.0000000000, |
||||
0.0017619529, |
||||
0.0000000000, |
||||
0.0019788241, |
||||
0.0000000000, |
||||
0.0022143787, |
||||
0.0000000000, |
||||
0.0024697715, |
||||
0.0000000000, |
||||
0.0027462425, |
||||
0.0000000000, |
||||
0.0030451312, |
||||
0.0000000000, |
||||
0.0033678928, |
||||
0.0000000000, |
||||
0.0037161183, |
||||
0.0000000000, |
||||
0.0040915578, |
||||
0.0000000000, |
||||
0.0044961498, |
||||
0.0000000000, |
||||
0.0049320558, |
||||
0.0000000000, |
||||
0.0054017033, |
||||
0.0000000000, |
||||
0.0059078375, |
||||
0.0000000000, |
||||
0.0064535860, |
||||
0.0000000000, |
||||
0.0070425380, |
||||
0.0000000000, |
||||
0.0076788436, |
||||
0.0000000000, |
||||
0.0083673390, |
||||
0.0000000000, |
||||
0.0091137048, |
||||
0.0000000000, |
||||
0.0099246683, |
||||
0.0000000000, |
||||
0.0108082660, |
||||
0.0000000000, |
||||
0.0117741868, |
||||
0.0000000000, |
||||
0.0128342256, |
||||
0.0000000000, |
||||
0.0140028938, |
||||
0.0000000000, |
||||
0.0152982506, |
||||
0.0000000000, |
||||
0.0167430570, |
||||
0.0000000000, |
||||
0.0183664064, |
||||
0.0000000000, |
||||
0.0202060801, |
||||
0.0000000000, |
||||
0.0223120327, |
||||
0.0000000000, |
||||
0.0247516963, |
||||
0.0000000000, |
||||
0.0276183140, |
||||
0.0000000000, |
||||
0.0310445375, |
||||
0.0000000000, |
||||
0.0352256211, |
||||
0.0000000000, |
||||
0.0404611696, |
||||
0.0000000000, |
||||
0.0472354231, |
||||
0.0000000000, |
||||
0.0563851215, |
||||
0.0000000000, |
||||
0.0694911881, |
||||
0.0000000000, |
||||
0.0899418673, |
||||
0.0000000000, |
||||
0.1265473875, |
||||
0.0000000000, |
||||
0.2116132716, |
||||
0.0000000000, |
||||
0.6358933477, |
||||
0.0000000000, |
||||
-0.6358933478, |
||||
0.0000000000, |
||||
-0.2116132717, |
||||
0.0000000000, |
||||
-0.1265473876, |
||||
0.0000000000, |
||||
-0.0899418674, |
||||
0.0000000000, |
||||
-0.0694911882, |
||||
0.0000000000, |
||||
-0.0563851216, |
||||
0.0000000000, |
||||
-0.0472354232, |
||||
0.0000000000, |
||||
-0.0404611697, |
||||
0.0000000000, |
||||
-0.0352256212, |
||||
0.0000000000, |
||||
-0.0310445376, |
||||
0.0000000000, |
||||
-0.0276183141, |
||||
0.0000000000, |
||||
-0.0247516964, |
||||
0.0000000000, |
||||
-0.0223120328, |
||||
0.0000000000, |
||||
-0.0202060802, |
||||
0.0000000000, |
||||
-0.0183664065, |
||||
0.0000000000, |
||||
-0.0167430571, |
||||
0.0000000000, |
||||
-0.0152982507, |
||||
0.0000000000, |
||||
-0.0140028939, |
||||
0.0000000000, |
||||
-0.0128342257, |
||||
0.0000000000, |
||||
-0.0117741869, |
||||
0.0000000000, |
||||
-0.0108082661, |
||||
0.0000000000, |
||||
-0.0099246684, |
||||
0.0000000000, |
||||
-0.0091137049, |
||||
0.0000000000, |
||||
-0.0083673391, |
||||
0.0000000000, |
||||
-0.0076788437, |
||||
0.0000000000, |
||||
-0.0070425381, |
||||
0.0000000000, |
||||
-0.0064535861, |
||||
0.0000000000, |
||||
-0.0059078376, |
||||
0.0000000000, |
||||
-0.0054017034, |
||||
0.0000000000, |
||||
-0.0049320559, |
||||
0.0000000000, |
||||
-0.0044961499, |
||||
0.0000000000, |
||||
-0.0040915579, |
||||
0.0000000000, |
||||
-0.0037161184, |
||||
0.0000000000, |
||||
-0.0033678929, |
||||
0.0000000000, |
||||
-0.0030451313, |
||||
0.0000000000, |
||||
-0.0027462426, |
||||
0.0000000000, |
||||
-0.0024697716, |
||||
0.0000000000, |
||||
-0.0022143788, |
||||
0.0000000000, |
||||
-0.0019788242, |
||||
0.0000000000, |
||||
-0.0017619530, |
||||
0.0000000000, |
||||
-0.0015626849, |
||||
0.0000000000, |
||||
-0.0013800042, |
||||
0.0000000000, |
||||
-0.0012129516, |
||||
0.0000000000, |
||||
-0.0010606179, |
||||
0.0000000000, |
||||
-0.0009221384, |
||||
0.0000000000, |
||||
-0.0007966882, |
||||
0.0000000000, |
||||
-0.0006834783, |
||||
0.0000000000, |
||||
-0.0005817526, |
||||
0.0000000000, |
||||
-0.0004907854, |
||||
0.0000000000, |
||||
-0.0004098791, |
||||
0.0000000000, |
||||
-0.0003383626, |
||||
0.0000000000, |
||||
-0.0002755900, |
||||
0.0000000000, |
||||
-0.0002209394, |
||||
0.0000000000, |
||||
-0.0001738121, |
||||
0.0000000000, |
||||
-0.0001336321, |
||||
0.0000000000, |
||||
-0.0000998452, |
||||
0.0000000000, |
||||
-0.0000719194, |
||||
0.0000000000, |
||||
-0.0000493437, |
||||
0.0000000000, |
||||
-0.0000316288, |
||||
0.0000000000, |
||||
-0.0000183062, |
||||
0.0000000000, |
||||
-0.0000089287, |
||||
0.0000000000, |
||||
-0.0000030703, |
||||
0.0000000000, |
||||
-0.0000003256}; |
||||
@ -0,0 +1,81 @@ |
||||
/* TestEqualizer1.ino Bob Larkin 10 May 2020
|
||||
* This is a test of the Filter Equalizer for Teensy Audio. |
||||
* This version is for the Chip Audette _F32 Library. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
AudioInputI2S i2s1; // Start interrupts
|
||||
AudioSynthWaveformSine_F32 sine1; // Test signal
|
||||
AudioFilterEqualizer_F32 equalize1; |
||||
AudioRecordQueue_F32 queue1; // The LSB output
|
||||
AudioConnection_F32 patchCord1(sine1, 0, equalize1, 0); |
||||
AudioConnection_F32 patchCord2(equalize1, 0, queue1, 0); |
||||
|
||||
//nBands = 10 This octave band equalizer is set strangely to demonstrate the Equalizer
|
||||
float fBand[] = { 40.0, 80.0, 160.0, 320.0, 640.0, 1280.0, 2560.0, 5120.0, 10240.0, 22058.5}; |
||||
float dbBand[] = {10.0, 2.0, -2.0, -5.0, -2.0, -4.0, -20.0, 6.0, 10.0, -100}; |
||||
float equalizeCoeffs[249]; |
||||
float dt1[128]; |
||||
float *pq1, *pd1; |
||||
int i, k; |
||||
float32_t dBResponse[500]; // Show lots of detail
|
||||
|
||||
void setup(void) { |
||||
AudioMemory(5);
|
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
Serial.println("*** Test Audio Equalizer ***"); |
||||
sine1.frequency(1000.0f); |
||||
|
||||
// Initialize the equalizer with 10 bands, 199 FIR coefficients and -65 dB sidelobes
|
||||
uint16_t eq = equalize1.equalizerNew(10, &fBand[0], &dbBand[0], 199, &equalizeCoeffs[0], 65); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("Equalizer failed: Invalid number of frequency bands."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("Equalizer failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("Equalizer failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("Equalizer initialized successfully."); |
||||
|
||||
// Get frequency response in dB for 500 points, uniformly spaced from 0 to 21058 Hz
|
||||
equalize1.getResponse(500, dBResponse); |
||||
Serial.println("Freq Hz, Response dB"); |
||||
for(int m=0; m<500; m++) { // Print the response in Hz, dB, suitable for a spread sheet
|
||||
Serial.print((float32_t)m * 22058.5f / 500.0f); |
||||
Serial.print(","); |
||||
Serial.println(dBResponse[m]); |
||||
} |
||||
i = -10; k=0; |
||||
} |
||||
|
||||
void loop(void) { |
||||
if(i<0) i++; // Get past startup
|
||||
if(i==0) queue1.begin(); |
||||
|
||||
// Collect 128 samples and output to Serial
|
||||
// This "if" will be active for i == 0
|
||||
if (queue1.available() >= 1 && i >= 0) { |
||||
pq1 = queue1.readBuffer(); |
||||
pd1 = &dt1[0]; |
||||
for(k=0; k<128; k++) { |
||||
*pd1++ = *pq1++; |
||||
} |
||||
i=1; // Only collect 1 block
|
||||
Serial.print("Maximum F32 memory usage at loop:"); |
||||
Serial.println( AudioMemoryUsageMax_F32() ); |
||||
queue1.freeBuffer(); |
||||
queue1.end(); // No more data to queue1
|
||||
} |
||||
if(i == 1) {
|
||||
// Uncomment the next 3 lines to printout a sample of the sine wave.
|
||||
/*Serial.println("128 Samples: ");
|
||||
for (k=0; k<128; k++) { |
||||
Serial.println (dt1[k],7); */ |
||||
} |
||||
i = 2; |
||||
delay(2); |
||||
} |
||||
@ -0,0 +1,123 @@ |
||||
// Following is 121 term Hilbert FIR filter
|
||||
float32_t hilbert121A[121] = { |
||||
0.000000000000000000, |
||||
0.000773378567767513, |
||||
0.000000000000000000, |
||||
0.001046207887980644, |
||||
0.000000000000000000, |
||||
0.001368896533613985, |
||||
0.000000000000000000, |
||||
0.001746769975247667, |
||||
0.000000000000000000, |
||||
0.002185555845922462, |
||||
0.000000000000000000, |
||||
0.002691457154069645, |
||||
0.000000000000000000, |
||||
0.003271251311125927, |
||||
0.000000000000000000, |
||||
0.003932423233774751, |
||||
0.000000000000000000, |
||||
0.004683343721596901, |
||||
0.000000000000000000, |
||||
0.005533508538632429, |
||||
0.000000000000000000, |
||||
0.006493859804516438, |
||||
0.000000000000000000, |
||||
0.007577220484233372, |
||||
0.000000000000000000, |
||||
0.008798886675905997, |
||||
0.000000000000000000, |
||||
0.010177443901536392, |
||||
0.000000000000000000, |
||||
0.011735907609641917, |
||||
0.000000000000000000, |
||||
0.013503343224246872, |
||||
0.000000000000000000, |
||||
0.015517212970554440, |
||||
0.000000000000000000, |
||||
0.017826854793349920, |
||||
0.000000000000000000, |
||||
0.020498780519188083, |
||||
0.000000000000000000, |
||||
0.023625003856774591, |
||||
0.000000000000000000, |
||||
0.027336628208641155, |
||||
0.000000000000000000, |
||||
0.031827023036304102, |
||||
0.000000000000000000, |
||||
0.037393534868609392, |
||||
0.000000000000000000, |
||||
0.044517689704988733, |
||||
0.000000000000000000, |
||||
0.054032871748808158, |
||||
0.000000000000000000, |
||||
0.067515548043274365, |
||||
0.000000000000000000, |
||||
0.088347125250410385, |
||||
0.000000000000000000, |
||||
0.125324201622410869, |
||||
0.000000000000000000, |
||||
0.210709715079613419, |
||||
0.000000000000000000, |
||||
0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.634897508268964295, |
||||
0.000000000000000000, |
||||
-0.210709715079613419, |
||||
0.000000000000000000, |
||||
-0.125324201622410869, |
||||
0.000000000000000000, |
||||
-0.088347125250410385, |
||||
0.000000000000000000, |
||||
-0.067515548043274365, |
||||
0.000000000000000000, |
||||
-0.054032871748808158, |
||||
0.000000000000000000, |
||||
-0.044517689704988733, |
||||
0.000000000000000000, |
||||
-0.037393534868609392, |
||||
0.000000000000000000, |
||||
-0.031827023036304102, |
||||
0.000000000000000000, |
||||
-0.027336628208641155, |
||||
0.000000000000000000, |
||||
-0.023625003856774591, |
||||
0.000000000000000000, |
||||
-0.020498780519188083, |
||||
0.000000000000000000, |
||||
-0.017826854793349920, |
||||
0.000000000000000000, |
||||
-0.015517212970554440, |
||||
0.000000000000000000, |
||||
-0.013503343224246872, |
||||
0.000000000000000000, |
||||
-0.011735907609641917, |
||||
0.000000000000000000, |
||||
-0.010177443901536392, |
||||
0.000000000000000000, |
||||
-0.008798886675905997, |
||||
0.000000000000000000, |
||||
-0.007577220484233372, |
||||
0.000000000000000000, |
||||
-0.006493859804516438, |
||||
0.000000000000000000, |
||||
-0.005533508538632429, |
||||
0.000000000000000000, |
||||
-0.004683343721596901, |
||||
0.000000000000000000, |
||||
-0.003932423233774751, |
||||
0.000000000000000000, |
||||
-0.003271251311125927, |
||||
0.000000000000000000, |
||||
-0.002691457154069645, |
||||
0.000000000000000000, |
||||
-0.002185555845922462, |
||||
0.000000000000000000, |
||||
-0.001746769975247667, |
||||
0.000000000000000000, |
||||
-0.001368896533613985, |
||||
0.000000000000000000, |
||||
-0.001046207887980644, |
||||
0.000000000000000000, |
||||
-0.000773378567767513, |
||||
0.000000000000000000}; |
||||
@ -0,0 +1,21 @@ |
||||
// Following is a 19 term Hilbert FIR filter
|
||||
float hilbert19A[]={ |
||||
0.003006666677728199, |
||||
0.000000000000000000, |
||||
0.017439390086760032, |
||||
0.000000000000000000, |
||||
0.058235158985840196, |
||||
0.000000000000000000, |
||||
0.161905323559397157, |
||||
0.000000000000000000, |
||||
0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.617838316334015092, |
||||
0.000000000000000000, |
||||
-0.161905323559397157, |
||||
0.000000000000000000, |
||||
-0.058235158985840196, |
||||
0.000000000000000000, |
||||
-0.017439390086760032, |
||||
0.000000000000000000, |
||||
-0.003006666677728199}; |
||||
@ -0,0 +1,253 @@ |
||||
// Following is 251 term Hilbert FIR filter
|
||||
float32_t hilbert251A[]={ |
||||
0.0000003255, |
||||
0.0000000000, |
||||
0.0000030702, |
||||
0.0000000000, |
||||
0.0000089286, |
||||
0.0000000000, |
||||
0.0000183061, |
||||
0.0000000000, |
||||
0.0000316287, |
||||
0.0000000000, |
||||
0.0000493436, |
||||
0.0000000000, |
||||
0.0000719193, |
||||
0.0000000000, |
||||
0.0000998451, |
||||
0.0000000000, |
||||
0.0001336320, |
||||
0.0000000000, |
||||
0.0001738120, |
||||
0.0000000000, |
||||
0.0002209393, |
||||
0.0000000000, |
||||
0.0002755899, |
||||
0.0000000000, |
||||
0.0003383625, |
||||
0.0000000000, |
||||
0.0004098790, |
||||
0.0000000000, |
||||
0.0004907853, |
||||
0.0000000000, |
||||
0.0005817525, |
||||
0.0000000000, |
||||
0.0006834782, |
||||
0.0000000000, |
||||
0.0007966881, |
||||
0.0000000000, |
||||
0.0009221383, |
||||
0.0000000000, |
||||
0.0010606178, |
||||
0.0000000000, |
||||
0.0012129515, |
||||
0.0000000000, |
||||
0.0013800041, |
||||
0.0000000000, |
||||
0.0015626848, |
||||
0.0000000000, |
||||
0.0017619529, |
||||
0.0000000000, |
||||
0.0019788241, |
||||
0.0000000000, |
||||
0.0022143787, |
||||
0.0000000000, |
||||
0.0024697715, |
||||
0.0000000000, |
||||
0.0027462425, |
||||
0.0000000000, |
||||
0.0030451312, |
||||
0.0000000000, |
||||
0.0033678928, |
||||
0.0000000000, |
||||
0.0037161183, |
||||
0.0000000000, |
||||
0.0040915578, |
||||
0.0000000000, |
||||
0.0044961498, |
||||
0.0000000000, |
||||
0.0049320558, |
||||
0.0000000000, |
||||
0.0054017033, |
||||
0.0000000000, |
||||
0.0059078375, |
||||
0.0000000000, |
||||
0.0064535860, |
||||
0.0000000000, |
||||
0.0070425380, |
||||
0.0000000000, |
||||
0.0076788436, |
||||
0.0000000000, |
||||
0.0083673390, |
||||
0.0000000000, |
||||
0.0091137048, |
||||
0.0000000000, |
||||
0.0099246683, |
||||
0.0000000000, |
||||
0.0108082660, |
||||
0.0000000000, |
||||
0.0117741868, |
||||
0.0000000000, |
||||
0.0128342256, |
||||
0.0000000000, |
||||
0.0140028938, |
||||
0.0000000000, |
||||
0.0152982506, |
||||
0.0000000000, |
||||
0.0167430570, |
||||
0.0000000000, |
||||
0.0183664064, |
||||
0.0000000000, |
||||
0.0202060801, |
||||
0.0000000000, |
||||
0.0223120327, |
||||
0.0000000000, |
||||
0.0247516963, |
||||
0.0000000000, |
||||
0.0276183140, |
||||
0.0000000000, |
||||
0.0310445375, |
||||
0.0000000000, |
||||
0.0352256211, |
||||
0.0000000000, |
||||
0.0404611696, |
||||
0.0000000000, |
||||
0.0472354231, |
||||
0.0000000000, |
||||
0.0563851215, |
||||
0.0000000000, |
||||
0.0694911881, |
||||
0.0000000000, |
||||
0.0899418673, |
||||
0.0000000000, |
||||
0.1265473875, |
||||
0.0000000000, |
||||
0.2116132716, |
||||
0.0000000000, |
||||
0.6358933477, |
||||
0.0000000000, |
||||
-0.6358933478, |
||||
0.0000000000, |
||||
-0.2116132717, |
||||
0.0000000000, |
||||
-0.1265473876, |
||||
0.0000000000, |
||||
-0.0899418674, |
||||
0.0000000000, |
||||
-0.0694911882, |
||||
0.0000000000, |
||||
-0.0563851216, |
||||
0.0000000000, |
||||
-0.0472354232, |
||||
0.0000000000, |
||||
-0.0404611697, |
||||
0.0000000000, |
||||
-0.0352256212, |
||||
0.0000000000, |
||||
-0.0310445376, |
||||
0.0000000000, |
||||
-0.0276183141, |
||||
0.0000000000, |
||||
-0.0247516964, |
||||
0.0000000000, |
||||
-0.0223120328, |
||||
0.0000000000, |
||||
-0.0202060802, |
||||
0.0000000000, |
||||
-0.0183664065, |
||||
0.0000000000, |
||||
-0.0167430571, |
||||
0.0000000000, |
||||
-0.0152982507, |
||||
0.0000000000, |
||||
-0.0140028939, |
||||
0.0000000000, |
||||
-0.0128342257, |
||||
0.0000000000, |
||||
-0.0117741869, |
||||
0.0000000000, |
||||
-0.0108082661, |
||||
0.0000000000, |
||||
-0.0099246684, |
||||
0.0000000000, |
||||
-0.0091137049, |
||||
0.0000000000, |
||||
-0.0083673391, |
||||
0.0000000000, |
||||
-0.0076788437, |
||||
0.0000000000, |
||||
-0.0070425381, |
||||
0.0000000000, |
||||
-0.0064535861, |
||||
0.0000000000, |
||||
-0.0059078376, |
||||
0.0000000000, |
||||
-0.0054017034, |
||||
0.0000000000, |
||||
-0.0049320559, |
||||
0.0000000000, |
||||
-0.0044961499, |
||||
0.0000000000, |
||||
-0.0040915579, |
||||
0.0000000000, |
||||
-0.0037161184, |
||||
0.0000000000, |
||||
-0.0033678929, |
||||
0.0000000000, |
||||
-0.0030451313, |
||||
0.0000000000, |
||||
-0.0027462426, |
||||
0.0000000000, |
||||
-0.0024697716, |
||||
0.0000000000, |
||||
-0.0022143788, |
||||
0.0000000000, |
||||
-0.0019788242, |
||||
0.0000000000, |
||||
-0.0017619530, |
||||
0.0000000000, |
||||
-0.0015626849, |
||||
0.0000000000, |
||||
-0.0013800042, |
||||
0.0000000000, |
||||
-0.0012129516, |
||||
0.0000000000, |
||||
-0.0010606179, |
||||
0.0000000000, |
||||
-0.0009221384, |
||||
0.0000000000, |
||||
-0.0007966882, |
||||
0.0000000000, |
||||
-0.0006834783, |
||||
0.0000000000, |
||||
-0.0005817526, |
||||
0.0000000000, |
||||
-0.0004907854, |
||||
0.0000000000, |
||||
-0.0004098791, |
||||
0.0000000000, |
||||
-0.0003383626, |
||||
0.0000000000, |
||||
-0.0002755900, |
||||
0.0000000000, |
||||
-0.0002209394, |
||||
0.0000000000, |
||||
-0.0001738121, |
||||
0.0000000000, |
||||
-0.0001336321, |
||||
0.0000000000, |
||||
-0.0000998452, |
||||
0.0000000000, |
||||
-0.0000719194, |
||||
0.0000000000, |
||||
-0.0000493437, |
||||
0.0000000000, |
||||
-0.0000316288, |
||||
0.0000000000, |
||||
-0.0000183062, |
||||
0.0000000000, |
||||
-0.0000089287, |
||||
0.0000000000, |
||||
-0.0000030703, |
||||
0.0000000000, |
||||
-0.0000003256}; |
||||
@ -0,0 +1,105 @@ |
||||
/* TestEqualizer1Audio.ino Bob Larkin 18 May 2020
|
||||
* This is a test of the Filter Equalizer using the Audette OpenAudio_ArduinoLibrary. |
||||
* Runs two different equalizers, switching every 5 seconds to demonstrate |
||||
* dynamic filter changing. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
#include <Audio.h> |
||||
|
||||
// Signals from ADC to Equalizer to DAC using Teensy Audio Adaptor
|
||||
AudioInputI2S i2sIn; |
||||
AudioConvert_I16toF32 convertItoF1; |
||||
AudioConvert_I16toF32 convertItoF2; |
||||
AudioFilterEqualizer_F32 equalize1; |
||||
AudioFilterEqualizer_F32 equalize2; |
||||
AudioConvert_F32toI16 convertFtoI1; |
||||
AudioConvert_F32toI16 convertFtoI2; |
||||
AudioOutputI2S i2sOut; |
||||
AudioConnection patchCord1(i2sIn, 0, convertItoF1, 0);
|
||||
AudioConnection patchCord3(i2sIn, 1, convertItoF2, 0); |
||||
AudioConnection_F32 patchCord5(convertItoF1, 0, equalize1, 0); |
||||
AudioConnection_F32 patchCord7(convertItoF2, 0, equalize2, 0); |
||||
AudioConnection_F32 patchCord9(equalize1, 0, convertFtoI1, 0); |
||||
AudioConnection_F32 patchCordB(equalize2, 0, convertFtoI2, 0); |
||||
AudioConnection patchCordD(convertFtoI1, 0, i2sOut, 0);
|
||||
AudioConnection patchCordF(convertFtoI2, 0, i2sOut, 1); |
||||
AudioControlSGTL5000 codec; |
||||
|
||||
// Some sort of octave band equalizer as a one alternative, 10 bands
|
||||
float32_t fBand1[] = { 40.0, 80.0, 160.0, 320.0, 640.0, 1280.0, 2560.0, 5120.0, 10240.0, 22058.5}; |
||||
float32_t dbBand1[] = {10.0, 2.0, -2.0, -5.0, -2.0, -4.0, -10.0, 6.0, 10.0, -100}; |
||||
|
||||
// To contrast, put a strange bandpass filter as an alternative, 5 bands
|
||||
float32_t fBand2[] = { 300.0, 500.0, 800.0, 1000.0, 22058.5}; |
||||
float32_t dbBand2[] = {-60.0, 0.0, -20.0, 0.0, -60.0}; |
||||
|
||||
float32_t equalizeCoeffs[200]; |
||||
int16_t k = 0; |
||||
|
||||
void setup(void) { |
||||
AudioMemory(5);
|
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
Serial.println("*** Test Audio Equalizer ***"); |
||||
codec.enable(); |
||||
codec.inputSelect(AUDIO_INPUT_LINEIN);
|
||||
codec.adcHighPassFilterDisable(); // necessary to suppress noise
|
||||
codec.lineInLevel(2); |
||||
codec.volume(0.8); |
||||
|
||||
// Initialize the equalizer with 10 bands, fBand1[] 199 FIR coefficients
|
||||
// -65 dB sidelobes, 16384 Max coefficient value
|
||||
uint16_t eq = equalize1.equalizerNew(10, &fBand1[0], &dbBand1[0], 30, &equalizeCoeffs[0], 60.0); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("Equalizer failed: Invalid number of frequency bands."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("Equalizer failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("Equalizer failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("Equalizer initialized successfully."); |
||||
|
||||
eq = equalize2.equalizerNew(10, &fBand1[0], &dbBand1[0], 30, &equalizeCoeffs[0], 60.0); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("Equalizer failed: Invalid number of frequency bands."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("Equalizer failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("Equalizer failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("Equalizer initialized successfully."); |
||||
// for (k=0; k<200; k++) Serial.println(equalizeCoeffs[k]);
|
||||
} |
||||
|
||||
void loop(void) { |
||||
#if 0 // Change between two filters every 10 seconds.
|
||||
|
||||
// To run with just the 10-band equalizer, comment out the entire loop with "/* ... */"
|
||||
|
||||
if(k == 0) { |
||||
k = 1; |
||||
equalize1.equalizerNew(10, &fBand1[0], &dbBand1[0], 200, &equalizeCoeffs[0], 65.0); |
||||
equalize2.equalizerNew(10, &fBand1[0], &dbBand1[0], 200, &equalizeCoeffs[0], 65.0); |
||||
} |
||||
else { // Swap back and forth
|
||||
k = 0; |
||||
equalize1.equalizerNew(5, &fBand2[0], &dbBand2[0], 100, &equalizeCoeffs[0], 40.0); |
||||
equalize2.equalizerNew(5, &fBand2[0], &dbBand2[0], 100, &equalizeCoeffs[0], 40.0); |
||||
} |
||||
#endif |
||||
delay(10000); // Interrupts will keep the audio going
|
||||
Serial.print("Proc = "); |
||||
Serial.print(AudioProcessorUsage()); |
||||
Serial.print(" ("); |
||||
Serial.print(AudioProcessorUsageMax()); |
||||
Serial.print("), Mem = "); |
||||
Serial.print(AudioMemoryUsage()); |
||||
Serial.print(" ("); |
||||
Serial.print(AudioMemoryUsageMax()); |
||||
Serial.println(")"); |
||||
AudioProcessorUsageMaxReset(); |
||||
AudioMemoryUsageMaxReset(); |
||||
} |
||||
@ -0,0 +1,88 @@ |
||||
/* TestEqualizer3General.ino Bob Larkin 22 May 2020
|
||||
* This is a test of the Filter Equalizer for Teensy Audio. |
||||
* This version is for the Chip Audette _F32 Library. |
||||
* See TestEqualizer2.ino for the int16 version. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
AudioInputI2S i2s1; |
||||
AudioSynthWaveformSine_F32 sine1; // Test signal
|
||||
AudioFilterFIRGeneral_F32 firg1; |
||||
AudioRecordQueue_F32 queue1; // The LSB output
|
||||
AudioConnection_F32 patchCord1(sine1, 0, firg1, 0); |
||||
AudioConnection_F32 patchCord2(firg1, 0, queue1, 0); |
||||
|
||||
float dbA[51]; |
||||
float equalizeCoeffs[51]; |
||||
float dt1[128]; |
||||
float *pq1, *pd1; |
||||
int i, k; |
||||
float32_t dBResponse[500]; // Show lots of detail
|
||||
|
||||
void setup(void) { |
||||
AudioMemory(5);
|
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
Serial.println("*** Test FIR General routine for F32 ***"); |
||||
sine1.frequency(1000.0f); |
||||
|
||||
// This simple case corresponds to Burris and Parks, "Digital Filter Design."
|
||||
// Example 3.1when sidelobes are set to 0.0.
|
||||
for (i=0; i<6; i++) dbA[i] = 0.0f; |
||||
for (i=6; i<11; i++) dbA[i] = -120.0f; |
||||
|
||||
// Initialize the FIR with 21 points, 21 FIR coefficients and -0 dB sidelobes
|
||||
uint16_t eq = firg1.FIRGeneralNew(&dbA[0], 21, &equalizeCoeffs[0], 0.0); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("FIR General failed: Invalid number of frequency points."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("FIR General failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("FIR General failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("FIR General initialized successfully."); |
||||
|
||||
Serial.println("FIR Coeffs"); |
||||
for(i=0; i<21; i++) Serial.println(equalizeCoeffs[i], 7); |
||||
Serial.println(""); |
||||
|
||||
// Get frequency response in dB for 200 points, uniformly spaced from 0 to 21058 Hz
|
||||
firg1.getResponse(200, dBResponse); |
||||
Serial.println("Freq Hz, Response dB"); |
||||
for(int m=0; m<200; m++) { // Print the response in Hz, dB, suitable for a spread sheet
|
||||
Serial.print((float32_t)m * 22058.5f / 200.0f); |
||||
Serial.print(","); |
||||
Serial.println(dBResponse[m]); |
||||
} |
||||
|
||||
i = -10; k=0; |
||||
} |
||||
|
||||
void loop(void) { |
||||
if(i<0) i++; // Get past startup
|
||||
if(i==0) queue1.begin(); |
||||
|
||||
// Collect 128 samples and output to Serial
|
||||
// This "if" will be active for i == 0
|
||||
if (queue1.available() >= 1 && i >= 0) { |
||||
pq1 = queue1.readBuffer(); |
||||
pd1 = &dt1[0]; |
||||
for(k=0; k<128; k++) { |
||||
*pd1++ = *pq1++; |
||||
} |
||||
i=1; // Only collect 1 block
|
||||
queue1.freeBuffer(); |
||||
queue1.end(); // No more data to queue1
|
||||
} |
||||
if(i == 1) {
|
||||
// Uncomment the next 3 lines to printout a sample of the sine wave.
|
||||
Serial.println("128 Samples: "); |
||||
for (k=0; k<128; k++) |
||||
Serial.println (dt1[k],7);
|
||||
i = 2; |
||||
} |
||||
delay(2); |
||||
} |
||||
|
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||||
/* TestFIRGeneralLarge4.ino Bob Larkin 24 May 2020
|
||||
* Test the generation of FIR filters and obtaining their |
||||
* responses. See TestFIRGeneralLarge5.ino to run the filters. |
||||
* This is a test of the AudioFilterFIR_F32 for Teensy Audio. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
AudioInputI2S i2s1; // Creates timing
|
||||
AudioFilterFIRGeneral_F32 firg1; |
||||
|
||||
float32_t dbA[2000]; |
||||
float32_t workSpace[4128]; |
||||
float32_t equalizeCoeffs[4000]; |
||||
int i, k; |
||||
float32_t dBResponse[5000]; // Extreme size, can show lots of detail!
|
||||
|
||||
// Following is based on Rabiner, McGonegal and Paul, FWFIR test case, but has
|
||||
// had the windowing effect divided out to show their basic filter response.
|
||||
float32_t hpCoeff[55]={ |
||||
-0.003643388f, -0.007195844f, 0.012732424f, -0.007796006f, -0.004276372f, |
||||
0.013760401f, -0.012262941f, 0.000000304f, 0.013553423f, -0.016818445f, |
||||
0.005786355f, 0.011693321f, -0.021220577f, 0.013364335f, 0.007566100f, |
||||
-0.025227439f, 0.023410856f, -0.000000305f, -0.028612852f, 0.037841329f, |
||||
-0.014052131f, -0.031182658f, 0.063661835f, -0.046774701f, -0.032787389f, |
||||
0.151364865f, -0.257518316f, 0.300000700f, -0.257518316f, 0.151364865f, |
||||
-0.032787389f, -0.046774701f, 0.063661835f, -0.031182658f, -0.014052131f, |
||||
0.037841329f, -0.028612852f, -0.000000305f, 0.023410856f, -0.025227439f, |
||||
0.007566100f, 0.013364335f, -0.021220577f, 0.011693321f, 0.005786355f, |
||||
-0.016818445f, 0.013553423f, 0.000000304f, -0.012262941f, 0.013760401f, |
||||
-0.004276372f, -0.007796006f, 0.012732424f, -0.007195844f, -0.003643388}; |
||||
|
||||
void setup(void) { |
||||
uint16_t nFIR = 4; |
||||
float32_t sidelobes = 45; |
||||
AudioMemory(5); |
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
Serial.println("*** Test Response of FIR General routine for F32 ***"); |
||||
|
||||
// Un-comment one of the following, or use the direct Coefficient load that follows
|
||||
|
||||
// Test case for odd nFIR from Burris and Parks Example 3.1
|
||||
nFIR = 21; sidelobes = 0.0; // No window
|
||||
dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=0.0; dbA[5]=0.0; |
||||
dbA[6]=-140.0; dbA[7]=-140.0; dbA[8]=-140.0; dbA[9]=-140.0; dbA[10]=-140.0; |
||||
|
||||
// Test case for even nFIR from Burris and Parks Example 3.2
|
||||
// nFIR = 20; sidelobes = 0.0;
|
||||
// dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=-120.0; dbA[5]=-120.0;
|
||||
// dbA[6]=-120.0; dbA[7]=-120.0; dbA[8]=-120.0; dbA[9]=-120.0;
|
||||
|
||||
// Test case for odd nFIR from Burris and Parks Example 3.1 with Kaiser Window
|
||||
// nFIR = 21; sidelobes = 45.0f;
|
||||
// dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=0.0; dbA[5]=0.0;
|
||||
// dbA[6]=-140.0; dbA[7]=-140.0; dbA[8]=-140.0; dbA[9]=-140.0; dbA[10]=-140.0f;
|
||||
|
||||
// Test case for even nFIR from Burris and Parks Example 3.2
|
||||
// nFIR = 20; sidelobes = 45.0f;
|
||||
// dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=0.0; dbA[5]=-120.0;
|
||||
// dbA[6]=-120.0; dbA[7]=-120.0; dbA[8]=-120.0; dbA[9]=-120.0;
|
||||
|
||||
// Rabiner, McGonegal and Paul, FWFIR test case, HP Kaiser SL=60, fco=0.35
|
||||
// nFIR = 55; sidelobes = 60.0;
|
||||
// for(i=0; i<28; i++) {
|
||||
// if (i<20) dbA[i] = -120.0f;
|
||||
// else dbA[i] = 0.0f;
|
||||
// }
|
||||
|
||||
// The next 3 examples are tests of extremely large FIR arrays and only for Teensy 4
|
||||
// Generate a response curve for our "octave" equalizer, tried with 30 dB sidelobes
|
||||
// nFIR = 3999; sidelobes = 30.0f;
|
||||
// dbA[0] = 12.0f; // Avoid log(0)
|
||||
// for(i=1; i<2000; i++)
|
||||
// dbA[i] = 12.0f*cosf(10.0f*log10f(11.02925f*(float32_t)i));
|
||||
|
||||
// Generate a slowly changing response, up 12 dB at bass and treble and -12 dB
|
||||
// for mid-frequencies. Scaled to "octaves" by the form cos(log(f))
|
||||
// nFIR = 3999; sidelobes = 30.0f;
|
||||
// dbA[0] = 12.0f;
|
||||
// dbA[1] = 12.0f;
|
||||
// for(i=2; i<2000; i++)
|
||||
// dbA[i] = -12.0f*cosf(2.2f*log10f(11.02925f*(float32_t)i));
|
||||
|
||||
// Narrow band "CW" filter, tried with 60 dB sidelobes (60 to 80 or 70 to 72
|
||||
// nFIR = 3999; sidelobes = 60;
|
||||
// for(i=0; i<2000; i++) {
|
||||
// if (i<70 || i>72) dbA[i] = -120.0f;
|
||||
// else dbA[i] = 0.0f;
|
||||
// }
|
||||
|
||||
// Initialize the FIR filter design and run
|
||||
uint16_t eq = firg1.FIRGeneralNew(&dbA[0], nFIR, &equalizeCoeffs[0], sidelobes, &workSpace[0]); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("FIR General failed: Invalid number of frequency points."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("FIR General failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("FIR General failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("FIR General initialized successfully."); |
||||
|
||||
// Load (if not commented) non-windowed HP filter coefficients, test case:
|
||||
// firg1.LoadCoeffs(55, hpCoeff, workSpace); // Direct load of FIR coefficients, already calculated
|
||||
|
||||
Serial.println("FIR Coeffs"); |
||||
for(i=0; i<nFIR; i++) |
||||
Serial.println(equalizeCoeffs[i], 7); |
||||
Serial.println(""); |
||||
|
||||
// Get frequency response in dB for 5000 points, uniformly spaced from 0 to 22058 Hz
|
||||
// This is chosen at 5000 for test. Normally, a smaller value,
|
||||
// perhaps 500, would be adequate.
|
||||
firg1.getResponse(5000, dBResponse); |
||||
Serial.println("Freq Hz, Response dB"); |
||||
for(int m=0; m<5000; m++) { // Print the response in Hz, dB, suitable for a spread sheet
|
||||
// Uncomment/comment next 3 for print/no print
|
||||
Serial.print((float32_t)m * 22058.5f / 5000.0f); |
||||
Serial.print(","); |
||||
Serial.println(dBResponse[m]); |
||||
} |
||||
} |
||||
|
||||
void loop(void) { |
||||
}
|
||||
@ -0,0 +1,82 @@ |
||||
/* TestFIRGeneralLarge5.ino Bob Larkin 24 May 2020
|
||||
* This is a test of the FIRGeneralLarge block for Teensy Audio, |
||||
* that has two new elements: |
||||
* * The FIR filter is specified by an arbitrary frequency response |
||||
* * The number of FIR taps is not restricted in number or in even/odd. |
||||
* |
||||
* This test program passes a signal from the SGTL5000 CODEC through |
||||
* the FIR filter and back out through the CODEC. |
||||
* This version is for the Chip Audette _F32 Library. A separate version |
||||
* with adjustments, is for the Teensy Audio Library with I16 data types |
||||
* called AudioFilterFIRGeneral_I16. |
||||
*/ |
||||
|
||||
#include "Audio.h" |
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "DSP_TeensyAudio_F32.h" |
||||
|
||||
// Input as I16 and convert to be Teensy 4 compatible
|
||||
AudioInputI2S i2sIn; |
||||
AudioConvert_I16toF32 convertItoF1; |
||||
AudioFilterFIRGeneral_F32 firg1; |
||||
AudioConvert_F32toI16 convertFtoI1; |
||||
AudioOutputI2S i2sOut; |
||||
AudioConnection patchCord1(i2sIn, 0, convertItoF1, 0);
|
||||
AudioConnection_F32 patchCord5(convertItoF1, 0, firg1, 0); |
||||
AudioConnection_F32 patchCord9(firg1, 0, convertFtoI1, 0); |
||||
AudioConnection patchCordD(convertFtoI1, 0, i2sOut, 0);
|
||||
AudioControlSGTL5000 codec; |
||||
|
||||
float32_t dbA[2000]; |
||||
float32_t workSpace[4128]; |
||||
float32_t equalizeCoeffs[4000]; |
||||
int i; |
||||
|
||||
void setup(void) { |
||||
uint16_t nFIR = 4; |
||||
float32_t sidelobes = 45.0f; |
||||
AudioMemory(5); |
||||
AudioMemory_F32(10); |
||||
Serial.begin(300); delay(1000); |
||||
Serial.println("*** Test FIR General routine for F32 ***"); |
||||
|
||||
codec.enable(); |
||||
codec.inputSelect(AUDIO_INPUT_LINEIN);
|
||||
codec.adcHighPassFilterDisable(); |
||||
codec.lineInLevel(2); |
||||
codec.volume(0.8); |
||||
|
||||
// Un-comment one of the following, or use the direct Coefficient load that follows
|
||||
|
||||
// Test case for odd nFIR from Burris and Parks Example 3.1 with Kaiser Window
|
||||
// nFIR = 21; sidelobes = 45.0f;
|
||||
// dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=0.0; dbA[5]=0.0;
|
||||
// dbA[6]=-140.0; dbA[7]=-140.0; dbA[8]=-140.0; dbA[9]=-140.0; dbA[10]=-140.0;
|
||||
|
||||
// Test case for even nFIR from Burris and Parks Example 3.2
|
||||
// nFIR = 20; sidelobes = 45.0f;
|
||||
// dbA[0]=0.0; dbA[1]=0.0; dbA[2]=0.0; dbA[3]=0.0; dbA[4]=0.0; dbA[5]=-120.0;
|
||||
// dbA[6]=-120.0; dbA[7]=-120.0; dbA[8]=-120.0; dbA[9]=-120.0;
|
||||
|
||||
// The next example tests extremely large FIR arrays and *only for Teensy 4*
|
||||
// Narrow band "CW" filter, tried with 60 dB sidelobes (60 to 80 or 70 to 72
|
||||
nFIR = 3999; sidelobes = 60; |
||||
for(i=0; i<2000; i++) { |
||||
if (i<70 || i>72) dbA[i] = -120.0f; |
||||
else dbA[i] = 0.0f; |
||||
} |
||||
|
||||
// Initialize the FIR filter design and run
|
||||
uint16_t eq = firg1.FIRGeneralNew(&dbA[0], nFIR, &equalizeCoeffs[0], sidelobes, &workSpace[0]); |
||||
if (eq == ERR_EQ_BANDS) |
||||
Serial.println("FIR General failed: Invalid number of frequency points."); |
||||
else if (eq == ERR_EQ_SIDELOBES) |
||||
Serial.println("FIR General failed: Invalid sidelobe specification."); |
||||
else if (eq == ERR_EQ_NFIR) |
||||
Serial.println("FIR General failed: Invalid number of FIR coefficients."); |
||||
else |
||||
Serial.println("FIR General initialized successfully."); |
||||
} |
||||
|
||||
void loop(void) { |
||||
}
|
||||
@ -0,0 +1,90 @@ |
||||
/*
|
||||
* radioNoiseBlanker_F32.cpp |
||||
* |
||||
* 22 March 2020 |
||||
* Bob Larkin, in support of the library: |
||||
* Chip Audette, OpenAudio, Apr 2017 |
||||
* ------------------- |
||||
* |
||||
* MIT License, Use at your own risk. |
||||
*/ |
||||
|
||||
#include "radioNoiseBlanker_F32.h" |
||||
|
||||
void radioNoiseBlanker_F32::update(void) { |
||||
audio_block_f32_t *blockIn, *blockOut=NULL; |
||||
uint16_t i; |
||||
float32_t absSignal; |
||||
|
||||
// Get input block // <<Writable??
|
||||
blockIn = AudioStream_F32::receiveWritable_f32(0); |
||||
if (!blockIn) { |
||||
if(errorPrint) Serial.println("NB-ERR: No input memory"); |
||||
return; |
||||
} |
||||
|
||||
// Are we noise blanking?
|
||||
if(! runNB) { |
||||
AudioStream_F32::transmit(blockIn, 0); // send the delayed or blanked data
|
||||
AudioStream_F32::release (blockIn); |
||||
return; |
||||
} |
||||
|
||||
// Get a block for the output
|
||||
blockOut = AudioStream_F32::allocate_f32(); |
||||
if (!blockOut){ // Didn't have any
|
||||
if(errorPrint) Serial.println("NB-ERR: No output memory"); |
||||
AudioStream_F32::release(blockIn); |
||||
return; |
||||
} |
||||
|
||||
// delayData[] always represents 256 points of I-F data. It is pre-gate and includes noise pulses.
|
||||
// Go through new data, point i at a time, entering to delay line, looking
|
||||
// for noise pulses. Then in same loop, move data to output buffer blockOut->data
|
||||
// based on whether gate is open or not.
|
||||
for(i=0; i<block_size; i++) { |
||||
float32_t datai = blockIn->data[i]; // ith data
|
||||
delayData[(i+in_index) & delayBufferMask] = datai; // Put ith data to circular delay buffer
|
||||
|
||||
absSignal = fabsf(datai); // Rectified I-F
|
||||
runningSum += fabsf(datai); // Update by adding one rectified point
|
||||
runningSum -= delayData[(i + in_index - RUNNING_SUM_SIZE) & delayBufferMask]; // & subtract one
|
||||
|
||||
pulseTime++; // This keeps track of leading and trailing delays of the gate pulse
|
||||
if (absSignal > (threshold * runningSum)) { // A noise pulse event
|
||||
if (gateOn == true) { // Signals are flowing, this is beginning of noise pulse event
|
||||
gateOn = false; |
||||
pulseTime = -nAnticipation; |
||||
} |
||||
else { // gateOn==false, we are already in a noise pulse event
|
||||
if (pulseTime > 0) { // Waiting for pulse event to end
|
||||
pulseTime = 0; // Keep waiting
|
||||
} |
||||
} |
||||
} |
||||
else { // Noise pulse is below threshold
|
||||
if (gateOn == true) { // Signals are flowing normally
|
||||
pulseTime = -9999; |
||||
} |
||||
else { // gateOn==false, we are already in a noise pulse event
|
||||
if(pulseTime >= nDecay) { // End of a pulse event, turn gate on
|
||||
gateOn = true; |
||||
pulseTime = -9999; |
||||
} |
||||
} |
||||
} |
||||
// Ready to enter I-F data to output, offset in time by "nAnticipation"
|
||||
if (pulseTime == -9999) |
||||
blockOut->data[i] = delayData[(256 + i - nAnticipation) & delayBufferMask]; // Need 256??
|
||||
else // -nAnticipation < pulseTime < nDecay i.e., blanked out
|
||||
blockOut->data[i] = 0.0f; |
||||
} // End of loop point by point over input 128 data points
|
||||
|
||||
AudioStream_F32::release (blockIn); |
||||
AudioStream_F32::transmit(blockOut, 0); // send the delayed or blanked data
|
||||
AudioStream_F32::release (blockOut); |
||||
|
||||
// Update pointer in_index to delay line for next 128 update
|
||||
in_index = (in_index + block_size) & delayBufferMask; |
||||
} |
||||
|
||||
@ -0,0 +1,126 @@ |
||||
/* radioNoiseBlanker_F32.h
|
||||
* 15 May 2020 Bob Larkin |
||||
*
|
||||
* Thanks to PJRC and Pau Stoffregen for the Teensy processor, Teensyduino |
||||
* and Teensy Audio. Thanks to Chip Audette for the F32 extension |
||||
* work. Alll of that makes this possible. Bob |
||||
*
|
||||
* Copyright (c) 2020 Bob Larkin |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice, development funding notice, and this permission |
||||
* notice shall be included in all copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN |
||||
* THE SOFTWARE. |
||||
*/ |
||||
|
||||
/*
|
||||
* An I-F signal comes in with occassional noise pulses. This block watches for |
||||
* the pulses and turns off the I-F while the pulse exists. In order to |
||||
* be as smart as possible this looks ahead by a number of samples, nAnticipation. |
||||
* Likewise, the I-F is left off for nDecay samples after the pulse ends. |
||||
* Various methods could be to be used to "turn off" the i-f, |
||||
* including replacement with waveforms. |
||||
* As of this initial write, zeros are used in the waveform. |
||||
*
|
||||
* A threshold can be adjusted via setNB(). This is compared with the |
||||
* average rectified voltage being received. If this is too small, like 1.5 |
||||
* or 2.0, we will be loosing data by blanking. If we set it too high, like 20.0, |
||||
* we will not blank noise pulses. Experiments will find a god setting. |
||||
* With a sine wave input and no impulse noise, the average rectified signal |
||||
* should be about 0.637. To catch the top of that requires a threshold of |
||||
* 1/0.637 = 1.57. That would seem to be a very minimal threshold setting. |
||||
* |
||||
* Status: Tested on computer data |
||||
* Functions: |
||||
* setNB( ... ); // See below
|
||||
* showError(e); // e=1 means to print update() error messages
|
||||
* enable(runNB); // When runNB=false the signals bypasses the update() processing
|
||||
* Examples: |
||||
* TestNoiseBlanker1.ino 128 data for plotting and examination |
||||
* Time: Update() of 128 samples 32 microseconds |
||||
*/ |
||||
|
||||
#ifndef _radio_noise_blanker_f32_h |
||||
#define _radio_noise_blanker_f32_h |
||||
|
||||
#include "AudioStream_F32.h" |
||||
#include "arm_math.h" |
||||
|
||||
#define RUNNING_SUM_SIZE 125 |
||||
|
||||
class radioNoiseBlanker_F32 : public AudioStream_F32 { |
||||
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
|
||||
//GUI: shortName: NoiseBlanker
|
||||
public: |
||||
// Option of AudioSettings_F32 change to block size (no sample rate dependent variables here):
|
||||
radioNoiseBlanker_F32(void) : AudioStream_F32(1, inputQueueArray_f32) { |
||||
block_size = AUDIO_BLOCK_SAMPLES; |
||||
} |
||||
radioNoiseBlanker_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) { |
||||
block_size = settings.audio_block_samples; |
||||
} |
||||
|
||||
void enable(bool _runNB) { |
||||
runNB = _runNB; |
||||
}
|
||||
|
||||
void setNoiseBlanker(float32_t _threshold, uint16_t _nAnticipation, uint16_t _nDecay) {
|
||||
if (_threshold < 0.0) threshold = 0.0; |
||||
else threshold = _threshold/(float32_t)RUNNING_SUM_SIZE; |
||||
|
||||
if(_nAnticipation < 1) nAnticipation = 1; |
||||
else if (_nAnticipation >125) nAnticipation = 125; |
||||
else nAnticipation = _nAnticipation; |
||||
|
||||
if (_nDecay < 1) nDecay = 1; |
||||
else if (_nDecay > 10) nDecay = 10; // Is this handled OK at end of block?
|
||||
else nDecay = _nDecay; |
||||
} |
||||
|
||||
void showError(uint16_t e) { // This is for debug
|
||||
errorPrint = e; |
||||
} |
||||
|
||||
void update(void); |
||||
|
||||
private: |
||||
uint16_t block_size = AUDIO_BLOCK_SAMPLES; |
||||
// Input data pointers
|
||||
audio_block_f32_t *inputQueueArray_f32[1]; |
||||
|
||||
// Control error printing in update() 0=No print
|
||||
uint16_t errorPrint = 0; |
||||
|
||||
// To look ahead we need a delay of up to 128 samples. Too much removes good data.
|
||||
// Too little enters noise pulse data to the output. This can be a simple circular
|
||||
// buffer if we make the buffer a power of 2 in length and binary-truncate the index.
|
||||
// Choose 2^8 = 256.
|
||||
float32_t delayData[256]; // The circular delay line
|
||||
uint16_t in_index = 0; // Pointer to next block update entry
|
||||
// And a mask to make the circular buffer limit to a power of 2
|
||||
uint16_t delayBufferMask = 0X00FF; |
||||
|
||||
// Three variables to allow .INO control of Noise Blanker
|
||||
float32_t threshold = 1.0E6f; // Start disabled
|
||||
uint16_t nAnticipation = 5; |
||||
uint16_t nDecay = 8; |
||||
|
||||
int16_t pulseTime = -9999; // Tracks nAnticipation and nDecay
|
||||
bool gateOn = true; // Signals going through NB
|
||||
float32_t runningSum = 0.0; |
||||
bool runNB = false; |
||||
}; |
||||
#endif |
||||
@ -0,0 +1,103 @@ |
||||
/* synth_sin_cos_f32.cpp
|
||||
*
|
||||
* SynthSinCos_F32 Bob Larkin April 17, 2020 |
||||
* |
||||
* Based on Chip Audette's OpenAudio sine(), that was |
||||
* Modeled on: AudioSynthWaveformSine from Teensy Audio Library |
||||
* |
||||
* Purpose: Create sine and cosine wave of given amplitude, frequency |
||||
* and phase. Outputs in float32_t floating point. |
||||
* Routines are from the arm CMSIS library and use a 512 point lookup |
||||
* table with linear interpolation to achieve float accuracy limits. |
||||
* |
||||
* Copyright (c) 2020 Bob Larkin |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
#include "synth_sin_cos_f32.h" |
||||
|
||||
// 513 values of the sine wave in a float array:
|
||||
#include "sinTable512_f32.h" |
||||
|
||||
void AudioSynthSineCosine_F32::update(void) { |
||||
audio_block_f32_t *blockS, *blockC; |
||||
uint16_t index, i; |
||||
float32_t a, b, deltaPhase, phaseC; |
||||
|
||||
blockS = AudioStream_F32::allocate_f32(); // Output blocks
|
||||
if (!blockS) return; |
||||
|
||||
blockC = AudioStream_F32::allocate_f32(); |
||||
if (!blockC) { |
||||
AudioStream_F32::release(blockS); |
||||
return; |
||||
} |
||||
|
||||
// doSimple has amplitude (-1, 1) and sin/cos differ by 90.00 degrees.
|
||||
if (doSimple) { |
||||
for (i=0; i < block_length; i++) { |
||||
phaseS += phaseIncrement; |
||||
if (phaseS > 512.0f) |
||||
phaseS -= 512.0f; |
||||
index = (uint16_t) phaseS; |
||||
deltaPhase = phaseS -(float32_t) index; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index]; |
||||
b = sinTable512_f32[index+1]; |
||||
blockS->data[i] = a + 0.001953125*(b-a)*deltaPhase; /* Linear interpolation process */ |
||||
|
||||
/* Repeat for cosine by adding 90 degrees phase */ |
||||
index = (index + 128) & 0x01ff; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index]; |
||||
b = sinTable512_f32[index+1]; |
||||
/* deltaPhase will be the same as used for sin */ |
||||
blockC->data[i] = a + 0.001953125*(b-a)*deltaPhase; /* Linear interpolation process */ |
||||
} |
||||
} |
||||
else { // Do a more flexible update, i.e., not doSimple
|
||||
for (i=0; i < block_length; i++) { |
||||
phaseS += phaseIncrement; |
||||
if (phaseS > 512.0f) phaseS -= 512.0f; |
||||
index = (uint16_t) phaseS; |
||||
deltaPhase = phaseS -(float32_t) index; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index]; |
||||
b = sinTable512_f32[index+1]; |
||||
blockS->data[i] = amplitude_pk * (a + 0.001953125*(b-a)*deltaPhase); /* Linear interpolation process */ |
||||
|
||||
/* Shift forward phaseS_C and get cos. First, the calculation of index of the table */ |
||||
phaseC = phaseS + phaseS_C; |
||||
if (phaseC > 512.0f) phaseC -= 512.0f; |
||||
index = (uint16_t) phaseC; |
||||
deltaPhase = phaseC -(float32_t) index; |
||||
/* Read two nearest values of input value from the sin table */ |
||||
a = sinTable512_f32[index]; |
||||
b = sinTable512_f32[index+1]; |
||||
blockC->data[i] = amplitude_pk * (a + 0.001953125*(b-a)*deltaPhase); /* Linear interpolation process */ |
||||
} |
||||
} |
||||
AudioStream_F32::transmit(blockS, 0); |
||||
AudioStream_F32::release (blockS);
|
||||
AudioStream_F32::transmit(blockC, 1); |
||||
AudioStream_F32::release (blockC); |
||||
return; |
||||
} |
||||
|
||||
@ -0,0 +1,162 @@ |
||||
/* synth_sin_cos_f32.h
|
||||
* AudioSynthSineCosine_F32 |
||||
*
|
||||
* Status: Checked for function and accuracy. 19 April 2020 |
||||
* |
||||
* Created: Bob Larkin 15 April 2020 |
||||
* |
||||
* Based on Chip Audette's OpenAudio sine(), that was |
||||
* Modeled on: AudioSynthWaveformSine from Teensy Audio Library |
||||
* |
||||
* Purpose: Create sine and cosine wave of given amplitude, frequency |
||||
* and phase. Outputs are audio_block_f32_t blocks of float32_t. |
||||
* Routines are from the arm CMSIS library and use a 512 point lookup |
||||
* table with linear interpolation to achieve float accuracy limits. |
||||
* |
||||
* This provides for setting the phase of the sine, setting the difference |
||||
* in phase between the sine and cosine and setting the |
||||
* (-amplitude, amplitude) range. If these are at the default values, |
||||
* called doSimple, the caluclation is faster. |
||||
* For doSimple either true or false, the frequency can be changed |
||||
* at will using the frequency() method.
|
||||
*
|
||||
* Defaults: |
||||
* Frequency: 1000.0 Hz |
||||
* Phase of Sine: 0.0 radians (0.0 deg) |
||||
* Phase of Cosine: pi/2 radians (90.0 deg) ahead of Sine |
||||
* Amplitude: -1.0 to 1.0 |
||||
*
|
||||
* Time: T3.6 update() block of 128 with doSimple is 36 microseconds |
||||
* Same using flexible doSimple=false is 49 microseconds |
||||
* T4.0 update() block of 128 with doSimple is 16 microseconds |
||||
* Same using flexible doSimple=false is 24 microseconds |
||||
*
|
||||
* Copyright (c) 2020 Bob Larkin |
||||
* |
||||
* Permission is hereby granted, free of charge, to any person obtaining a copy |
||||
* of this software and associated documentation files (the "Software"), to deal |
||||
* in the Software without restriction, including without limitation the rights |
||||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell |
||||
* copies of the Software, and to permit persons to whom the Software is |
||||
* furnished to do so, subject to the following conditions: |
||||
* |
||||
* The above copyright notice and this permission notice shall be included in all |
||||
* copies or substantial portions of the Software. |
||||
* |
||||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR |
||||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, |
||||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE |
||||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER |
||||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, |
||||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE |
||||
* SOFTWARE. |
||||
*/ |
||||
|
||||
#ifndef synth_sin_cos_f32_h_ |
||||
#define synth_sin_cos_f32_h_ |
||||
|
||||
#include "AudioStream_F32.h" |
||||
#include "arm_math.h" |
||||
|
||||
#ifndef M_PI |
||||
#define M_PI 3.14159265358979323846 |
||||
#endif |
||||
|
||||
#ifndef M_PI_2 |
||||
#define M_PI_2 1.57079632679489661923 |
||||
#endif |
||||
|
||||
#ifndef M_TWOPI |
||||
#define M_TWOPI (M_PI * 2.0) |
||||
#endif |
||||
|
||||
#define MF2_PI 6.2831853f |
||||
|
||||
class AudioSynthSineCosine_F32 : public AudioStream_F32 { |
||||
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI node
|
||||
//GUI: shortName:SineCosine //this line used for automatic generation of GUI node
|
||||
public: |
||||
AudioSynthSineCosine_F32(void) : AudioStream_F32(0, NULL) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
|
||||
|
||||
AudioSynthSineCosine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
|
||||
setSampleRate_Hz(settings.sample_rate_Hz); |
||||
setBlockLength(settings.audio_block_samples); |
||||
} |
||||
|
||||
void frequency(float32_t fr) { // Frequency in Hz
|
||||
freq = fr; |
||||
if (freq < 0.0f) freq = 0.0f; |
||||
else if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.0f; |
||||
phaseIncrement = 512.0f * freq / sample_rate_Hz; |
||||
} |
||||
|
||||
/* Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
|
||||
* Internally, the full circle is represented as (0.0, 512.0). This is |
||||
* convenient for finding the entry to the sine table. |
||||
*/ |
||||
void phase_r(float32_t a) { |
||||
while (a < 0.0f) a += MF2_PI; |
||||
while (a > MF2_PI) a -= MF2_PI; |
||||
phaseS = 512.0f * a / MF2_PI; |
||||
doSimple = false; |
||||
return; |
||||
} |
||||
|
||||
// phaseS_C_r is the number of radians that the cosine output leads the
|
||||
// sine output. The default is M_PI_2 = pi/2 = 1.57079633 radians,
|
||||
// corresponding to 90.00 degrees cosine leading sine.
|
||||
void phaseS_C_r(float32_t a) { |
||||
while (a < 0.0f) a += MF2_PI; |
||||
while (a > MF2_PI) a -= MF2_PI; |
||||
// Internally a full circle is 512.00 of phase
|
||||
phaseS_C = 512.0f * a / MF2_PI; |
||||
doSimple = false; |
||||
return; |
||||
} |
||||
|
||||
// The amplitude, a, is the peak, as in zero-to-peak. This produces outputs
|
||||
// ranging from -a to +a. Both outputs are the same amplitude.
|
||||
void amplitude(float32_t a) { |
||||
amplitude_pk = a; |
||||
doSimple = false; |
||||
return; |
||||
} |
||||
|
||||
// Speed up calculations by setting phaseS_C=90deg, amplitude=1
|
||||
// Note, s=true will override any setting of phaseS_C_r or amplitude.
|
||||
void simple(bool s) { |
||||
doSimple = s; |
||||
if(doSimple) { |
||||
phaseS_C = 128.0f; |
||||
amplitude_pk = 1.0f; |
||||
} |
||||
return; |
||||
}
|
||||
|
||||
void setSampleRate_Hz(float32_t fs_Hz) { |
||||
// Check freq range
|
||||
if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.f; |
||||
// update phase increment for new frequency
|
||||
phaseIncrement = 512.0f * freq / fs_Hz; |
||||
} |
||||
|
||||
void setBlockLength(uint16_t bl) { |
||||
if(bl > 128) bl = 128; |
||||
block_length = bl; |
||||
} |
||||
|
||||
virtual void update(void); |
||||
|
||||
private: |
||||
float32_t freq = 1000.0f; |
||||
float32_t phaseS = 0.0f; |
||||
float32_t phaseS_C = 128.00; |
||||
float32_t amplitude_pk = 1.0f; |
||||
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE; |
||||
float32_t phaseIncrement = 512.0f * freq /sample_rate_Hz; |
||||
uint16_t block_length = 128; |
||||
// if only freq() is used, the complexities of phase, phaseS_C,
|
||||
// and amplitude are not used, speeding up the sin and cos:
|
||||
bool doSimple = true; |
||||
}; |
||||
#endif |
||||
Loading…
Reference in new issue