fft: Define MicroBitAudioProcessor

Signed-off-by: Raphael Gault <[email protected]>

Author: raphaelgault

inc/MicroBitAudioProcessor.h

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+/*
+The MIT License (MIT)
+
+Copyright (c) 2020 Arm Limited.
+
+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 "MicroBit.h"
+#include "DataStream.h"
+#define ARM_MATH_CM4
+#include "arm_math.h"
+
+#ifndef MICROBIT_AUDIO_PROCESSOR_H
+#define MICROBIT_AUDIO_PROCESSOR_H
+
+#define AUDIO_SAMPLES_NUMBER    256
+
+class MicroBitAudioProcessor : public DataSink
+{
+    DataSource      &audiostream;          
+    int             zeroOffset;             // unsigned value that is the best effort guess of the zero point of the data source
+    int             divisor;                // Used for LINEAR modes
+    arm_rfft_fast_instance_f32 fft_instance;
+    float *buf; 
+    float *output; 
+    uint16_t position;
+
+    public:
+    MicroBitAudioProcessor(DataSource& source); 
+    ~MicroBitAudioProcessor(); 
+    virtual int pullRequest();
+    int setDivisor(int d);
+};
+
+#endif

source/MicroBitAudioProcessor.cpp

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+/*
+The MIT License (MIT)
+
+Copyright (c) 2020 Arm Limited.
+
+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 "MicroBit.h"
+#include "MicroBitAudioProcessor.h"
+
+MicroBitAudioProcessor::MicroBitAudioProcessor(DataSource& source) : audiostream(source)
+{
+    audiostream.connect(*this);
+    zeroOffset = 0;
+    divisor = 1;
+    //arm_rfft_fast_init_f32(&fft_instance, AUDIO_SAMPLES_NUMBER);
+
+    /* Double Buffering: We allocate twice the number of samples*/
+    buf = (float *)malloc(sizeof(float) * AUDIO_SAMPLES_NUMBER * 2);
+    output = (float *)malloc(sizeof(float) * AUDIO_SAMPLES_NUMBER);
+
+    position = 0;
+
+    if (buf == NULL || output == NULL) {
+        DMESG("DEVICE_NO_RESOURCES");
+        target_panic(DEVICE_OOM);
+    }
+}
+
+MicroBitAudioProcessor::~MicroBitAudioProcessor()
+{
+    free(buf);
+    free(output);
+}
+
+int MicroBitAudioProcessor::pullRequest()
+{
+    int z = 0;
+    int minimum = 0;
+    int maximum = 0;
+    int s;
+    int8_t result;
+
+    auto mic_samples = audiostream.pull().getBytes();
+
+    int16_t *data = (int16_t *) &mic_samples[0];
+    int samples = AUDIO_SAMPLES_NUMBER / 2;
+
+    for (int i=0; i < samples; i++)
+    {
+        z += *data;
+
+        s = (int) *data;
+        s = s - zeroOffset;
+        s = s / divisor;
+        result = (int8_t)s;
+
+        if (s < minimum)
+            minimum = s;
+
+        if (s > maximum)
+            maximum = s;
+
+        data++;
+        buf[position++] = (float)result;
+
+        if (!(position % AUDIO_SAMPLES_NUMBER))
+        {
+        /* We have AUDIO_SAMPLES_NUMBER samples, we can run the FFT on them */
+            uint8_t offset = position <= AUDIO_SAMPLES_NUMBER ? 0 : AUDIO_SAMPLES_NUMBER;
+            if (offset != 0)
+                position = 0;
+
+            //arm_rfft_fast_f32(&fft_instance, buf + offset, output, 0);
+            DMESG("o[0]: %d", output[0]);
+        }
+    }
+
+    z = z / samples;
+    zeroOffset = z;
+    return DEVICE_OK;
+}
+
+int MicroBitAudioProcessor::setDivisor(int d)
+{
+    divisor = d;
+    return DEVICE_OK;
+}