Each object includes only required headers

analyze_fft256.cpp

@@ -1,9 +1,7 @@
-#include "Audio.h"
+#include "analyze_fft256.h"
 #include "arm_math.h"
-#include "utility/dspinst.h"
-
-
 
+// TODO: this should be a class member, so more than one FFT can be used
 static arm_cfft_radix4_instance_q15 fft_inst;
 
 void AudioAnalyzeFFT256::init(void)

analyze_fft256.h

@@ -2,7 +2,6 @@
 #define analyze_fft256_h_
 
 #include "AudioStream.h"
-#include "arm_math.h"
 
 // windows.c
 extern "C" {

analyze_print.cpp

@@ -1,11 +1,9 @@
-#include "Audio.h"
-
+#include "analyze_print.h"
 
 // TODO: this needs some sort of trigger or delay or other options to make
 // actually useful for watching interesting parts of data, without spewing
 // tremendous and endless data to the Arduino Serial Monitor
 
-
 void AudioPrint::update(void)
 {
 	audio_block_t *block;

analyze_tonedetect.cpp

@@ -1,9 +1,6 @@
-#include "Audio.h"
-#include "arm_math.h"
+#include "analyze_tonedetect.h"
 #include "utility/dspinst.h"
 
-
-
 static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b) __attribute__((always_inline));
 static inline int32_t multiply_32x32_rshift30(int32_t a, int32_t b)
 {

control_sgtl5000.cpp

@@ -1,7 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-
-
+#include "control_sgtl5000.h"
 #include "Wire.h"
 
 #define CHIP_ID				0x0000

control_wm8731.cpp

@@ -1,7 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-
-
+#include "control_wm8731.h"
 #include "Wire.h"
 
 #define WM8731_I2C_ADDR 0x1A

effect_chorus.cpp

@@ -1,7 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-#include "utility/dspinst.h"
-
+#include "effect_chorus.h"
 
 /******************************************************************/
 

effect_fade.cpp

@@ -1,14 +1,10 @@
-#include "Audio.h"
-#include "arm_math.h"
+#include "effect_fade.h"
 #include "utility/dspinst.h"
 
-
-
 extern "C" {
 extern const int16_t fader_table[256];
 };
 
-
 void AudioEffectFade::update(void)
 {
 	audio_block_t *block;

effect_flange.cpp

@@ -1,9 +1,5 @@
-#include "Audio.h"
+#include "effect_flange.h"
 #include "arm_math.h"
-#include "utility/dspinst.h"
-
-
-
 
 /******************************************************************/
 //                A u d i o E f f e c t F l a n g e

filter_biquad.cpp

@@ -1,8 +1,6 @@
-#include "Audio.h"
-#include "arm_math.h"
+#include "filter_biquad.h"
 #include "utility/dspinst.h"
 
-
 void AudioFilterBiquad::update(void)
 {
 	audio_block_t *block;

filter_fir.cpp

@@ -1,8 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-#include "utility/dspinst.h"
-
-
+#include "filter_fir.h"
 
 void AudioFilterFIR::begin(short *cp,int n_coeffs)
 {

filter_fir.h

@@ -2,6 +2,7 @@
 #define filter_fir_h_
 
 #include "AudioStream.h"
+#include "arm_math.h"
 
 // Maximum number of coefficients in a FIR filter
 // The audio breaks up with 128 coefficients so a

input_adc.cpp

@@ -1,7 +1,6 @@
-#include "Audio.h"
+#include "input_adc.h"
 #include "utility/pdb.h"
 
-
 DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalog::block_left = NULL;
 uint16_t AudioInputAnalog::block_offset = 0;

input_i2s.cpp

@@ -1,6 +1,5 @@
-#include "Audio.h"
-#include "arm_math.h"
-
+#include "input_i2s.h"
+#include "output_i2s.h"
 
 DMAMEM static uint32_t i2s_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputI2S::block_left = NULL;

mixer.cpp

@@ -1,9 +1,6 @@
-#include "Audio.h"
-#include "arm_math.h"
+#include "mixer.h"
 #include "utility/dspinst.h"
 
-
-
 void applyGain(int16_t *data, int32_t mult)
 {
 	uint32_t *p = (uint32_t *)data;

output_dac.cpp

@@ -1,9 +1,6 @@
-#include "Audio.h"
+#include "output_dac.h"
 #include "utility/pdb.h"
 
-// #define PDB_CONFIG (PDB_SC_TRGSEL(15) | PDB_SC_PDBEN | PDB_SC_CONT)
-// #define PDB_PERIOD 1087 // 48e6 / 44100
-
 #if defined(__MK20DX256__)
 
 DMAMEM static uint16_t dac_buffer[AUDIO_BLOCK_SAMPLES*2];

output_i2s.cpp

@@ -1,6 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-
+#include "output_i2s.h"
 
 // MCLK needs to be 48e6 / 1088 * 256 = 11.29411765 MHz -> 44.117647 kHz sample rate
 // Possible to create using fractional divider for all USB-compatible Kinetis:
@@ -15,7 +13,6 @@
 // clock, which has audio band jitter from the PLL
 
 
-
 audio_block_t * AudioOutputI2S::block_left_1st = NULL;
 audio_block_t * AudioOutputI2S::block_right_1st = NULL;
 audio_block_t * AudioOutputI2S::block_left_2nd = NULL;
@@ -57,8 +54,6 @@ void AudioOutputI2S::begin(void)
 }
 
 
-
-
 void dma_ch0_isr(void)
 {
 	const int16_t *src, *end;

output_pwm.cpp

@@ -1,7 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-
-
+#include "output_pwm.h"
 
 audio_block_t * AudioOutputPWM::block_1st = NULL;
 audio_block_t * AudioOutputPWM::block_2nd = NULL;

play_memory.cpp

@@ -1,9 +1,7 @@
-#include "Audio.h"
-#include "arm_math.h"
+#include "play_memory.h"
 #include "utility/dspinst.h"
 
 
-
 void AudioPlayMemory::play(const unsigned int *data)
 {
 	uint32_t format;

play_sd_raw.cpp

@@ -1,7 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-#include "utility/dspinst.h"
-
+#include "play_sd_raw.h"
 
 
 void AudioPlaySDcardRAW::begin(void)

play_sd_wav.cpp

@@ -1,8 +1,4 @@
-#include "Audio.h"
-#include "arm_math.h"
-#include "utility/dspinst.h"
-
-
+#include "play_sd_wav.h"
 
 #define STATE_DIRECT_8BIT_MONO		0  // playing mono at native sample rate
 #define STATE_DIRECT_8BIT_STEREO	1  // playing stereo at native sample rate

synth_tonesweep.cpp

@@ -1,6 +1,5 @@
-#include "Audio.h"
+#include "synth_tonesweep.h"
 #include "arm_math.h"
-#include "utility/dspinst.h"
 
 
 // TODO: this object should be renamed to AudioSynthToneSweep

synth_waveform.cpp

@@ -1,10 +1,8 @@
-#include "Audio.h"
+#include "synth_waveform.h"
 #include "arm_math.h"
 #include "utility/dspinst.h"
 
 
-
-
 #ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
 /******************************************************************/
 // PAH - add ramp-up and ramp-down to the onset of the wave