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author | unknown <dm2515@eews303a-028.ic.ac.uk> | 2018-02-01 11:45:25 +0000 |
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committer | unknown <dm2515@eews303a-028.ic.ac.uk> | 2018-02-01 11:45:25 +0000 |
commit | e6ef21c67becd2e136608957e6cbadc9bc5eed73 (patch) | |
tree | 3233910f9f4539798210bb793b7d04b053d98d31 /lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd | |
parent | d15acaf23dcc779bd65b393ace28178bd8ac152d (diff) | |
download | NoiseSilencer-e6ef21c67becd2e136608957e6cbadc9bc5eed73.tar.gz NoiseSilencer-e6ef21c67becd2e136608957e6cbadc9bc5eed73.zip |
FInished lab4ex2
Diffstat (limited to 'lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd')
-rw-r--r-- | lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd/f0f0fa5f41070018153483e962c7925e | 153 |
1 files changed, 153 insertions, 0 deletions
diff --git a/lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd/f0f0fa5f41070018153483e962c7925e b/lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd/f0f0fa5f41070018153483e962c7925e new file mode 100644 index 0000000..d7dceeb --- /dev/null +++ b/lab4/.metadata/.plugins/org.eclipse.core.resources/.history/dd/f0f0fa5f41070018153483e962c7925e @@ -0,0 +1,153 @@ +/************************************************************************************* + DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING + IMPERIAL COLLEGE LONDON + + EE 3.19: Real Time Digital Signal Processing + Dr Paul Mitcheson and Daniel Harvey + + LAB 3: Interrupt I/O + + ********* I N T I O. C ********** + + Demonstrates inputing and outputing data from the DSK's audio port using interrupts. + + ************************************************************************************* + Updated for use on 6713 DSK by Danny Harvey: May-Aug 2006 + Updated for CCS V4 Sept 10 + ************************************************************************************/ +/* + * You should modify the code so that interrupts are used to service the + * audio port. + */ +/**************************** Pre-processor statements ******************************/ + +#include <stdlib.h> +// Included so program can make use of DSP/BIOS configuration tool. +#include "dsp_bios_cfg.h" + +/* The file dsk6713.h must be included in every program that uses the BSL. This + example also includes dsk6713_aic23.h because it uses the + AIC23 codec module (audio interface). */ +#include "dsk6713.h" +#include "dsk6713_aic23.h" + +// math library (trig functions) +#include <math.h> + +// Some functions to help with writing/reading the audio ports when using interrupts. +#include <helper_functions_ISR.h> +#include "Matlab/filter_coeff.txt" +// Some functions to help with configuring hardware +#include "helper_functions_polling.h" + + +// PI defined here for use in your code +#define PI 3.141592653589793 +#define N 249 +double buffer[N]= {0}; +unsigned int ptr = N-1; + +/******************************* Global declarations ********************************/ + +/* Audio port configuration settings: these values set registers in the AIC23 audio + interface to configure it. See TI doc SLWS106D 3-3 to 3-10 for more info. */ +DSK6713_AIC23_Config Config = { \ + /**********************************************************************/ + /* REGISTER FUNCTION SETTINGS */ + /**********************************************************************/\ + 0x0017, /* 0 LEFTINVOL Left line input channel volume 0dB */\ + 0x0017, /* 1 RIGHTINVOL Right line input channel volume 0dB */\ + 0x01f9, /* 2 LEFTHPVOL Left channel headphone volume 0dB */\ + 0x01f9, /* 3 RIGHTHPVOL Right channel headphone volume 0dB */\ + 0x0011, /* 4 ANAPATH Analog audio path control DAC on, Mic boost 20dB*/\ + 0x0000, /* 5 DIGPATH Digital audio path control All Filters off */\ + 0x0000, /* 6 DPOWERDOWN Power down control All Hardware on */\ + 0x0043, /* 7 DIGIF Digital audio interface format 16 bit */\ + 0x008d, /* 8 SAMPLERATE Sample rate control 8 KHZ */\ + 0x0001 /* 9 DIGACT Digital interface activation On */\ + /**********************************************************************/ +}; + + +// Codec handle:- a variable used to identify audio interface +DSK6713_AIC23_CodecHandle H_Codec; + + /******************************* Function prototypes ********************************/ +void init_hardware(void); +void init_HWI(void); +void ISR_AIC(void); +short non_circ_fir(void); +/********************************** Main routine ************************************/ +void main(){ + // initialize board and the audio port + init_hardware(); + + /* initialize hardware interrupts */ + init_HWI(); + + /* loop indefinitely, waiting for interrupts */ + while(1) {}; +} + +/********************************** init_hardware() **********************************/ +void init_hardware() +{ + // Initialize the board support library, must be called first + DSK6713_init(); + + // Start the AIC23 codec using the settings defined above in config + H_Codec = DSK6713_AIC23_openCodec(0, &Config); + + /* Function below sets the number of bits in word used by MSBSP (serial port) for + receives from AIC23 (audio port). We are using a 32 bit packet containing two + 16 bit numbers hence 32BIT is set for receive */ + MCBSP_FSETS(RCR1, RWDLEN1, 32BIT); + + /* Configures interrupt to activate on each consecutive available 32 bits + from Audio port hence an interrupt is generated for each L & R sample pair */ + MCBSP_FSETS(SPCR1, RINTM, FRM); + + /* These commands do the same thing as above but applied to data transfers to + the audio port */ + MCBSP_FSETS(XCR1, XWDLEN1, 32BIT); + MCBSP_FSETS(SPCR1, XINTM, FRM); +} + +/********************************** init_HWI() **************************************/ +void init_HWI() +{ + IRQ_globalDisable(); // Globally disables interrupts + IRQ_nmiEnable(); // Enables the NMI interrupt (used by the debugger) + IRQ_map(IRQ_EVT_RINT1,4); // Maps an event to a physical interrupt + IRQ_enable(IRQ_EVT_RINT1); // Enables the event + IRQ_globalEnable(); // Globally enables interrupts +} + +/******************** INTERRUPT SERVICE ROUTINE ***********************/ +void ISR_AIC() +{ + short sample_in, sample_out; + + sample_in = mono_read_16Bit(); + buffer[ptr] = (float) sample_in / 32767.f; + sample_out = non_circ_fir(); + mono_write_16Bit(sample_out); + + if (ptr == 0) + ptr = N; + ptr--; +} + +// Perform linear convolution +short non_circ_fir() +{ + + double y = 0; + int M, i; + M = sizeof(b) / sizeof(b[0]); + for(i = 0; i < N; i++) { + y += buffer[i] * b[M-i-1]; + } + return y*32767; +} + |