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refactor
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Signed-off-by: 82Flex <[email protected]>
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@Lessica
Lessica committed Sep 20, 2024
1 parent c3fb1a7 commit c502296
Showing 1 changed file with 70 additions and 44 deletions.
114 changes: 70 additions & 44 deletions cli/dtmf-decoder.mm
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Original file line number Diff line number Diff line change
Expand Up @@ -31,20 +31,24 @@
#import "ATAudioTapDescription.h"
#import "AudioQueue+Private.h"

#define SAMPLING_RATE 8000.0
#define NUM_BUFFERS 40
#define DETECTBUFFERLEN 8192
#pragma mark - Defines

#define SAMPLE_RATE 8000.0
#define NUM_BUF 40
#define DETECT_BUFLEN 8192

#define MIN_TONE_LENGTH 0.045
#define FRAMES_PER_TONE 2
#define BYTES_PER_CHANNEL 2
#define BUFFER_SIZE ((int)(MIN_TONE_LENGTH * SAMPLING_RATE * BYTES_PER_CHANNEL) / FRAMES_PER_TONE)
#define BUFFER_SIZE ((int)(MIN_TONE_LENGTH * SAMPLE_RATE * BYTES_PER_CHANNEL) / FRAMES_PER_TONE)

#define NUM_FREQS 8
#define NUM_FREQ 8

#define kMinNoiseToleranceFactor 1.5
#define kMaxNoiseToleranceFactor 6.5

#pragma mark - Structs

struct FilterCoefficientsEntry {
double unityGainCorrection;
double coeff1;
Expand All @@ -54,7 +58,7 @@
typedef struct {
AudioStreamBasicDescription dataFormat;
AudioQueueRef queue;
AudioQueueBufferRef buffers[NUM_BUFFERS];
AudioQueueBufferRef buffers[NUM_BUF];
BOOL isRecording;
SInt64 currentPacket;
short filteredBuffer[BUFFER_SIZE];
Expand All @@ -63,17 +67,10 @@
ATAudioTap *audioTap;
} RecordState;

static double powers[NUM_FREQS]; // Location to store the powers for all the frequencies.
static double filterBuf0[NUM_FREQS]; // Buffer for the IIR filter slot 0.
static double filterBuf1[NUM_FREQS]; // Buffer for the IIR filter slot 1.
static char holdingBuffer[2];
static int holdingBufferCount[2];
static int powerMeasurementMethod; // 0 = Peak Value -> RMS, 1 = Sqrt of Sum of Squares, 2 = Sum of Abs Values
static BOOL rawOutput;
static double noiseToleranceFactor;
#pragma mark - Constants

// Filter coefficients
static const struct FilterCoefficientsEntry filterCoefficients[NUM_FREQS] = {
static const struct FilterCoefficientsEntry filterCoefficients[NUM_FREQ] = {
{0.002729634465943104, 1.703076309365611, 0.994540731068114}, // 697 Hz
{0.003014658069540622, 1.640321076289727, 0.9939706838609188}, // 770 Hz
{0.003334626751652912, 1.563455998285116, 0.9933307464966943}, // 852 Hz
Expand All @@ -91,6 +88,19 @@
{'*', '0', '#', 'D'},
};

#pragma mark - Variables

static double powers[NUM_FREQ]; // Location to store the powers for all the frequencies.
static double filterBuf0[NUM_FREQ]; // Buffer for the IIR filter slot 0.
static double filterBuf1[NUM_FREQ]; // Buffer for the IIR filter slot 1.
static char holdingBuffer[2];
static int holdingBufferCount[2];
static int powerMeasurementMethod; // 0 = Peak Value -> RMS, 1 = Sqrt of Sum of Squares, 2 = Sum of Abs Values
static BOOL rawOutput;
static double noiseToleranceFactor;

#pragma mark - Functions

// BpRe/100/frequency == Bandpass resonator, Q=100 (0=>Inf), frequency
// e.g. ./fiview 8000 -i BpRe/100/1336
// Generated using http://uazu.net/fiview/
Expand All @@ -110,7 +120,7 @@ static double bandPassFilter(register double val, int filterIndex) {
static char lookupDTMFCode(void) {
// Find the highest powered frequency index.
int max1Index = 0;
for (int i = 0; i < NUM_FREQS; i++) {
for (int i = 0; i < NUM_FREQ; i++) {
if (powers[i] >= powers[max1Index])
max1Index = i;
}
Expand All @@ -124,14 +134,14 @@ static char lookupDTMFCode(void) {
max2Index = 0;
}

for (int i = 0; i < NUM_FREQS; i++) {
for (int i = 0; i < NUM_FREQ; i++) {
if ((powers[i] >= powers[max2Index]) && (i != max1Index))
max2Index = i;
}

// Check that fequency 1 and 2 are substantially bigger than any other frequencies.
BOOL valid = YES;
for (int i = 0; i < NUM_FREQS; i++) {
for (int i = 0; i < NUM_FREQ; i++) {
if ((i == max1Index) || (i == max2Index))
continue;

Expand Down Expand Up @@ -172,9 +182,11 @@ static char lookupDTMFCode(void) {
return ' ';
}

static void AudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime, UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription *inPacketDescs) {
#pragma mark - Audio Callback

static void gAudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer,
const AudioTimeStamp *inStartTime, UInt32 inNumberPacketDescriptions,
const AudioStreamPacketDescription *inPacketDescs) {

RecordState *recordState = (RecordState *)inUserData;

Expand Down Expand Up @@ -216,13 +228,13 @@ static void AudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueB
int t;
double val;

for (t = 0; t < NUM_FREQS; t++) {
for (t = 0; t < NUM_FREQ; t++) {
powers[t] = (double)0.0;
}

// Run the bandpass filter and calculate the power.
for (i = 0L; i < numberOfSamples; i++) {
for (t = 0; t < NUM_FREQS; t++) {
for (t = 0; t < NUM_FREQ; t++) {

// Find the highest value.
switch (powerMeasurementMethod) {
Expand All @@ -243,7 +255,7 @@ static void AudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueB
}

// Scale 0 - 1, then convert into an power value.
for (t = 0; t < NUM_FREQS; t++) {
for (t = 0; t < NUM_FREQ; t++) {
switch (powerMeasurementMethod) {
case 0:
powers[t] = (powers[t] / (double)32768.0) * ((double)1.0 / sqrt((double)2.0));
Expand Down Expand Up @@ -294,7 +306,7 @@ static void AudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueB
} else {
snprintf(tmp, 20, "%c", holdingBuffer[0]);
}
if (strlen(recordState->detectBuffer) + strlen(tmp) < DETECTBUFFERLEN) {
if (strlen(recordState->detectBuffer) + strlen(tmp) < DETECT_BUFLEN) {
strcat(recordState->detectBuffer, tmp);
recordState->bufferChanged = YES;
}
Expand All @@ -311,33 +323,44 @@ static void AudioInputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueB
AudioQueueEnqueueBuffer(recordState->queue, inBuffer, 0, NULL);
}

#pragma mark - DTMFDecoder

@interface DTMFDecoder : NSObject {
AudioStreamBasicDescription mAudioFormat;
RecordState mRecordState;
float mNoiseLevel;
NSInteger mPowerMethod;
}

- (instancetype)init NS_DESIGNATED_INITIALIZER;
+ (instancetype)sharedDecoder;
- (instancetype)init NS_UNAVAILABLE;
- (void)resetBuffer;
- (void)startRecording;
- (void)stopRecording;
- (NSString *)copyBuffer;

- (BOOL)isRecording;

@property(nonatomic, assign, getter=getBufferChanged) BOOL bufferChanged;
@property(nonatomic, assign, getter=getNoiseLevel) float noiseLevel;
@property(nonatomic, assign, getter=getPowerMethod) NSInteger powerMethod;
@property(nonatomic, assign, readonly, getter=isRecording) BOOL recording;
@property(nonatomic, assign, getter=isBufferChanged) BOOL bufferChanged;
@property(nonatomic, assign) float noiseLevel;
@property(nonatomic, assign) NSInteger powerMethod;

@end

@implementation DTMFDecoder

+ (instancetype)sharedDecoder {
static DTMFDecoder *sharedDecoder = nil;
static dispatch_once_t onceToken;
dispatch_once(&onceToken, ^{
sharedDecoder = [[self alloc] init];
});
return sharedDecoder;
}

- (instancetype)init {
self = [super init];
if (self) {
mRecordState.detectBuffer = (char *)calloc(1, DETECTBUFFERLEN);
mRecordState.detectBuffer = (char *)calloc(1, DETECT_BUFLEN);
[self setNoiseLevel:0];
[self setPowerMethod:0];
[self resetBuffer];
Expand All @@ -355,9 +378,9 @@ - (void)startRecording {
holdingBuffer[i] = ' ';
}

AudioQueueBufferRef qref[NUM_BUFFERS];
AudioQueueBufferRef qref[NUM_BUF];

mAudioFormat.mSampleRate = SAMPLING_RATE;
mAudioFormat.mSampleRate = SAMPLE_RATE;
mAudioFormat.mFormatID = kAudioFormatLinearPCM;
mAudioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
mAudioFormat.mFramesPerPacket = 1;
Expand All @@ -383,7 +406,7 @@ - (void)startRecording {
mRecordState.audioTap = [[ATAudioTap alloc] initWithTapDescription:audioTapDescription];

OSStatus status;
status = AudioQueueNewInput(&mAudioFormat, AudioInputCallback,
status = AudioQueueNewInput(&mAudioFormat, gAudioInputCallback,
&mRecordState, // User Data
CFRunLoopGetCurrent(), kCFRunLoopCommonModes,
0x800, // Reserved
Expand All @@ -408,11 +431,12 @@ - (void)startRecording {

AudioQueueGetProperty(mRecordState.queue, kAudioQueueProperty_StreamDescription, &mAudioFormat, &fsize);

if (mAudioFormat.mSampleRate != SAMPLING_RATE) {
if (mAudioFormat.mSampleRate != SAMPLE_RATE) {
HBLogError(@"Expected sample rate is %.1f", mAudioFormat.mSampleRate);
return;
}

for (int i = 0; i < NUM_BUFFERS; ++i) {
for (int i = 0; i < NUM_BUF; ++i) {
// Allocate buffer. Size is in bytes.
AudioQueueAllocateBuffer(mRecordState.queue, BUFFER_SIZE, &qref[i]);
AudioQueueEnqueueBuffer(mRecordState.queue, qref[i], 0, NULL);
Expand All @@ -423,7 +447,7 @@ - (void)startRecording {
}

- (void)resetBuffer {
memset(mRecordState.detectBuffer, '\0', DETECTBUFFERLEN);
memset(mRecordState.detectBuffer, '\0', DETECT_BUFLEN);
mRecordState.bufferChanged = YES;
}

Expand All @@ -433,13 +457,13 @@ - (void)stopRecording {
}
mRecordState.isRecording = NO;
AudioQueueStop(mRecordState.queue, true);
for (int i = 0; i < NUM_BUFFERS; i++) {
for (int i = 0; i < NUM_BUF; i++) {
AudioQueueFreeBuffer(mRecordState.queue, mRecordState.buffers[i]);
}
AudioQueueDispose(mRecordState.queue, true);
}

- (float)getNoiseLevel {
- (float)noiseLevel {
return mNoiseLevel;
}

Expand All @@ -452,7 +476,7 @@ - (void)setNoiseLevel:(float)noiseLevel {
kMinNoiseToleranceFactor);
}

- (NSInteger)getPowerMethod {
- (NSInteger)powerMethod {
return mPowerMethod;
}

Expand All @@ -464,7 +488,7 @@ - (void)setPowerMethod:(NSInteger)powerMethod {
powerMeasurementMethod = (int)powerMethod;
}

- (BOOL)getBufferChanged {
- (BOOL)isBufferChanged {
return mRecordState.bufferChanged;
}

Expand All @@ -482,15 +506,17 @@ - (BOOL)isRecording {

@end

#pragma mark - Main Procedure

int main(int argc, const char *argv[]) {
@autoreleasepool {
static DTMFDecoder *gDecoder;
gDecoder = [[DTMFDecoder alloc] init];
gDecoder = [DTMFDecoder sharedDecoder];
[gDecoder startRecording];
printf("DTMF > Press <Ctrl+C> to stop.\n");
while ([gDecoder isRecording]) {
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 1e-2, true);
if ([gDecoder getBufferChanged]) {
if ([gDecoder isBufferChanged]) {
NSLog(@"%@", [gDecoder copyBuffer]);
[gDecoder setBufferChanged:NO];
}
Expand Down

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