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remove warning about signed/unsigned comparison

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Also change n.f to be n.0f as per coding standard
This commit is contained in:
Andrew Meadows 2014-04-23 08:56:53 -07:00
parent da5ac3d664
commit bfe1ee377b
1 changed files with 25 additions and 28 deletions
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53
interface/src/Audio.cpp
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@ -480,7 +480,7 @@ void Audio::handleAudioInput() {
float thisSample = 0;
int samplesOverNoiseGate = 0;
const float NOISE_GATE_HEIGHT = 7.f;
const float NOISE_GATE_HEIGHT = 7.0f;
const int NOISE_GATE_WIDTH = 5;
const int NOISE_GATE_CLOSE_FRAME_DELAY = 5;
const int NOISE_GATE_FRAMES_TO_AVERAGE = 5;
@ -490,7 +490,7 @@ void Audio::handleAudioInput() {
//
// Check clipping, adjust DC offset, and check if should open noise gate
//
float measuredDcOffset = 0.f;
float measuredDcOffset = 0.0f;
// Increment the time since the last clip
if (_timeSinceLastClip >= 0.0f) {
_timeSinceLastClip += (float) NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL / (float) SAMPLE_RATE;
@ -500,7 +500,7 @@ void Audio::handleAudioInput() {
measuredDcOffset += monoAudioSamples[i];
monoAudioSamples[i] -= (int16_t) _dcOffset;
thisSample = fabsf(monoAudioSamples[i]);
if (thisSample >= (32767.f * CLIPPING_THRESHOLD)) {
if (thisSample >= (32767.0f * CLIPPING_THRESHOLD)) {
_timeSinceLastClip = 0.0f;
}
loudness += thisSample;
@ -511,18 +511,18 @@ void Audio::handleAudioInput() {
}
measuredDcOffset /= NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
if (_dcOffset == 0.f) {
if (_dcOffset == 0.0f) {
// On first frame, copy over measured offset
_dcOffset = measuredDcOffset;
} else {
_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.f - DC_OFFSET_AVERAGING) * measuredDcOffset;
_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.0f - DC_OFFSET_AVERAGING) * measuredDcOffset;
}
// Add tone injection if enabled
const float TONE_FREQ = 220.f / SAMPLE_RATE * TWO_PI;
const float QUARTER_VOLUME = 8192.f;
const float TONE_FREQ = 220.0f / SAMPLE_RATE * TWO_PI;
const float QUARTER_VOLUME = 8192.0f;
if (_toneInjectionEnabled) {
loudness = 0.f;
loudness = 0.0f;
for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
monoAudioSamples[i] = QUARTER_VOLUME * sinf(TONE_FREQ * (float)(i + _proceduralEffectSample));
loudness += fabsf(monoAudioSamples[i]);
@ -532,7 +532,7 @@ void Audio::handleAudioInput() {
// If Noise Gate is enabled, check and turn the gate on and off
if (!_toneInjectionEnabled && _noiseGateEnabled) {
float averageOfAllSampleFrames = 0.f;
float averageOfAllSampleFrames = 0.0f;
_noiseSampleFrames[_noiseGateSampleCounter++] = _lastInputLoudness;
if (_noiseGateSampleCounter == NUMBER_OF_NOISE_SAMPLE_FRAMES) {
float smallestSample = FLT_MAX;
@ -660,9 +660,9 @@ void Audio::addReceivedAudioToBuffer(const QByteArray& audioByteArray) {
_stdev.reset();
// Set jitter buffer to be a multiple of the measured standard deviation
const int MAX_JITTER_BUFFER_SAMPLES = _ringBuffer.getSampleCapacity() / 2;
const float NUM_STANDARD_DEVIATIONS = 3.f;
const float NUM_STANDARD_DEVIATIONS = 3.0f;
if (Menu::getInstance()->getAudioJitterBufferSamples() == 0) {
float newJitterBufferSamples = (NUM_STANDARD_DEVIATIONS * _measuredJitter) / 1000.f * SAMPLE_RATE;
float newJitterBufferSamples = (NUM_STANDARD_DEVIATIONS * _measuredJitter) / 1000.0f * SAMPLE_RATE;
setJitterBufferSamples(glm::clamp((int)newJitterBufferSamples, 0, MAX_JITTER_BUFFER_SAMPLES));
}
}
@ -903,10 +903,10 @@ void Audio::toggleAudioSpatialProcessing() {
void Audio::addProceduralSounds(int16_t* monoInput, int numSamples) {
float sample;
const float COLLISION_SOUND_CUTOFF_LEVEL = 0.01f;
const float COLLISION_SOUND_MAX_VOLUME = 1000.f;
const float COLLISION_SOUND_MAX_VOLUME = 1000.0f;
const float UP_MAJOR_FIFTH = powf(1.5f, 4.0f);
const float DOWN_TWO_OCTAVES = 4.f;
const float DOWN_FOUR_OCTAVES = 16.f;
const float DOWN_TWO_OCTAVES = 4.0f;
const float DOWN_FOUR_OCTAVES = 16.0f;
float t;
if (_collisionSoundMagnitude > COLLISION_SOUND_CUTOFF_LEVEL) {
for (int i = 0; i < numSamples; i++) {
@ -936,12 +936,12 @@ void Audio::addProceduralSounds(int16_t* monoInput, int numSamples) {
_proceduralEffectSample += numSamples;
// Add a drum sound
const float MAX_VOLUME = 32000.f;
const float MAX_DURATION = 2.f;
const float MAX_VOLUME = 32000.0f;
const float MAX_DURATION = 2.0f;
const float MIN_AUDIBLE_VOLUME = 0.001f;
const float NOISE_MAGNITUDE = 0.02f;
float frequency = (_drumSoundFrequency / SAMPLE_RATE) * TWO_PI;
if (_drumSoundVolume > 0.f) {
if (_drumSoundVolume > 0.0f) {
for (int i = 0; i < numSamples; i++) {
t = (float) _drumSoundSample + (float) i;
sample = sinf(t * frequency);
@ -961,12 +961,12 @@ void Audio::addProceduralSounds(int16_t* monoInput, int numSamples) {
_localProceduralSamples[i] = glm::clamp(_localProceduralSamples[i] + collisionSample,
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
_drumSoundVolume *= (1.f - _drumSoundDecay);
_drumSoundVolume *= (1.0f - _drumSoundDecay);
}
_drumSoundSample += numSamples;
_drumSoundDuration = glm::clamp(_drumSoundDuration - (AUDIO_CALLBACK_MSECS / 1000.f), 0.f, MAX_DURATION);
if (_drumSoundDuration == 0.f || (_drumSoundVolume < MIN_AUDIBLE_VOLUME)) {
_drumSoundVolume = 0.f;
_drumSoundDuration = glm::clamp(_drumSoundDuration - (AUDIO_CALLBACK_MSECS / 1000.0f), 0.0f, MAX_DURATION);
if (_drumSoundDuration == 0.0f || (_drumSoundVolume < MIN_AUDIBLE_VOLUME)) {
_drumSoundVolume = 0.0f;
}
}
}
@ -999,7 +999,7 @@ void Audio::renderToolBox(int x, int y, bool boxed) {
if (boxed) {
bool isClipping = ((getTimeSinceLastClip() > 0.f) && (getTimeSinceLastClip() < 1.f));
bool isClipping = ((getTimeSinceLastClip() > 0.0f) && (getTimeSinceLastClip() < 1.0f));
const int BOX_LEFT_PADDING = 5;
const int BOX_TOP_PADDING = 10;
const int BOX_WIDTH = 266;
@ -1010,9 +1010,9 @@ void Audio::renderToolBox(int x, int y, bool boxed) {
glBindTexture(GL_TEXTURE_2D, _boxTextureId);
if (isClipping) {
glColor3f(1.f,0.f,0.f);
glColor3f(1.0f, 0.0f, 0.0f);
} else {
glColor3f(.41f,.41f,.41f);
glColor3f(0.41f, 0.41f, 0.41f);
}
glBegin(GL_QUADS);
@ -1089,10 +1089,8 @@ void Audio::addBufferToScope(
// Short int pointer to mapped samples in byte array
int16_t* destination = (int16_t*) byteArray.data();
for (int i = 0; i < NETWORK_SAMPLES_PER_FRAME; i++) {
for (unsigned int i = 0; i < NETWORK_SAMPLES_PER_FRAME; i++) {
sample = (float)source[i * sourceNumberOfChannels + sourceChannel];
if (sample > 0) {
value = (int16_t)(multiplier * logf(sample));
} else if (sample < 0) {
@ -1100,7 +1098,6 @@ void Audio::addBufferToScope(
} else {
value = 0;
}
destination[i + frameOffset] = value;
}
}