LibAudio/Piano: Replace floats with doubles

We should default to double-precision so that clients can make the
choice to use float or double.
This commit is contained in:
William McPherson 2020-02-10 23:45:10 +11:00 committed by Andreas Kling
parent d55d2b2794
commit aa149b9330
Notes: sideshowbarker 2024-07-19 09:28:25 +09:00
5 changed files with 30 additions and 30 deletions

View file

@ -158,10 +158,10 @@ String AudioEngine::set_recorded_sample(const StringView& path)
m_recorded_sample.clear();
m_recorded_sample.resize(wav_buffer->sample_count());
float peak = 0;
double peak = 0;
for (int i = 0; i < wav_buffer->sample_count(); ++i) {
float left_abs = fabs(wav_buffer->samples()[i].left);
float right_abs = fabs(wav_buffer->samples()[i].right);
double left_abs = fabs(wav_buffer->samples()[i].left);
double right_abs = fabs(wav_buffer->samples()[i].right);
if (left_abs > peak)
peak = left_abs;
if (right_abs > peak)
@ -228,8 +228,8 @@ Audio::Sample AudioEngine::recorded_sample(size_t note)
int t = m_pos[note];
if (t >= m_recorded_sample.size())
return 0;
float w_left = m_recorded_sample[t].left;
float w_right = m_recorded_sample[t].right;
double w_left = m_recorded_sample[t].left;
double w_right = m_recorded_sample[t].right;
if (t + 1 < m_recorded_sample.size()) {
double t_fraction = m_pos[note] - t;
w_left += (m_recorded_sample[t + 1].left - m_recorded_sample[t].left) * t_fraction;

View file

@ -46,7 +46,7 @@ WaveEditor::~WaveEditor()
{
}
int WaveEditor::sample_to_y(float percentage) const
int WaveEditor::sample_to_y(double percentage) const
{
double portion_of_half_height = percentage * ((frame_inner_rect().height() - 1) / 2.0);
double y = (frame_inner_rect().height() / 2.0) + portion_of_half_height;

View file

@ -45,7 +45,7 @@ private:
virtual void paint_event(GUI::PaintEvent&) override;
int sample_to_y(float percentage) const;
int sample_to_y(double percentage) const;
AudioEngine& m_audio_engine;
};

View file

@ -43,14 +43,14 @@ struct Sample {
}
// For mono
Sample(float left)
Sample(double left)
: left(left)
, right(left)
{
}
// For stereo
Sample(float left, float right)
Sample(double left, double right)
: left(left)
, right(right)
{
@ -71,7 +71,7 @@ struct Sample {
void scale(int percent)
{
float pct = (float)percent / 100.0;
double pct = (double)percent / 100.0;
left *= pct;
right *= pct;
}
@ -83,8 +83,8 @@ struct Sample {
return *this;
}
float left;
float right;
double left;
double right;
};
// Small helper to resample from one playback rate to another
@ -92,16 +92,16 @@ struct Sample {
// Should do better...
class ResampleHelper {
public:
ResampleHelper(float source, float target);
ResampleHelper(double source, double target);
void process_sample(float sample_l, float sample_r);
bool read_sample(float& next_l, float& next_r);
void process_sample(double sample_l, double sample_r);
bool read_sample(double& next_l, double& next_r);
private:
const float m_ratio;
float m_current_ratio { 0 };
float m_last_sample_l { 0 };
float m_last_sample_r { 0 };
const double m_ratio;
double m_current_ratio { 0 };
double m_last_sample_l { 0 };
double m_last_sample_r { 0 };
};
// A buffer of audio samples, normalized to 44100hz.

View file

@ -182,19 +182,19 @@ bool WavLoader::parse_header()
return true;
}
ResampleHelper::ResampleHelper(float source, float target)
ResampleHelper::ResampleHelper(double source, double target)
: m_ratio(source / target)
{
}
void ResampleHelper::process_sample(float sample_l, float sample_r)
void ResampleHelper::process_sample(double sample_l, double sample_r)
{
m_last_sample_l = sample_l;
m_last_sample_r = sample_r;
m_current_ratio += 1;
}
bool ResampleHelper::read_sample(float& next_l, float& next_r)
bool ResampleHelper::read_sample(double& next_l, double& next_r)
{
if (m_current_ratio > 0) {
m_current_ratio -= m_ratio;
@ -209,8 +209,8 @@ bool ResampleHelper::read_sample(float& next_l, float& next_r)
template<typename SampleReader>
static void read_samples_from_stream(BufferStream& stream, SampleReader read_sample, Vector<Sample>& samples, ResampleHelper& resampler, int num_channels)
{
float norm_l = 0;
float norm_r = 0;
double norm_l = 0;
double norm_r = 0;
switch (num_channels) {
case 1:
@ -245,7 +245,7 @@ static void read_samples_from_stream(BufferStream& stream, SampleReader read_sam
}
}
static float read_norm_sample_24(BufferStream& stream)
static double read_norm_sample_24(BufferStream& stream)
{
u8 byte = 0;
stream >> byte;
@ -259,21 +259,21 @@ static float read_norm_sample_24(BufferStream& stream)
value = sample1 << 8;
value |= (sample2 << 16);
value |= (sample3 << 24);
return float(value) / std::numeric_limits<i32>::max();
return double(value) / std::numeric_limits<i32>::max();
}
static float read_norm_sample_16(BufferStream& stream)
static double read_norm_sample_16(BufferStream& stream)
{
i16 sample = 0;
stream >> sample;
return float(sample) / std::numeric_limits<i16>::max();
return double(sample) / std::numeric_limits<i16>::max();
}
static float read_norm_sample_8(BufferStream& stream)
static double read_norm_sample_8(BufferStream& stream)
{
u8 sample = 0;
stream >> sample;
return float(sample) / std::numeric_limits<u8>::max();
return double(sample) / std::numeric_limits<u8>::max();
}
// ### can't const this because BufferStream is non-const