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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
/*
Sonic Visualiser
An audio file viewer and annotation editor.
Centre for Digital Music, Queen Mary, University of London.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2 of the
License, or (at your option) any later version. See the file
COPYING included with this distribution for more information.
*/
/*
This is a modified version of a source file from the
Rubber Band audio timestretcher library.
This file copyright 2007 Chris Cannam.
*/
#include "Resampler.h"
#include <cstdlib>
#include <cmath>
#include <iostream>
#include <samplerate.h>
#include "Debug.h"
class Resampler::D
{
public:
D(Quality quality, int channels, sv_frame_t chunkSize);
~D();
sv_frame_t resample(float **in, float **out,
sv_frame_t incount, double ratio,
bool final);
sv_frame_t resampleInterleaved(float *in, float *out,
sv_frame_t incount, double ratio,
bool final);
void reset();
protected:
SRC_STATE *m_src;
float *m_iin;
float *m_iout;
int m_channels;
sv_frame_t m_iinsize;
sv_frame_t m_ioutsize;
};
Resampler::D::D(Quality quality, int channels, sv_frame_t chunkSize) :
m_src(0),
m_iin(0),
m_iout(0),
m_channels(channels),
m_iinsize(0),
m_ioutsize(0)
{
int err = 0;
m_src = src_new(quality == Best ? SRC_SINC_BEST_QUALITY :
quality == Fastest ? SRC_LINEAR :
SRC_SINC_FASTEST,
channels, &err);
//!!! check err, throw
if (chunkSize > 0 && m_channels > 1) {
//!!! alignment?
m_iinsize = chunkSize * m_channels;
m_ioutsize = chunkSize * m_channels * 2;
m_iin = (float *)malloc(m_iinsize * sizeof(float));
m_iout = (float *)malloc(m_ioutsize * sizeof(float));
}
}
Resampler::D::~D()
{
src_delete(m_src);
if (m_iinsize > 0) {
free(m_iin);
}
if (m_ioutsize > 0) {
free(m_iout);
}
}
sv_frame_t
Resampler::D::resample(float **in, float **out,
sv_frame_t incount, double ratio,
bool final)
{
if (m_channels == 1) {
return resampleInterleaved(*in, *out, incount, ratio, final);
}
sv_frame_t outcount = lrint(ceil(double(incount) * ratio));
if (incount * m_channels > m_iinsize) {
m_iinsize = incount * m_channels;
m_iin = (float *)realloc(m_iin, m_iinsize * sizeof(float));
}
if (outcount * m_channels > m_ioutsize) {
m_ioutsize = outcount * m_channels;
m_iout = (float *)realloc(m_iout, m_ioutsize * sizeof(float));
}
for (sv_frame_t i = 0; i < incount; ++i) {
for (int c = 0; c < m_channels; ++c) {
m_iin[i * m_channels + c] = in[c][i];
}
}
sv_frame_t gen = resampleInterleaved(m_iin, m_iout, incount, ratio, final);
for (sv_frame_t i = 0; i < gen; ++i) {
for (int c = 0; c < m_channels; ++c) {
out[c][i] = m_iout[i * m_channels + c];
}
}
return gen;
}
sv_frame_t
Resampler::D::resampleInterleaved(float *in, float *out,
sv_frame_t incount, double ratio,
bool final)
{
SRC_DATA data;
sv_frame_t outcount = lrint(ceil(double(incount) * ratio));
data.data_in = in;
data.data_out = out;
data.input_frames = incount;
data.output_frames = outcount;
data.src_ratio = ratio;
data.end_of_input = (final ? 1 : 0);
int err = src_process(m_src, &data);
if (err) {
cerr << "Resampler: ERROR: src_process returned error: " <<
src_strerror(err) << endl;
return 0;
}
if (data.input_frames_used != incount) {
cerr << "Resampler: NOTE: input_frames_used == " << data.input_frames_used << " (while incount = " << incount << ")" << endl;
}
return data.output_frames_gen;
}
void
Resampler::D::reset()
{
src_reset(m_src);
}
Resampler::Resampler(Quality quality, int channels, sv_frame_t chunkSize)
{
m_d = new D(quality, channels, chunkSize);
}
Resampler::~Resampler()
{
delete m_d;
}
sv_frame_t
Resampler::resample(float **in, float **out,
sv_frame_t incount, double ratio,
bool final)
{
return m_d->resample(in, out, incount, ratio, final);
}
sv_frame_t
Resampler::resampleInterleaved(float *in, float *out,
sv_frame_t incount, double ratio,
bool final)
{
return m_d->resampleInterleaved(in, out, incount, ratio, final);
}
void
Resampler::reset()
{
m_d->reset();
}
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