view audio/ContinuousSynth.cpp @ 725:16f6737fa557 spectrogram-export

Rework OSC handler so as to consume all available messages rather than having to wait for the timeout in between them. Pause to process events, and also wait for file loads and transforms to complete. (Should only certain kinds of OSC command wait for transforms?)

/* -*- 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.
*/

#include "ContinuousSynth.h"

#include "base/Debug.h"
#include "system/System.h"

#include <cmath>

ContinuousSynth::ContinuousSynth(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize, int waveType) :
    m_channels(channels),
    m_sampleRate(sampleRate),
    m_blockSize(blockSize),
    m_prevF0(-1.0),
    m_phase(0.0),
    m_wavetype(waveType) // 0: 3 sinusoids, 1: 1 sinusoid, 2: sawtooth, 3: square
{
}

ContinuousSynth::~ContinuousSynth()
{
}

void
ContinuousSynth::reset()
{
    m_phase = 0;
}

void
ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0f)
{
    double f0(f0f);
    if (f0 == 0.0) f0 = m_prevF0;

    bool wasOn = (m_prevF0 > 0.0);
    bool nowOn = (f0 > 0.0);

    if (!nowOn && !wasOn) {
        m_phase = 0;
        return;
    }

    sv_frame_t fadeLength = 100;

    float *levels = new float[m_channels];
    
    for (int c = 0; c < m_channels; ++c) {
        levels[c] = gain * 0.5f; // scale gain otherwise too loud compared to source
    }
    if (pan != 0.0 && m_channels == 2) {
        levels[0] *= 1.0f - pan;
        levels[1] *= pan + 1.0f;
    }

//    cerr << "ContinuousSynth::mix: f0 = " << f0 << " (from " << m_prevF0 << "), phase = " << m_phase << endl;

    for (sv_frame_t i = 0; i < m_blockSize; ++i) {

        double fHere = (nowOn ? f0 : m_prevF0);

        if (wasOn && nowOn && (f0 != m_prevF0) && (i < fadeLength)) {
            // interpolate the frequency shift
            fHere = m_prevF0 + ((f0 - m_prevF0) * double(i)) / double(fadeLength);
        }

        double phasor = (fHere * 2 * M_PI) / m_sampleRate;
    
        m_phase = m_phase + phasor;

        int harmonics = int((m_sampleRate / 4) / fHere - 1);
        if (harmonics < 1) harmonics = 1;

        switch (m_wavetype) {
        case 1:
            harmonics = 1;
            break;
        case 2:
            break;
        case 3:
            break;
        default:
            harmonics = 3;
            break;
        }

        for (int h = 0; h < harmonics; ++h) {

            double v = 0;
            double hn = 0;
            double hp = 0;

            switch (m_wavetype) {
            case 1: // single sinusoid
                v = sin(m_phase);
                break;
            case 2: // sawtooth
                if (h != 0) {
                    hn = h + 1;
                    hp = m_phase * hn;
                    v = -(1.0 / M_PI) * sin(hp) / hn;
                } else {
                    v = 0.5;
                }
                break;
            case 3: // square
                hn = h*2 + 1;
                hp = m_phase * hn;
                v = sin(hp) / hn;
                break;
            default: // 3 sinusoids
                hn = h + 1;
                hp = m_phase * hn;
                v = sin(hp) / hn;
                break;
            }

            if (!wasOn && i < fadeLength) {
                // fade in
                v = v * (double(i) / double(fadeLength));
            } else if (!nowOn) {
                // fade out
                if (i > fadeLength) v = 0;
                else v = v * (1.0 - (double(i) / double(fadeLength)));
            }

            for (int c = 0; c < m_channels; ++c) {
                toBuffers[c][i] += float(levels[c] * v);
            }
        }
    }    

    m_prevF0 = f0;

    delete[] levels;
}