view plugin/plugins/SamplePlayer.cpp @ 1196:c7b9c902642f spectrogram-minor-refactor

Fix threshold in spectrogram -- it wasn't working in the last release. There is a new protocol for this. Formerly the threshold parameter had a range from -50dB to 0 with the default at -50, and -50 treated internally as "no threshold". However, there was a hardcoded, hidden internal threshold for spectrogram colour mapping at -80dB with anything below this being rounded to zero. Now the threshold parameter has range -81 to -1 with the default at -80, -81 is treated internally as "no threshold", and there is no hidden internal threshold. So the default behaviour is the same as before, an effective -80dB threshold, but it is now possible to change this in both directions. Sessions reloaded from prior versions may look slightly different because, if the session says there should be no threshold, there will now actually be no threshold instead of having the hidden internal one. Still need to do something in the UI to make it apparent that the -81dB setting removes the threshold entirely. This is at least no worse than the previous, also obscured, magic -50dB setting.
author Chris Cannam
date Mon, 01 Aug 2016 16:21:01 +0100
parents 63b73a21bccd
children 1c9bbbb6116a
line wrap: on
line source
/* -*- 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.
*/

/*
    Based on trivial_sampler from the DSSI distribution
    (by Chris Cannam, public domain).
*/

#include "SamplePlayer.h"
#include "system/System.h"

#include "../api/dssi.h"

#include <cmath>
#include <cstdlib>

#include <QMutexLocker>
#include <QDir>
#include <QFileInfo>

#include <sndfile.h>
#include <samplerate.h>
#include <iostream>

//#define DEBUG_SAMPLE_PLAYER 1

const char *const
SamplePlayer::portNames[PortCount] =
{
    "Output",
    "Tuned (on/off)",
    "Base Pitch (MIDI)",
    "Tuning of A (Hz)",
    "Sustain (on/off)",
    "Release time (s)"
};

const LADSPA_PortDescriptor 
SamplePlayer::ports[PortCount] =
{
    LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO,
    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL,
    LADSPA_PORT_INPUT  | LADSPA_PORT_CONTROL
};

const LADSPA_PortRangeHint 
SamplePlayer::hints[PortCount] =
{
    { 0, 0, 0 },
    { LADSPA_HINT_DEFAULT_MAXIMUM | LADSPA_HINT_INTEGER |
      LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE, 0, 1 },
    { LADSPA_HINT_DEFAULT_MIDDLE | LADSPA_HINT_INTEGER |
      LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE, 0, 120 },
    { LADSPA_HINT_DEFAULT_440 | LADSPA_HINT_LOGARITHMIC |
      LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE, 400, 499 },
    { LADSPA_HINT_DEFAULT_MINIMUM | LADSPA_HINT_INTEGER |
      LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE, 0, 1 },
    { LADSPA_HINT_DEFAULT_MINIMUM | LADSPA_HINT_LOGARITHMIC |
      LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE, 0.001f, 2.0f }
};

const LADSPA_Properties
SamplePlayer::properties = LADSPA_PROPERTY_HARD_RT_CAPABLE;

const LADSPA_Descriptor 
SamplePlayer::ladspaDescriptor =
{
    0, // "Unique" ID
    "sample_player", // Label
    properties,
    "Library Sample Player", // Name
    "Chris Cannam", // Maker
    "GPL", // Copyright
    PortCount,
    ports,
    portNames,
    hints,
    0, // Implementation data
    instantiate,
    connectPort,
    activate,
    run,
    0, // Run adding
    0, // Set run adding gain
    deactivate,
    cleanup
};

const DSSI_Descriptor 
SamplePlayer::dssiDescriptor =
{
    2, // DSSI API version
    &ladspaDescriptor,
    configure,
    getProgram,
    selectProgram,
    getMidiController,
    runSynth,
    0, // Run synth adding
    0, // Run multiple synths
    0, // Run multiple synths adding
    receiveHostDescriptor
};

const DSSI_Host_Descriptor *
SamplePlayer::hostDescriptor = 0;


const DSSI_Descriptor *
SamplePlayer::getDescriptor(unsigned long index)
{
    if (index == 0) return &dssiDescriptor;
    return 0;
}

SamplePlayer::SamplePlayer(int sampleRate) :
    m_output(0),
    m_retune(0),
    m_basePitch(0),
    m_concertA(0),
    m_sustain(0),
    m_release(0),
    m_sampleData(0),
    m_sampleCount(0),
    m_sampleRate(sampleRate),
    m_sampleNo(0),
    m_sampleDir("samples"),
    m_sampleSearchComplete(false),
    m_pendingProgramChange(-1)
{
}

SamplePlayer::~SamplePlayer()
{
    if (m_sampleData) free(m_sampleData);
}
    
LADSPA_Handle
SamplePlayer::instantiate(const LADSPA_Descriptor *, unsigned long rate)
{
    if (!hostDescriptor || !hostDescriptor->request_non_rt_thread) {
	SVDEBUG << "SamplePlayer::instantiate: Host does not provide request_non_rt_thread, not instantiating" << endl;
	return 0;
    }

    SamplePlayer *player = new SamplePlayer(int(rate));
	// std::cerr << "Instantiated sample player " << std::endl;

    if (hostDescriptor->request_non_rt_thread(player, workThreadCallback)) {
	SVDEBUG << "SamplePlayer::instantiate: Host rejected request_non_rt_thread call, not instantiating" << endl;
	delete player;
	return 0;
    }

    return player;
}

void
SamplePlayer::connectPort(LADSPA_Handle handle,
			  unsigned long port, LADSPA_Data *location)
{
    SamplePlayer *player = (SamplePlayer *)handle;

    float **ports[PortCount] = {
	&player->m_output,
	&player->m_retune,
	&player->m_basePitch,
        &player->m_concertA,
	&player->m_sustain,
	&player->m_release
    };

    *ports[port] = (float *)location;
}

void
SamplePlayer::activate(LADSPA_Handle handle)
{
    SamplePlayer *player = (SamplePlayer *)handle;
    QMutexLocker locker(&player->m_mutex);

    player->m_sampleNo = 0;

    for (size_t i = 0; i < Polyphony; ++i) {
	player->m_ons[i] = -1;
	player->m_offs[i] = -1;
	player->m_velocities[i] = 0;
    }
}

void
SamplePlayer::run(LADSPA_Handle handle, unsigned long samples)
{
    runSynth(handle, samples, 0, 0);
}

void
SamplePlayer::deactivate(LADSPA_Handle handle)
{
    activate(handle); // both functions just reset the plugin
}

void
SamplePlayer::cleanup(LADSPA_Handle handle)
{
    delete (SamplePlayer *)handle;
}

char *
SamplePlayer::configure(LADSPA_Handle handle, const char *key, const char *value)
{
    if (key && !strcmp(key, "sampledir")) {

        SamplePlayer *player = (SamplePlayer *)handle;

	QMutexLocker locker(&player->m_mutex);

        if (QFileInfo(value).exists() &&
            QFileInfo(value).isDir()) {

            player->m_sampleDir = value;

            if (player->m_sampleSearchComplete) {
                player->m_sampleSearchComplete = false;
                player->searchSamples();
            }

            return 0;

        } else {
            char *buffer = (char *)malloc(strlen(value) + 80);
            sprintf(buffer, "Sample directory \"%s\" does not exist, leaving unchanged", value);
            return buffer;
        }
    }

    return strdup("Unknown configure key");
}

const DSSI_Program_Descriptor *
SamplePlayer::getProgram(LADSPA_Handle handle, unsigned long program)
{
    SamplePlayer *player = (SamplePlayer *)handle;

    if (!player->m_sampleSearchComplete) {
	QMutexLocker locker(&player->m_mutex);
	if (!player->m_sampleSearchComplete) {
	    player->searchSamples();
	}
    }
    if (program >= player->m_samples.size()) return 0;

    static DSSI_Program_Descriptor descriptor;
    static char name[60];

    strncpy(name, player->m_samples[program].first.toLocal8Bit().data(), 60);
    name[59] = '\0';

    descriptor.Bank = 0;
    descriptor.Program = program;
    descriptor.Name = name;

    return &descriptor;
}

void
SamplePlayer::selectProgram(LADSPA_Handle handle,
			    unsigned long,
			    unsigned long program)
{
    SamplePlayer *player = (SamplePlayer *)handle;
    player->m_pendingProgramChange = (int)program;
}

int
SamplePlayer::getMidiController(LADSPA_Handle, unsigned long port)
{
    int controllers[PortCount] = {
	DSSI_NONE,
	DSSI_CC(12),
	DSSI_CC(13),
	DSSI_CC(64),
	DSSI_CC(72)
    };

    return controllers[port];
}

void
SamplePlayer::runSynth(LADSPA_Handle handle, unsigned long samples,
		       snd_seq_event_t *events, unsigned long eventCount)
{
    SamplePlayer *player = (SamplePlayer *)handle;

    player->runImpl(samples, events, eventCount);
}

void
SamplePlayer::receiveHostDescriptor(const DSSI_Host_Descriptor *descriptor)
{
    hostDescriptor = descriptor;
}

void
SamplePlayer::workThreadCallback(LADSPA_Handle handle)
{
    SamplePlayer *player = (SamplePlayer *)handle;

    if (player->m_pendingProgramChange >= 0) {

#ifdef DEBUG_SAMPLE_PLAYER
	SVDEBUG << "SamplePlayer::workThreadCallback: pending program change " << player->m_pendingProgramChange << endl;
#endif

	player->m_mutex.lock();

	int program = player->m_pendingProgramChange;
	player->m_pendingProgramChange = -1;

	if (!player->m_sampleSearchComplete) {
	    player->searchSamples();
	}
	
	if (program < int(player->m_samples.size())) {
	    QString path = player->m_samples[program].second;
	    QString programName = player->m_samples[program].first;
	    if (programName != player->m_program) {
		player->m_program = programName;
		player->m_mutex.unlock();
		player->loadSampleData(path);
	    } else {
		player->m_mutex.unlock();
	    }
	}
    }

    if (!player->m_sampleSearchComplete) {

	QMutexLocker locker(&player->m_mutex);

	if (!player->m_sampleSearchComplete) {
	    player->searchSamples();
	}
    }
}

void
SamplePlayer::searchSamples()
{
    if (m_sampleSearchComplete) return;

    m_samples.clear();

#ifdef DEBUG_SAMPLE_PLAYER
    SVDEBUG << "SamplePlayer::searchSamples: Directory is \""
	      << m_sampleDir << "\"" << endl;
#endif

    QDir dir(m_sampleDir, "*.wav");
    
    for (unsigned int i = 0; i < dir.count(); ++i) {
        QFileInfo file(dir.filePath(dir[i]));
        if (file.isReadable()) {
            m_samples.push_back(std::pair<QString, QString>
                                (file.baseName(), file.filePath()));
#ifdef DEBUG_SAMPLE_PLAYER
            cerr << "Found: " << dir[i] << endl;
#endif
        }
    }
    
    m_sampleSearchComplete = true;
}

void
SamplePlayer::loadSampleData(QString path)
{
    SF_INFO info;
    SNDFILE *file;
    size_t samples = 0;
    float *tmpFrames, *tmpSamples, *tmpResamples, *tmpOld;
    size_t i;

    info.format = 0;
    file = sf_open(path.toLocal8Bit().data(), SFM_READ, &info);
    if (!file) {
	cerr << "SamplePlayer::loadSampleData: Failed to open file "
		  << path << ": "
		  << sf_strerror(file) << endl;
	return;
    }
    
    samples = info.frames;
    tmpFrames = (float *)malloc(info.frames * info.channels * sizeof(float));
    if (!tmpFrames) return;

    sf_readf_float(file, tmpFrames, info.frames);
    sf_close(file);

    tmpResamples = 0;

    if (info.samplerate != m_sampleRate) {
	
	double ratio = (double)m_sampleRate / (double)info.samplerate;
	size_t target = (size_t)(double(info.frames) * ratio);
	SRC_DATA data;

	tmpResamples = (float *)malloc(target * info.channels * sizeof(float));
	if (!tmpResamples) {
	    free(tmpFrames);
	    return;
	}

	memset(tmpResamples, 0, target * info.channels * sizeof(float));

	data.data_in = tmpFrames;
	data.data_out = tmpResamples;
	data.input_frames = info.frames;
	data.output_frames = target;
	data.src_ratio = ratio;

	if (!src_simple(&data, SRC_SINC_BEST_QUALITY, info.channels)) {
	    free(tmpFrames);
	    tmpFrames = tmpResamples;
	    samples = target;
	} else {
	    free(tmpResamples);
	}
    }

    /* add an extra sample for linear interpolation */
    tmpSamples = (float *)malloc((samples + 1) * sizeof(float));
    if (!tmpSamples) {
	free(tmpFrames);
	return;
    }

    for (i = 0; i < samples; ++i) {
	int j;
	tmpSamples[i] = 0.0f;
	for (j = 0; j < info.channels; ++j) {
	    tmpSamples[i] += tmpFrames[i * info.channels + j];
	}
    }

    free(tmpFrames);

    /* add an extra sample for linear interpolation */
    tmpSamples[samples] = 0.0f;
    
    QMutexLocker locker(&m_mutex);

    tmpOld = m_sampleData;
    m_sampleData = tmpSamples;
    m_sampleCount = samples;

    for (i = 0; i < Polyphony; ++i) {
	m_ons[i] = -1;
	m_offs[i] = -1;
	m_velocities[i] = 0;
    }

    if (tmpOld) free(tmpOld);

    printf("%s: loaded %s (%ld samples from original %ld channels resampled from %ld frames at %ld Hz)\n", "sampler", path.toLocal8Bit().data(), (long)samples, (long)info.channels, (long)info.frames, (long)info.samplerate);
}

void
SamplePlayer::runImpl(unsigned long sampleCount,
		      snd_seq_event_t *events,
		      unsigned long eventCount)
{
    unsigned long pos;
    unsigned long count;
    unsigned long event_pos;
    int i;

    memset(m_output, 0, sampleCount * sizeof(float));

    if (!m_mutex.tryLock()) return;

    if (!m_sampleData || !m_sampleCount) {
	m_sampleNo += sampleCount;
	m_mutex.unlock();
	return;
    }

    for (pos = 0, event_pos = 0; pos < sampleCount; ) {

	while (event_pos < eventCount
	       && pos >= events[event_pos].time.tick) {

	    if (events[event_pos].type == SND_SEQ_EVENT_NOTEON) {
#ifdef DEBUG_SAMPLE_PLAYER
                cerr << "SamplePlayer: found NOTEON at time " 
                          << events[event_pos].time.tick << endl;
#endif
		snd_seq_ev_note_t n = events[event_pos].data.note;
		if (n.velocity > 0) {
		    m_ons[n.note] =
			m_sampleNo + events[event_pos].time.tick;
		    m_offs[n.note] = -1;
		    m_velocities[n.note] = n.velocity;
		} else {
		    if (!m_sustain || (*m_sustain < 0.001)) {
			m_offs[n.note] = 
			    m_sampleNo + events[event_pos].time.tick;
		    }
		}
	    } else if (events[event_pos].type == SND_SEQ_EVENT_NOTEOFF &&
		       (!m_sustain || (*m_sustain < 0.001))) {
#ifdef DEBUG_SAMPLE_PLAYER
                cerr << "SamplePlayer: found NOTEOFF at time " 
                          << events[event_pos].time.tick << endl;
#endif
		snd_seq_ev_note_t n = events[event_pos].data.note;
		m_offs[n.note] = 
		    m_sampleNo + events[event_pos].time.tick;
	    }

	    ++event_pos;
	}

	count = sampleCount - pos;
	if (event_pos < eventCount &&
	    events[event_pos].time.tick < sampleCount) {
	    count = events[event_pos].time.tick - pos;
	}

        int notecount = 0;

	for (i = 0; i < Polyphony; ++i) {
	    if (m_ons[i] >= 0) {
                ++notecount;
		addSample(i, pos, count);
	    }
	}

#ifdef DEBUG_SAMPLE_PLAYER
        cerr << "SamplePlayer: have " << notecount << " note(s) sounding currently" << endl;
#endif

	pos += count;
    }

    m_sampleNo += sampleCount;
    m_mutex.unlock();
}

void
SamplePlayer::addSample(int n, unsigned long pos, unsigned long count)
{
    float ratio = 1.f;
    float gain = 1.f;
    unsigned long i, s;

    if (m_retune && *m_retune) {
        if (m_concertA) {
            ratio *= *m_concertA / 440.f;
        }
	if (m_basePitch && float(n) != *m_basePitch) {
	    ratio *= powf(1.059463094f, float(n) - *m_basePitch);
	}
    }

    if (long(pos + m_sampleNo) < m_ons[n]) return;

    gain = (float)m_velocities[n] / 127.0f;

    for (i = 0, s = pos + m_sampleNo - m_ons[n];
	 i < count;
	 ++i, ++s) {

	float         lgain = gain;
	float         rs = float(s) * ratio;
	unsigned long rsi = lrintf(floorf(rs));

	if (rsi >= m_sampleCount) {
#ifdef DEBUG_SAMPLE_PLAYER
            cerr << "Note " << n << " has run out of samples (were " << m_sampleCount << " available at ratio " << ratio << "), ending" << endl;
#endif
	    m_ons[n] = -1;
	    break;
	}

	if (m_offs[n] >= 0 &&
	    long(pos + i + m_sampleNo) > m_offs[n]) {

	    unsigned long dist =
		pos + i + m_sampleNo - m_offs[n];

	    unsigned long releaseFrames = 200;
	    if (m_release) {
		releaseFrames = long(*m_release * float(m_sampleRate) + 0.0001f);
	    }

	    if (dist > releaseFrames) {
#ifdef DEBUG_SAMPLE_PLAYER
                cerr << "Note " << n << " has expired its release time (" << releaseFrames << " frames), ending" << endl;
#endif
		m_ons[n] = -1;
		break;
	    } else {
		lgain = lgain * (float)(releaseFrames - dist) /
		    (float)releaseFrames;
	    }
	}
	
	float sample = m_sampleData[rsi] +
	    ((m_sampleData[rsi + 1] -
	      m_sampleData[rsi]) *
	     (rs - (float)rsi));

	m_output[pos + i] += lgain * sample;
    }
}