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paulstretchplugin/Source/PluginProcessor.cpp

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  1. /*

  2. Copyright (C) 2006-2011 Nasca Octavian Paul

  3. Author: Nasca Octavian Paul

  4. 

  5. Copyright (C) 2017 Xenakios

  6. 

  7. This program is free software; you can redistribute it and/or modify

  8. it under the terms of version 2 of the GNU General Public License

  9. as published by the Free Software Foundation.

  10. 

  11. This program is distributed in the hope that it will be useful,

  12. but WITHOUT ANY WARRANTY; without even the implied warranty of

  13. MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

  14. GNU General Public License (version 2) for more details.

  15. 

  16. You should have received a copy of the GNU General Public License (version 2)

  17. along with this program; if not, write to the Free Software Foundation,

  18. Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA

  19. */

  20. 

  21. #include "PluginProcessor.h"

  22. #include "PluginEditor.h"

  23. #include <set>

  24. 

  25. #ifdef WIN32

  26. #undef min

  27. #undef max

  28. #endif

  29. 

  30. String g_plugintitle{ "PaulXStretch 1.0.1" };

  31. 

  32. std::set<PaulstretchpluginAudioProcessor*> g_activeprocessors;

  33. 

  34. int get_optimized_updown(int n, bool up) {

  35. 	int orig_n = n;

  36. 	while (true) {

  37. 		n = orig_n;

  38. 

  39. 		while (!(n % 11)) n /= 11;

  40. 		while (!(n % 7)) n /= 7;

  41. 

  42. 		while (!(n % 5)) n /= 5;

  43. 		while (!(n % 3)) n /= 3;

  44. 		while (!(n % 2)) n /= 2;

  45. 		if (n<2) break;

  46. 		if (up) orig_n++;

  47. 		else orig_n--;

  48. 		if (orig_n<4) return 4;

  49. 	};

  50. 	return orig_n;

  51. };

  52. 

  53. int optimizebufsize(int n) {

  54. 	int n1 = get_optimized_updown(n, false);

  55. 	int n2 = get_optimized_updown(n, true);

  56. 	if ((n - n1)<(n2 - n)) return n1;

  57. 	else return n2;

  58. };

  59. 

  60. inline AudioParameterFloat* make_floatpar(String id, String name, float minv, float maxv, float defv, float step, float skew)

  61. {

  62. 	return new AudioParameterFloat(id, name, NormalisableRange<float>(minv, maxv, step, skew), defv);

  63. }

  64. 

  65. //==============================================================================

  66. PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()

  67. 	: m_bufferingthread("pspluginprebufferthread")

  68. #ifndef JucePlugin_PreferredChannelConfigurations

  69.      : AudioProcessor (BusesProperties()

  70.                      #if ! JucePlugin_IsMidiEffect

  71.                       #if ! JucePlugin_IsSynth

  72.                        .withInput  ("Input",  AudioChannelSet::stereo(), true)

  73.                       #endif

  74.                        .withOutput ("Output", AudioChannelSet::stereo(), true)

  75.                      #endif

  76.                        )

  77. #endif

  78. {

  79. 	

  80.     g_activeprocessors.insert(this);

  81. 	m_playposinfo.timeInSeconds = 0.0;

  82. 	m_recbuffer.setSize(2, 44100);

  83. 	m_recbuffer.clear();

  84. 	if (m_afm->getNumKnownFormats()==0)

  85. 		m_afm->registerBasicFormats();

  86. 	m_thumb = std::make_unique<AudioThumbnail>(512, *m_afm, *m_thumbcache);

  87. 	// The default priority of 2 is a bit too low in some cases, it seems...

  88. 	m_thumbcache->getTimeSliceThread().setPriority(3);

  89. 	m_stretch_source = std::make_unique<StretchAudioSource>(2, m_afm);

  90. 	

  91. 	m_stretch_source->setOnsetDetection(0.0);

  92. 	m_stretch_source->setLoopingEnabled(true);

  93. 	m_stretch_source->setFFTWindowingType(1);

  94. 	addParameter(make_floatpar("mainvolume0", "Main volume", -24.0, 12.0, -3.0, 0.1, 1.0)); 

  95. 	addParameter(make_floatpar("stretchamount0", "Stretch amount", 0.1, 1024.0, 2.0, 0.1, 0.25)); 

  96. 	addParameter(make_floatpar("fftsize0", "FFT size", 0.0, 1.0, 0.7, 0.01, 1.0));

  97. 	addParameter(make_floatpar("pitchshift0", "Pitch shift", -24.0f, 24.0f, 0.0f, 0.1,1.0)); // 3

  98. 	addParameter(make_floatpar("freqshift0", "Frequency shift", -1000.0f, 1000.0f, 0.0f, 1.0, 1.0)); // 4

  99. 	addParameter(make_floatpar("playrange_start0", "Sound start", 0.0f, 1.0f, 0.0f, 0.0001,1.0)); // 5

  100. 	addParameter(make_floatpar("playrange_end0", "Sound end", 0.0f, 1.0f, 1.0f, 0.0001,1.0)); // 6

  101. 	addParameter(new AudioParameterBool("freeze0", "Freeze", false)); // 7

  102. 	addParameter(make_floatpar("spread0", "Frequency spread", 0.0f, 1.0f, 0.0f, 0.001,1.0)); // 8

  103. 	addParameter(make_floatpar("compress0", "Compress", 0.0f, 1.0f, 0.0f, 0.001,1.0)); // 9

  104. 	addParameter(make_floatpar("loopxfadelen0", "Loop xfade length", 0.0f, 1.0f, 0.01f, 0.001, 1.0)); // 10

  105.     addParameter(new AudioParameterInt("numharmonics0", "Num harmonics", 1, 100, 10)); // 11

  106. 	addParameter(make_floatpar("harmonicsfreq0", "Harmonics base freq", 1.0, 5000.0, 128.0, 0.1, 0.5));

  107. 	addParameter(make_floatpar("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f, 0.01, 1.0)); // 13

  108. 	addParameter(new AudioParameterBool("harmonicsgauss0", "Gaussian harmonics", false)); // 14

  109. 	addParameter(make_floatpar("octavemixm2_0", "2 octaves down level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 15

  110. 	addParameter(make_floatpar("octavemixm1_0", "Octave down level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 16

  111. 	addParameter(make_floatpar("octavemix0_0", "Normal pitch level", 0.0f, 1.0f, 1.0f, 0.001, 1.0)); // 17

  112. 	addParameter(make_floatpar("octavemix1_0", "1 octave up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 18

  113. 	addParameter(make_floatpar("octavemix15_0", "1 octave and fifth up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 19

  114. 	addParameter(make_floatpar("octavemix2_0", "2 octaves up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 20

  115. 	addParameter(make_floatpar("tonalvsnoisebw_0", "Tonal vs Noise BW", 0.74f, 1.0f, 0.74f, 0.001, 1.0)); // 21

  116. 	addParameter(make_floatpar("tonalvsnoisepreserve_0", "Tonal vs Noise preserve", -1.0f, 1.0f, 0.5f, 0.001, 1.0)); // 22

  117. 	auto filt_convertFrom0To1Func = [](float rangemin, float rangemax, float value) 

  118. 	{

  119. 		if (value < 0.5f)

  120. 			return jmap<float>(value, 0.0f, 0.5f, 20.0f, 1000.0f);

  121. 		return jmap<float>(value, 0.5f, 1.0f, 1000.0f, 20000.0f);

  122. 	};

  123. 	auto filt_convertTo0To1Func = [](float rangemin, float rangemax, float value)

  124. 	{

  125. 		if (value < 1000.0f)

  126. 			return jmap<float>(value, 20.0f, 1000.0f, 0.0f, 0.5f);

  127. 		return jmap<float>(value, 1000.0f, 20000.0f, 0.5f, 1.0f);

  128. 	};

  129. 	addParameter(new AudioParameterFloat("filter_low_0", "Filter low",

  130.                                          NormalisableRange<float>(20.0f, 20000.0f, 

  131. 											 filt_convertFrom0To1Func, filt_convertTo0To1Func), 20.0f)); // 23

  132. 	addParameter(new AudioParameterFloat("filter_high_0", "Filter high",

  133.                                          NormalisableRange<float>(20.0f, 20000.0f, 

  134. 											 filt_convertFrom0To1Func,filt_convertTo0To1Func), 20000.0f));; // 24

  135. 	addParameter(make_floatpar("onsetdetect_0", "Onset detection", 0.0f, 1.0f, 0.0f, 0.01, 1.0)); // 25

  136. 	addParameter(new AudioParameterBool("capture_enabled0", "Capture", false)); // 26

  137. 	m_outchansparam = new AudioParameterInt("numoutchans0", "Num output channels", 2, 8, 2); // 27

  138. 	addParameter(m_outchansparam); // 27

  139. 	addParameter(new AudioParameterBool("pause_enabled0", "Pause", false)); // 28

  140. 	addParameter(new AudioParameterFloat("maxcapturelen_0", "Max capture length", 1.0f, 120.0f, 10.0f)); // 29

  141. 	addParameter(new AudioParameterBool("passthrough0", "Pass input through", false)); // 30

  142. 	addParameter(new AudioParameterBool("markdirty0", "Internal (don't use)", false)); // 31

  143. 	m_inchansparam = new AudioParameterInt("numinchans0", "Num input channels", 2, 8, 2); // 32

  144. 	addParameter(m_inchansparam); // 32

  145. 	auto& pars = getParameters();

  146. 	for (const auto& p : pars)

  147. 		m_reset_pars.push_back(p->getValue());

  148. 	setPreBufferAmount(2);

  149.     startTimer(1, 50);

  150. }

  151. 

  152. PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()

  153. {

  154. 	g_activeprocessors.erase(this);

  155. 	m_thumb->removeAllChangeListeners();

  156. 	m_thumb = nullptr;

  157. 	m_bufferingthread.stopThread(1000);

  158. }

  159. 

  160. void PaulstretchpluginAudioProcessor::resetParameters()

  161. {

  162. 	ScopedLock locker(m_cs);

  163. 	for (int i = 0; i < m_reset_pars.size(); ++i)

  164. 	{

  165. 		if (i!=cpi_main_volume && i!=cpi_passthrough)

  166. 			setParameter(i, m_reset_pars[i]);

  167. 	}

  168. }

  169. 

  170. void PaulstretchpluginAudioProcessor::setPreBufferAmount(int x)

  171. {

  172. 	int temp = jlimit(0, 5, x);

  173. 	if (temp != m_prebuffer_amount || m_use_backgroundbuffering == false)

  174. 	{

  175.         m_use_backgroundbuffering = true;

  176.         m_prebuffer_amount = temp;

  177. 		m_recreate_buffering_source = true;

  178.         ScopedLock locker(m_cs);

  179.         m_prebuffering_inited = false;

  180.         m_cur_num_out_chans = *m_outchansparam;

  181.         //Logger::writeToLog("Switching to use " + String(m_cur_num_out_chans) + " out channels");

  182.         String err;

  183.         startplay({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) },

  184.                   m_cur_num_out_chans, m_curmaxblocksize, err);

  185.         m_prebuffering_inited = true;

  186. 	}

  187. }

  188. 

  189. int PaulstretchpluginAudioProcessor::getPreBufferAmount()

  190. {

  191. 	if (m_use_backgroundbuffering == false)

  192. 		return -1;

  193. 	return m_prebuffer_amount;

  194. }

  195. 

  196. ValueTree PaulstretchpluginAudioProcessor::getStateTree(bool ignoreoptions, bool ignorefile)

  197. {

  198. 	ValueTree paramtree("paulstretch3pluginstate");

  199. 	for (int i = 0; i<getNumParameters(); ++i)

  200. 	{

  201. 		auto par = getFloatParameter(i);

  202. 		if (par != nullptr)

  203. 		{

  204. 			paramtree.setProperty(par->paramID, (double)*par, nullptr);

  205. 		}

  206. 	}

  207. 	paramtree.setProperty(m_outchansparam->paramID, (int)*m_outchansparam, nullptr);

  208. 	if (m_current_file != File() && ignorefile == false)

  209. 	{

  210. 		paramtree.setProperty("importedfile", m_current_file.getFullPathName(), nullptr);

  211. 	}

  212. 	auto specorder = m_stretch_source->getSpectrumProcessOrder();

  213. 	paramtree.setProperty("numspectralstages", (int)specorder.size(), nullptr);

  214. 	for (int i = 0; i < specorder.size(); ++i)

  215. 	{

  216. 		paramtree.setProperty("specorder" + String(i), specorder[i].m_index, nullptr);

  217. 		paramtree.setProperty("specstepenabled" + String(i), specorder[i].m_enabled, nullptr);

  218. 	}

  219. 	if (ignoreoptions == false)

  220. 	{

  221. 		if (m_use_backgroundbuffering)

  222. 			paramtree.setProperty("prebufamount", m_prebuffer_amount, nullptr);

  223. 		else

  224. 			paramtree.setProperty("prebufamount", -1, nullptr);

  225. 		paramtree.setProperty("loadfilewithstate", m_load_file_with_state, nullptr);

  226. 	}

  227. 	return paramtree;

  228. }

  229. 

  230. void PaulstretchpluginAudioProcessor::setStateFromTree(ValueTree tree)

  231. {

  232. 	if (tree.isValid())

  233. 	{

  234. 		{

  235. 			ScopedLock locker(m_cs);

  236. 			m_load_file_with_state = tree.getProperty("loadfilewithstate", true);

  237. 			if (tree.hasProperty("numspectralstages"))

  238. 			{

  239. 				std::vector<SpectrumProcess> order;

  240. 				int ordersize = tree.getProperty("numspectralstages");

  241. 				for (int i = 0; i < ordersize; ++i)

  242. 				{

  243. 					bool step_enabled = tree.getProperty("specstepenabled" + String(i));

  244. 					order.push_back({ (int)tree.getProperty("specorder" + String(i)), step_enabled });

  245. 				}

  246. 				m_stretch_source->setSpectrumProcessOrder(order);

  247. 			}

  248. 			for (int i = 0; i < getNumParameters(); ++i)

  249. 			{

  250. 				auto par = getFloatParameter(i);

  251. 				if (par != nullptr)

  252. 				{

  253. 					double parval = tree.getProperty(par->paramID, (double)*par);

  254. 					*par = parval;

  255. 				}

  256. 			}

  257. 			if (tree.hasProperty(m_outchansparam->paramID))

  258. 				*m_outchansparam = tree.getProperty(m_outchansparam->paramID, 2);

  259. 

  260. 		}

  261. 		int prebufamt = tree.getProperty("prebufamount", 2);

  262. 		if (prebufamt == -1)

  263. 			m_use_backgroundbuffering = false;

  264. 		else

  265. 			setPreBufferAmount(prebufamt);

  266. 		if (m_load_file_with_state == true)

  267. 		{

  268. 			String fn = tree.getProperty("importedfile");

  269. 			if (fn.isEmpty() == false)

  270. 			{

  271. 				File f(fn);

  272. 				setAudioFile(f);

  273. 			}

  274. 		}

  275. 		m_state_dirty = true;

  276. 	}

  277. }

  278. 

  279. //==============================================================================

  280. const String PaulstretchpluginAudioProcessor::getName() const

  281. {

  282.     return JucePlugin_Name;

  283. }

  284. 

  285. bool PaulstretchpluginAudioProcessor::acceptsMidi() const

  286. {

  287.    #if JucePlugin_WantsMidiInput

  288.     return true;

  289.    #else

  290.     return false;

  291.    #endif

  292. }

  293. 

  294. bool PaulstretchpluginAudioProcessor::producesMidi() const

  295. {

  296.    #if JucePlugin_ProducesMidiOutput

  297.     return true;

  298.    #else

  299.     return false;

  300.    #endif

  301. }

  302. 

  303. bool PaulstretchpluginAudioProcessor::isMidiEffect() const

  304. {

  305.    #if JucePlugin_IsMidiEffect

  306.     return true;

  307.    #else

  308.     return false;

  309.    #endif

  310. }

  311. 

  312. double PaulstretchpluginAudioProcessor::getTailLengthSeconds() const

  313. {

  314. 	return 0.0;

  315. 	//return (double)m_bufamounts[m_prebuffer_amount]/getSampleRate();

  316. }

  317. 

  318. int PaulstretchpluginAudioProcessor::getNumPrograms()

  319. {

  320. 	return 1;

  321. }

  322. 

  323. int PaulstretchpluginAudioProcessor::getCurrentProgram()

  324. {

  325.     return 0;

  326. }

  327. 

  328. void PaulstretchpluginAudioProcessor::setCurrentProgram (int index)

  329. {

  330. 	

  331. }

  332. 

  333. const String PaulstretchpluginAudioProcessor::getProgramName (int index)

  334. {

  335. 	return String();

  336. }

  337. 

  338. void PaulstretchpluginAudioProcessor::changeProgramName (int index, const String& newName)

  339. {

  340. }

  341. 

  342. void PaulstretchpluginAudioProcessor::setFFTSize(double size)

  343. {

  344. 	if (m_prebuffer_amount == 5)

  345. 		m_fft_size_to_use = pow(2, 7.0 + size * 14.5);

  346. 	else m_fft_size_to_use = pow(2, 7.0 + size * 10.0); // chicken out from allowing huge FFT sizes if not enough prebuffering

  347. 	int optim = optimizebufsize(m_fft_size_to_use);

  348. 	m_fft_size_to_use = optim;

  349. 	m_stretch_source->setFFTSize(optim);

  350. 	//Logger::writeToLog(String(m_fft_size_to_use));

  351. }

  352. 

  353. void PaulstretchpluginAudioProcessor::startplay(Range<double> playrange, int numoutchans, int maxBlockSize, String& err)

  354. {

  355. 	m_stretch_source->setPlayRange(playrange, true);

  356. 

  357. 	int bufamt = m_bufamounts[m_prebuffer_amount];

  358. 

  359. 	if (m_buffering_source != nullptr && numoutchans != m_buffering_source->getNumberOfChannels())

  360. 		m_recreate_buffering_source = true;

  361. 	if (m_recreate_buffering_source == true)

  362. 	{

  363. 		m_buffering_source = std::make_unique<MyBufferingAudioSource>(m_stretch_source.get(),

  364. 			m_bufferingthread, false, bufamt, numoutchans, false);

  365. 		m_recreate_buffering_source = false;

  366. 	}

  367. 	if (m_bufferingthread.isThreadRunning() == false)

  368. 		m_bufferingthread.startThread();

  369. 	m_stretch_source->setNumOutChannels(numoutchans);

  370. 	m_stretch_source->setFFTSize(m_fft_size_to_use);

  371. 	m_stretch_source->setProcessParameters(&m_ppar);

  372. 	m_last_outpos_pos = 0.0;

  373. 	m_last_in_pos = playrange.getStart()*m_stretch_source->getInfileLengthSeconds();

  374. 	m_buffering_source->prepareToPlay(maxBlockSize, getSampleRateChecked());

  375. }

  376. 

  377. void PaulstretchpluginAudioProcessor::setParameters(const std::vector<double>& pars)

  378. {

  379. 	ScopedLock locker(m_cs);

  380. 	for (int i = 0; i < getNumParameters(); ++i)

  381. 	{

  382. 		if (i<pars.size())

  383. 			setParameter(i, pars[i]);

  384. 	}

  385. }

  386. 

  387. double PaulstretchpluginAudioProcessor::getSampleRateChecked()

  388. {

  389. 	if (m_cur_sr < 1.0)

  390. 		return 44100.0;

  391. 	return m_cur_sr;

  392. }

  393. 

  394. void PaulstretchpluginAudioProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)

  395. {

  396. 	ScopedLock locker(m_cs);

  397. 	m_cur_sr = sampleRate;

  398. 	m_curmaxblocksize = samplesPerBlock;

  399. 	m_input_buffer.setSize(getMainBusNumInputChannels(), samplesPerBlock);

  400. 	int numoutchans = *m_outchansparam;

  401. 	if (numoutchans != m_cur_num_out_chans)

  402. 		m_prebuffering_inited = false;

  403. 	if (m_using_memory_buffer == true)

  404. 	{

  405. 		int len = jlimit(100,m_recbuffer.getNumSamples(), 

  406. 			int(getSampleRateChecked()*(*getFloatParameter(cpi_max_capture_len))));

  407. 		m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer, 

  408. 			getSampleRateChecked(), 

  409. 			len);

  410. 		m_thumb->reset(m_recbuffer.getNumChannels(), sampleRate, len);

  411. 	}

  412. 	if (m_prebuffering_inited == false)

  413. 	{

  414. 		setFFTSize(*getFloatParameter(cpi_fftsize));

  415. 		m_stretch_source->setProcessParameters(&m_ppar);

  416. 		m_stretch_source->setFFTWindowingType(1);

  417. 		String err;

  418. 		startplay({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) },

  419. 		numoutchans, samplesPerBlock, err);

  420. 		m_cur_num_out_chans = numoutchans;

  421. 		m_prebuffering_inited = true;

  422. 	}

  423. 	else

  424. 	{

  425. 		m_buffering_source->prepareToPlay(samplesPerBlock, getSampleRateChecked());

  426. 	}

  427. }

  428. 

  429. void PaulstretchpluginAudioProcessor::releaseResources()

  430. {

  431. 	//m_control->stopplay();

  432. 	//m_ready_to_play = false;

  433. }

  434. 

  435. #ifndef JucePlugin_PreferredChannelConfigurations

  436. bool PaulstretchpluginAudioProcessor::isBusesLayoutSupported (const BusesLayout& layouts) const

  437. {

  438.   #if JucePlugin_IsMidiEffect

  439.     ignoreUnused (layouts);

  440.     return true;

  441.   #else

  442.     // This is the place where you check if the layout is supported.

  443.     // In this template code we only support mono or stereo.

  444.     if (layouts.getMainOutputChannelSet() != AudioChannelSet::mono()

  445.      && layouts.getMainOutputChannelSet() != AudioChannelSet::stereo())

  446.         return false;

  447. 

  448.     // This checks if the input layout matches the output layout

  449.    #if ! JucePlugin_IsSynth

  450.     if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())

  451.         return false;

  452.    #endif

  453. 

  454.     return true;

  455.   #endif

  456. }

  457. #endif

  458. 

  459. void copyAudioBufferWrappingPosition(const AudioBuffer<float>& src, AudioBuffer<float>& dest, int destbufpos, int maxdestpos)

  460. {

  461. 	for (int i = 0; i < dest.getNumChannels(); ++i)

  462. 	{

  463. 		int channel_to_copy = i % src.getNumChannels();

  464. 		if (destbufpos + src.getNumSamples() > maxdestpos)

  465. 		{

  466. 			int wrappos = (destbufpos + src.getNumSamples()) % maxdestpos;

  467. 			int partial_len = src.getNumSamples() - wrappos;

  468. 			dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, partial_len);

  469. 			dest.copyFrom(channel_to_copy, partial_len, src, channel_to_copy, 0, wrappos);

  470. 		}

  471. 		else

  472. 		{

  473. 			dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, src.getNumSamples());

  474. 		}

  475. 	}

  476. }

  477. 

  478. void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)

  479. {

  480. 	ScopedLock locker(m_cs);

  481. 	AudioPlayHead* phead = getPlayHead();

  482. 	if (phead != nullptr)

  483. 	{

  484. 		phead->getCurrentPosition(m_playposinfo);

  485. 	}

  486. 	else

  487. 		m_playposinfo.isPlaying = false;

  488. 	ScopedNoDenormals noDenormals;

  489. 	double srtemp = getSampleRate();

  490. 	if (srtemp != m_cur_sr)

  491. 		m_cur_sr = srtemp;

  492.     const int totalNumInputChannels  = getTotalNumInputChannels();

  493.     const int totalNumOutputChannels = getTotalNumOutputChannels();

  494. 	for (int i = 0; i < totalNumInputChannels; ++i)

  495. 		m_input_buffer.copyFrom(i, 0, buffer, i, 0, buffer.getNumSamples());

  496.     for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)

  497.         buffer.clear (i, 0, buffer.getNumSamples());

  498. 	if (m_prebuffering_inited == false)

  499. 		return;

  500. 	if (m_is_recording == true)

  501. 	{

  502. 		if (m_playposinfo.isPlaying == false && m_capture_when_host_plays == true)

  503. 			return;

  504. 		int recbuflenframes = m_max_reclen * getSampleRate();

  505. 		copyAudioBufferWrappingPosition(buffer, m_recbuffer, m_rec_pos, recbuflenframes);

  506. 		m_thumb->addBlock(m_rec_pos, buffer, 0, buffer.getNumSamples());

  507. 		m_rec_pos = (m_rec_pos + buffer.getNumSamples()) % recbuflenframes;

  508. 		return;

  509. 	}

  510. 	jassert(m_buffering_source != nullptr);

  511. 	jassert(m_bufferingthread.isThreadRunning());

  512. 	if (m_last_host_playing == false && m_playposinfo.isPlaying)

  513. 	{

  514. 		m_stretch_source->seekPercent(*getFloatParameter(cpi_soundstart));

  515. 		m_last_host_playing = true;

  516. 	}

  517. 	else if (m_last_host_playing == true && m_playposinfo.isPlaying == false)

  518. 	{

  519. 		m_last_host_playing = false;

  520. 	}

  521. 	if (m_play_when_host_plays == true && m_playposinfo.isPlaying == false)

  522. 		return;

  523. 	m_stretch_source->setMainVolume(*getFloatParameter(cpi_main_volume));

  524. 	m_stretch_source->setRate(*getFloatParameter(cpi_stretchamount));

  525. 

  526. 	setFFTSize(*getFloatParameter(cpi_fftsize));

  527. 	m_ppar.pitch_shift.cents = *getFloatParameter(cpi_pitchshift) * 100.0;

  528. 	m_ppar.freq_shift.Hz = *getFloatParameter(cpi_frequencyshift);

  529. 	

  530. 	m_ppar.spread.bandwidth = *getFloatParameter(cpi_spreadamount);

  531.     

  532.     m_ppar.compressor.power = *getFloatParameter(cpi_compress);

  533. 	

  534. 	m_ppar.harmonics.nharmonics = *getIntParameter(cpi_numharmonics);

  535. 	m_ppar.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);

  536.     m_ppar.harmonics.bandwidth = *getFloatParameter(cpi_harmonicsbw);

  537.     m_ppar.harmonics.gauss = getParameter(cpi_harmonicsgauss);

  538. 	

  539. 	m_ppar.octave.om2 = *getFloatParameter(cpi_octavesm2);

  540. 	m_ppar.octave.om1 = *getFloatParameter(cpi_octavesm1);

  541. 	m_ppar.octave.o0 = *getFloatParameter(cpi_octaves0);

  542. 	m_ppar.octave.o1 = *getFloatParameter(cpi_octaves1);

  543. 	m_ppar.octave.o15 = *getFloatParameter(cpi_octaves15);

  544. 	m_ppar.octave.o2 = *getFloatParameter(cpi_octaves2);

  545. 	

  546. 	m_ppar.filter.low = *getFloatParameter(cpi_filter_low);

  547. 	m_ppar.filter.high = *getFloatParameter(cpi_filter_high);

  548. 	

  549. 	m_ppar.tonal_vs_noise.bandwidth = *getFloatParameter(cpi_tonalvsnoisebw);

  550. 	m_ppar.tonal_vs_noise.preserve = *getFloatParameter(cpi_tonalvsnoisepreserve);

  551. 	m_stretch_source->setOnsetDetection(*getFloatParameter(cpi_onsetdetection));

  552. 	m_stretch_source->setLoopXFadeLength(*getFloatParameter(cpi_loopxfadelen));

  553. 	double t0 = *getFloatParameter(cpi_soundstart);

  554. 	double t1 = *getFloatParameter(cpi_soundend);

  555. 	if (t0 > t1)

  556. 		std::swap(t0, t1);

  557. 	if (t1 - t0 < 0.001)

  558. 		t1 = t0 + 0.001;

  559. 	m_stretch_source->setPlayRange({ t0,t1 }, true);

  560. 	m_stretch_source->setFreezing(getParameter(cpi_freeze));

  561. 	m_stretch_source->setPaused(getParameter(cpi_pause_enabled));

  562. 	m_stretch_source->setProcessParameters(&m_ppar);

  563. 	AudioSourceChannelInfo aif(buffer);

  564. 	if (isNonRealtime() || m_use_backgroundbuffering == false)

  565. 	{

  566. 		m_stretch_source->getNextAudioBlock(aif);

  567. 	}

  568. 	else

  569. 	{

  570. 		m_buffering_source->getNextAudioBlock(aif);

  571. 	}

  572. 	if (getParameter(cpi_passthrough) > 0.5f)

  573. 	{

  574. 		for (int i = 0; i < totalNumInputChannels; ++i)

  575. 		{

  576. 			buffer.addFrom(i, 0, m_input_buffer, i, 0, buffer.getNumSamples());

  577. 		}

  578. 	}

  579. 	for (int i = 0; i < buffer.getNumChannels(); ++i)

  580. 	{

  581. 		for (int j = 0; j < buffer.getNumSamples(); ++j)

  582. 		{

  583. 			float sample = buffer.getSample(i,j);

  584. 			if (std::isnan(sample) || std::isinf(sample))

  585. 				++m_abnormal_output_samples;

  586. 		}

  587. 	}

  588. }

  589. 

  590. //==============================================================================

  591. bool PaulstretchpluginAudioProcessor::hasEditor() const

  592. {

  593.     return true; // (change this to false if you choose to not supply an editor)

  594. }

  595. 

  596. AudioProcessorEditor* PaulstretchpluginAudioProcessor::createEditor()

  597. {

  598. 	return new PaulstretchpluginAudioProcessorEditor (*this);

  599. }

  600. 

  601. //==============================================================================

  602. void PaulstretchpluginAudioProcessor::getStateInformation (MemoryBlock& destData)

  603. {

  604. 	ValueTree paramtree = getStateTree(false,false);

  605. 	MemoryOutputStream stream(destData,true);

  606. 	paramtree.writeToStream(stream);

  607. }

  608. 

  609. void PaulstretchpluginAudioProcessor::setStateInformation (const void* data, int sizeInBytes)

  610. {

  611. 	ValueTree tree = ValueTree::readFromData(data, sizeInBytes);

  612. 	setStateFromTree(tree);

  613. }

  614. 

  615. void PaulstretchpluginAudioProcessor::setDirty()

  616. {

  617. 	*getBoolParameter(cpi_markdirty) = !(*getBoolParameter(cpi_markdirty));

  618. }

  619. 

  620. void PaulstretchpluginAudioProcessor::setRecordingEnabled(bool b)

  621. {

  622. 	ScopedLock locker(m_cs);

  623. 	int lenbufframes = getSampleRateChecked()*m_max_reclen;

  624. 	if (b == true)

  625. 	{

  626. 		m_using_memory_buffer = true;

  627. 		m_current_file = File();

  628. 		m_recbuffer.setSize(getMainBusNumInputChannels(), m_max_reclen*getSampleRateChecked()+4096,false,false,true);

  629. 		m_recbuffer.clear();

  630. 		m_rec_pos = 0;

  631. 		m_thumb->reset(m_recbuffer.getNumChannels(), getSampleRateChecked(), lenbufframes);

  632. 		m_is_recording = true;

  633. 	}

  634. 	else

  635. 	{

  636. 		if (m_is_recording == true)

  637. 		{

  638. 			finishRecording(lenbufframes);

  639. 		}

  640. 	}

  641. }

  642. 

  643. double PaulstretchpluginAudioProcessor::getRecordingPositionPercent()

  644. {

  645. 	if (m_is_recording==false)

  646. 		return 0.0;

  647. 	return 1.0 / m_recbuffer.getNumSamples()*m_rec_pos;

  648. }

  649. 

  650. String PaulstretchpluginAudioProcessor::setAudioFile(File f)

  651. {

  652.     auto ai = unique_from_raw(m_afm->createReaderFor(f));

  653. 	if (ai != nullptr)

  654. 	{

  655. 		if (ai->numChannels > 32)

  656. 		{

  657. 			//MessageManager::callAsync([cb, file]() { cb("Too many channels in file " + file.getFullPathName()); });

  658. 			return "Too many channels in file "+f.getFullPathName();

  659. 		}

  660. 		if (ai->bitsPerSample>32)

  661. 		{

  662. 			//MessageManager::callAsync([cb, file]() { cb("Too high bit depth in file " + file.getFullPathName()); });

  663. 			return "Too high bit depth in file " + f.getFullPathName();

  664. 		}

  665. 		m_thumb->setSource(new FileInputSource(f));

  666. 		ScopedLock locker(m_cs);

  667. 		m_stretch_source->setAudioFile(f);

  668. 		//Range<double> currange{ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) };

  669. 		//if (currange.contains(m_stretch_source->getInfilePositionPercent())==false)

  670. 			m_stretch_source->seekPercent(*getFloatParameter(cpi_soundstart));

  671. 		m_current_file = f;

  672.         m_current_file_date = m_current_file.getLastModificationTime();

  673. 		m_using_memory_buffer = false;

  674. 		setDirty();

  675. 		return String();

  676. 		//MessageManager::callAsync([cb, file]() { cb(String()); });

  677. 

  678. 	}

  679. 	

  680. 	return "Could not open file " + f.getFullPathName();

  681. }

  682. 

  683. Range<double> PaulstretchpluginAudioProcessor::getTimeSelection()

  684. {

  685. 	return { *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) };

  686. }

  687. 

  688. double PaulstretchpluginAudioProcessor::getPreBufferingPercent()

  689. {

  690. 	if (m_buffering_source==nullptr)

  691. 		return 0.0;

  692. 	return m_buffering_source->getPercentReady();

  693. }

  694. 

  695. void PaulstretchpluginAudioProcessor::timerCallback(int id)

  696. {

  697. 	if (id == 1)

  698. 	{

  699. 		bool capture = getParameter(cpi_capture_enabled);

  700. 		if (capture == false && m_max_reclen != *getFloatParameter(cpi_max_capture_len))

  701. 		{

  702. 			m_max_reclen = *getFloatParameter(cpi_max_capture_len);

  703. 			//Logger::writeToLog("Changing max capture len to " + String(m_max_reclen));

  704. 		}

  705. 		if (capture == true && m_is_recording == false)

  706. 		{

  707. 			setRecordingEnabled(true);

  708. 			return;

  709. 		}

  710. 		if (capture == false && m_is_recording == true)

  711. 		{

  712. 			setRecordingEnabled(false);

  713. 			return;

  714. 		}

  715. 		if (m_cur_num_out_chans != *m_outchansparam)

  716. 		{

  717. 			jassert(m_curmaxblocksize > 0);

  718. 			ScopedLock locker(m_cs);

  719. 			m_prebuffering_inited = false;

  720. 			m_cur_num_out_chans = *m_outchansparam;

  721. 			//Logger::writeToLog("Switching to use " + String(m_cur_num_out_chans) + " out channels");

  722. 			String err;

  723. 			startplay({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) },

  724. 				m_cur_num_out_chans, m_curmaxblocksize, err);

  725. 			m_prebuffering_inited = true;

  726. 		}

  727. 	}

  728. }

  729. 

  730. void PaulstretchpluginAudioProcessor::finishRecording(int lenrecording)

  731. {

  732. 	m_is_recording = false;

  733. 	m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer, getSampleRateChecked(), lenrecording);

  734. 	m_stretch_source->setPlayRange({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) }, true);

  735. }

  736. 

  737. AudioProcessor* JUCE_CALLTYPE createPluginFilter()

  738. {

  739.     return new PaulstretchpluginAudioProcessor();

  740. }