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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 1.0.0 preview 5" };

  31. 

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

  33. 

  34. struct PresetEntry

  35. {

  36. 	PresetEntry(String name, String data) : m_name(name), m_data(data) {}

  37. 	String m_name;

  38. 	String m_data;

  39. };

  40. 

  41. const int g_num_presets = 4;

  42. 

  43. static const PresetEntry g_presets[g_num_presets] = 

  44. { 

  45. 	{"Factory reset","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"},

  46. 	{"Chipmunk","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"},

  47. 	{"Chipmunk harmonic series","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"},

  48. 	{"Dark noise","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"}

  49. };

  50. 

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

  52. 	int orig_n = n;

  53. 	while (true) {

  54. 		n = orig_n;

  55. 

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

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

  58. 

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

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

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

  62. 		if (n<2) break;

  63. 		if (up) orig_n++;

  64. 		else orig_n--;

  65. 		if (orig_n<4) return 4;

  66. 	};

  67. 	return orig_n;

  68. };

  69. 

  70. int optimizebufsize(int n) {

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

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

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

  74. 	else return n2;

  75. };

  76. 

  77. //==============================================================================

  78. PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()

  79. 	: m_bufferingthread("pspluginprebufferthread")

  80. #ifndef JucePlugin_PreferredChannelConfigurations

  81.      : AudioProcessor (BusesProperties()

  82.                      #if ! JucePlugin_IsMidiEffect

  83.                       #if ! JucePlugin_IsSynth

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

  85.                       #endif

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

  87.                      #endif

  88.                        )

  89. #endif

  90. {

  91. 	

  92.     g_activeprocessors.insert(this);

  93. 	

  94. 	m_recbuffer.setSize(2, 44100);

  95. 	m_recbuffer.clear();

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

  97. 		m_afm->registerBasicFormats();

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

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

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

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

  102. 	

  103. 	m_stretch_source->setOnsetDetection(0.0);

  104. 	m_stretch_source->setLoopingEnabled(true);

  105. 	m_stretch_source->setFFTWindowingType(1);

  106. 	addParameter(new AudioParameterFloat("mainvolume0", "Main volume", -24.0f, 12.0f, -3.0f)); // 0

  107. 	addParameter(new AudioParameterFloat("stretchamount0", "Stretch amount", 

  108. 		NormalisableRange<float>(0.1f, 1024.0f, 0.01f, 0.25),1.0f)); // 1

  109. 	addParameter(new AudioParameterFloat("fftsize0", "FFT size", 0.0f, 1.0f, 0.7f)); // 2

  110. 	addParameter(new AudioParameterFloat("pitchshift0", "Pitch shift", -24.0f, 24.0f, 0.0f)); // 3

  111. 	addParameter(new AudioParameterFloat("freqshift0", "Frequency shift", -1000.0f, 1000.0f, 0.0f)); // 4

  112. 	addParameter(new AudioParameterFloat("playrange_start0", "Sound start", 0.0f, 1.0f, 0.0f)); // 5

  113. 	addParameter(new AudioParameterFloat("playrange_end0", "Sound end", 0.0f, 1.0f, 1.0f)); // 6

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

  115. 	addParameter(new AudioParameterFloat("spread0", "Frequency spread", 0.0f, 1.0f, 0.0f)); // 8

  116. 	addParameter(new AudioParameterFloat("compress0", "Compress", 0.0f, 1.0f, 0.0f)); // 9

  117. 	addParameter(new AudioParameterFloat("loopxfadelen0", "Loop xfade length", 0.0f, 1.0f, 0.01f)); // 10

  118.     auto numhar_convertFrom0To1Func = [](float rangemin, float rangemax, float value)

  119.     {

  120.         return jmap<float>(value, 0.0f, 1.0f, 101.0f, 1.0f);

  121.     };

  122.     auto numhar_convertTo0To1Func = [](float rangemin, float rangemax, float value)

  123.     {

  124.         return jmap<float>(value, 101.0f, 1.0f, 0.0f, 1.0f);

  125.     };

  126.     /*

  127. 	addParameter(new AudioParameterFloat("numharmonics0", "Num harmonics",

  128.                                          NormalisableRange<float>(1.0f, 101.0f,

  129.                                         numhar_convertFrom0To1Func, numhar_convertTo0To1Func), 101.0f)); // 11

  130. 	*/

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

  132. 	addParameter(new AudioParameterFloat("harmonicsfreq0", "Harmonics base freq", 

  133. 		NormalisableRange<float>(1.0f, 5000.0f, 1.00f, 0.5), 128.0f)); // 12

  134. 	addParameter(new AudioParameterFloat("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f)); // 13

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

  136. 	addParameter(new AudioParameterFloat("octavemixm2_0", "2 octaves down level", 0.0f, 1.0f, 0.0f)); // 15

  137. 	addParameter(new AudioParameterFloat("octavemixm1_0", "Octave down level", 0.0f, 1.0f, 0.0f)); // 16

  138. 	addParameter(new AudioParameterFloat("octavemix0_0", "Normal pitch level", 0.0f, 1.0f, 1.0f)); // 17

  139. 	addParameter(new AudioParameterFloat("octavemix1_0", "1 octave up level", 0.0f, 1.0f, 0.0f)); // 18

  140. 	addParameter(new AudioParameterFloat("octavemix15_0", "1 octave and fifth up level", 0.0f, 1.0f, 0.0f)); // 19

  141. 	addParameter(new AudioParameterFloat("octavemix2_0", "2 octaves up level", 0.0f, 1.0f, 0.0f)); // 20

  142. 	addParameter(new AudioParameterFloat("tonalvsnoisebw_0", "Tonal vs Noise BW", 0.74f, 1.0f, 0.74f)); // 21

  143. 	addParameter(new AudioParameterFloat("tonalvsnoisepreserve_0", "Tonal vs Noise preserve", -1.0f, 1.0f, 0.5f)); // 22

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

  145. 	{

  146. 		if (value < 0.5f)

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

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

  149. 	};

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

  151. 	{

  152. 		if (value < 1000.0f)

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

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

  155. 	};

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

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

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

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

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

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

  162. 	addParameter(new AudioParameterFloat("onsetdetect_0", "Onset detection", 0.0f, 1.0f, 0.0f)); // 25

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

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

  165. 	addParameter(m_outchansparam); // 27

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

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

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

  169. 	auto& pars = getParameters();

  170. 	for (const auto& p : pars)

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

  172. 	setPreBufferAmount(2);

  173.     startTimer(1, 50);

  174. }

  175. 

  176. PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()

  177. {

  178. 	g_activeprocessors.erase(this);

  179. 	m_thumb->removeAllChangeListeners();

  180. 	m_thumb = nullptr;

  181. 	m_bufferingthread.stopThread(1000);

  182. }

  183. 

  184. void PaulstretchpluginAudioProcessor::resetParameters()

  185. {

  186. 	ScopedLock locker(m_cs);

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

  188. 	{

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

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

  191. 	}

  192. }

  193. 

  194. void PaulstretchpluginAudioProcessor::setPreBufferAmount(int x)

  195. {

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

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

  198. 	{

  199.         m_use_backgroundbuffering = true;

  200.         m_prebuffer_amount = temp;

  201. 		m_recreate_buffering_source = true;

  202.         ScopedLock locker(m_cs);

  203.         m_prebuffering_inited = false;

  204.         m_cur_num_out_chans = *m_outchansparam;

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

  206.         String err;

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

  208.                   m_cur_num_out_chans, m_curmaxblocksize, err);

  209.         m_prebuffering_inited = true;

  210. 	}

  211. }

  212. 

  213. int PaulstretchpluginAudioProcessor::getPreBufferAmount()

  214. {

  215. 	if (m_use_backgroundbuffering == false)

  216. 		return -1;

  217. 	return m_prebuffer_amount;

  218. }

  219. 

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

  221. {

  222. 	ValueTree paramtree("paulstretch3pluginstate");

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

  224. 	{

  225. 		auto par = getFloatParameter(i);

  226. 		if (par != nullptr)

  227. 		{

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

  229. 		}

  230. 	}

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

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

  233. 	{

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

  235. 	}

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

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

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

  239. 	{

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

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

  242. 	}

  243. 	if (ignoreoptions == false)

  244. 	{

  245. 		if (m_use_backgroundbuffering)

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

  247. 		else

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

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

  250. 	}

  251. 	return paramtree;

  252. }

  253. 

  254. void PaulstretchpluginAudioProcessor::setStateFromTree(ValueTree tree)

  255. {

  256. 	if (tree.isValid())

  257. 	{

  258. 		{

  259. 			ScopedLock locker(m_cs);

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

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

  262. 			{

  263. 				std::vector<SpectrumProcess> order;

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

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

  266. 				{

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

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

  269. 				}

  270. 				m_stretch_source->setSpectrumProcessOrder(order);

  271. 			}

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

  273. 			{

  274. 				auto par = getFloatParameter(i);

  275. 				if (par != nullptr)

  276. 				{

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

  278. 					*par = parval;

  279. 				}

  280. 			}

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

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

  283. 

  284. 		}

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

  286. 		if (prebufamt == -1)

  287. 			m_use_backgroundbuffering = false;

  288. 		else

  289. 			setPreBufferAmount(prebufamt);

  290. 		if (m_load_file_with_state == true)

  291. 		{

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

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

  294. 			{

  295. 				File f(fn);

  296. 				setAudioFile(f);

  297. 			}

  298. 		}

  299. 		m_state_dirty = true;

  300. 	}

  301. }

  302. 

  303. //==============================================================================

  304. const String PaulstretchpluginAudioProcessor::getName() const

  305. {

  306.     return JucePlugin_Name;

  307. }

  308. 

  309. bool PaulstretchpluginAudioProcessor::acceptsMidi() const

  310. {

  311.    #if JucePlugin_WantsMidiInput

  312.     return true;

  313.    #else

  314.     return false;

  315.    #endif

  316. }

  317. 

  318. bool PaulstretchpluginAudioProcessor::producesMidi() const

  319. {

  320.    #if JucePlugin_ProducesMidiOutput

  321.     return true;

  322.    #else

  323.     return false;

  324.    #endif

  325. }

  326. 

  327. bool PaulstretchpluginAudioProcessor::isMidiEffect() const

  328. {

  329.    #if JucePlugin_IsMidiEffect

  330.     return true;

  331.    #else

  332.     return false;

  333.    #endif

  334. }

  335. 

  336. double PaulstretchpluginAudioProcessor::getTailLengthSeconds() const

  337. {

  338. 	return 0.0;

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

  340. }

  341. 

  342. int PaulstretchpluginAudioProcessor::getNumPrograms()

  343. {

  344. 	return g_num_presets;

  345. }

  346. 

  347. int PaulstretchpluginAudioProcessor::getCurrentProgram()

  348. {

  349.     return m_cur_program;

  350. }

  351. 

  352. void PaulstretchpluginAudioProcessor::setCurrentProgram (int index)

  353. {

  354. 	index = jlimit(0, g_num_presets-1, index);

  355. 	m_cur_program = index;

  356. 	bool temp = m_load_file_with_state;

  357. 	m_load_file_with_state = false;

  358. 	MemoryBlock mb;

  359. 	MemoryOutputStream stream(mb, true);

  360. 	if (Base64::convertFromBase64(stream, g_presets[index].m_data)==true)

  361. 	{

  362. 		ValueTree tree = ValueTree::readFromData(mb.getData(), mb.getSize());

  363. 		tree.setProperty("loadfilewithstate", false, nullptr);

  364. 		setStateFromTree(tree);

  365. 	}

  366. 	m_load_file_with_state = temp;

  367. }

  368. 

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

  370. {

  371. 	index = jlimit(0, g_num_presets-1, index);

  372. 	return g_presets[index].m_name;

  373. }

  374. 

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

  376. {

  377. }

  378. 

  379. void PaulstretchpluginAudioProcessor::setFFTSize(double size)

  380. {

  381. 	if (m_prebuffer_amount == 5)

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

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

  384. 	int optim = optimizebufsize(m_fft_size_to_use);

  385. 	m_fft_size_to_use = optim;

  386. 	m_stretch_source->setFFTSize(optim);

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

  388. }

  389. 

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

  391. {

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

  393. 

  394. 	int bufamt = m_bufamounts[m_prebuffer_amount];

  395. 

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

  397. 		m_recreate_buffering_source = true;

  398. 	if (m_recreate_buffering_source == true)

  399. 	{

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

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

  402. 		m_recreate_buffering_source = false;

  403. 	}

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

  405. 		m_bufferingthread.startThread();

  406. 	m_stretch_source->setNumOutChannels(numoutchans);

  407. 	m_stretch_source->setFFTSize(m_fft_size_to_use);

  408. 	m_stretch_source->setProcessParameters(&m_ppar);

  409. 	m_last_outpos_pos = 0.0;

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

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

  412. }

  413. 

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

  415. {

  416. 	ScopedLock locker(m_cs);

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

  418. 	{

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

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

  421. 	}

  422. }

  423. 

  424. double PaulstretchpluginAudioProcessor::getSampleRateChecked()

  425. {

  426. 	if (m_cur_sr < 1.0)

  427. 		return 44100.0;

  428. 	return m_cur_sr;

  429. }

  430. 

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

  432. {

  433. 	ScopedLock locker(m_cs);

  434. 	m_cur_sr = sampleRate;

  435. 	m_curmaxblocksize = samplesPerBlock;

  436. 	m_input_buffer.setSize(2, samplesPerBlock);

  437. 	int numoutchans = *m_outchansparam;

  438. 	if (numoutchans != m_cur_num_out_chans)

  439. 		m_prebuffering_inited = false;

  440. 	if (m_using_memory_buffer == true)

  441. 	{

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

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

  444. 		m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer, 

  445. 			getSampleRateChecked(), 

  446. 			len);

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

  448. 	}

  449. 	if (m_prebuffering_inited == false)

  450. 	{

  451. 		setFFTSize(*getFloatParameter(cpi_fftsize));

  452. 		m_stretch_source->setProcessParameters(&m_ppar);

  453. 		m_stretch_source->setFFTWindowingType(1);

  454. 		String err;

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

  456. 		numoutchans, samplesPerBlock, err);

  457. 		m_cur_num_out_chans = numoutchans;

  458. 		m_prebuffering_inited = true;

  459. 	}

  460. 	else

  461. 	{

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

  463. 	}

  464. }

  465. 

  466. void PaulstretchpluginAudioProcessor::releaseResources()

  467. {

  468. 	//m_control->stopplay();

  469. 	//m_ready_to_play = false;

  470. }

  471. 

  472. #ifndef JucePlugin_PreferredChannelConfigurations

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

  474. {

  475.   #if JucePlugin_IsMidiEffect

  476.     ignoreUnused (layouts);

  477.     return true;

  478.   #else

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

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

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

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

  483.         return false;

  484. 

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

  486.    #if ! JucePlugin_IsSynth

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

  488.         return false;

  489.    #endif

  490. 

  491.     return true;

  492.   #endif

  493. }

  494. #endif

  495. 

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

  497. {

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

  499. 	{

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

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

  502. 		{

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

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

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

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

  507. 		}

  508. 		else

  509. 		{

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

  511. 		}

  512. 	}

  513. }

  514. 

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

  516. {

  517. 	ScopedLock locker(m_cs);

  518. 	AudioPlayHead* phead = getPlayHead();

  519. 	if (phead != nullptr)

  520. 	{

  521. 		phead->getCurrentPosition(m_playposinfo);

  522. 	}

  523. 	ScopedNoDenormals noDenormals;

  524. 	double srtemp = getSampleRate();

  525. 	if (srtemp != m_cur_sr)

  526. 		m_cur_sr = srtemp;

  527.     const int totalNumInputChannels  = getTotalNumInputChannels();

  528.     const int totalNumOutputChannels = getTotalNumOutputChannels();

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

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

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

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

  533. 	if (m_prebuffering_inited == false)

  534. 		return;

  535. 	if (m_is_recording == true)

  536. 	{

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

  538. 			return;

  539. 		int recbuflenframes = m_max_reclen * getSampleRate();

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

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

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

  543. 		return;

  544. 	}

  545. 	jassert(m_buffering_source != nullptr);

  546. 	jassert(m_bufferingthread.isThreadRunning());

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

  548. 	{

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

  550. 		m_last_host_playing = true;

  551. 	}

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

  553. 	{

  554. 		m_last_host_playing = false;

  555. 	}

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

  557. 		return;

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

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

  560. 

  561. 	setFFTSize(*getFloatParameter(cpi_fftsize));

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

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

  564. 	

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

  566.     

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

  568. 	

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

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

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

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

  573. 	

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

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

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

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

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

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

  580. 	

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

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

  583. 	

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

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

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

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

  588. 	double t0 = *getFloatParameter(cpi_soundstart);

  589. 	double t1 = *getFloatParameter(cpi_soundend);

  590. 	if (t0 > t1)

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

  592. 	if (t1 - t0 < 0.001)

  593. 		t1 = t0 + 0.001;

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

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

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

  597. 	m_stretch_source->setProcessParameters(&m_ppar);

  598. 	AudioSourceChannelInfo aif(buffer);

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

  600. 	{

  601. 		m_stretch_source->getNextAudioBlock(aif);

  602. 	}

  603. 	else

  604. 	{

  605. 		m_buffering_source->getNextAudioBlock(aif);

  606. 	}

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

  608. 	{

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

  610. 		{

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

  612. 		}

  613. 	}

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

  615. 	{

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

  617. 		{

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

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

  620. 				++m_abnormal_output_samples;

  621. 		}

  622. 	}

  623. }

  624. 

  625. //==============================================================================

  626. bool PaulstretchpluginAudioProcessor::hasEditor() const

  627. {

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

  629. }

  630. 

  631. AudioProcessorEditor* PaulstretchpluginAudioProcessor::createEditor()

  632. {

  633. 	return new PaulstretchpluginAudioProcessorEditor (*this);

  634. }

  635. 

  636. //==============================================================================

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

  638. {

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

  640. 	MemoryOutputStream stream(destData,true);

  641. 	paramtree.writeToStream(stream);

  642. }

  643. 

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

  645. {

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

  647. 	setStateFromTree(tree);

  648. }

  649. 

  650. void PaulstretchpluginAudioProcessor::setRecordingEnabled(bool b)

  651. {

  652. 	ScopedLock locker(m_cs);

  653. 	int lenbufframes = getSampleRateChecked()*m_max_reclen;

  654. 	if (b == true)

  655. 	{

  656. 		m_using_memory_buffer = true;

  657. 		m_current_file = File();

  658. 		m_recbuffer.setSize(2, m_max_reclen*getSampleRateChecked()+4096,false,false,true);

  659. 		m_recbuffer.clear();

  660. 		m_rec_pos = 0;

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

  662. 		m_is_recording = true;

  663. 	}

  664. 	else

  665. 	{

  666. 		if (m_is_recording == true)

  667. 		{

  668. 			finishRecording(lenbufframes);

  669. 		}

  670. 	}

  671. }

  672. 

  673. double PaulstretchpluginAudioProcessor::getRecordingPositionPercent()

  674. {

  675. 	if (m_is_recording==false)

  676. 		return 0.0;

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

  678. }

  679. 

  680. String PaulstretchpluginAudioProcessor::setAudioFile(File f)

  681. {

  682.     //if (f==File())

  683.     //    return String();

  684.     //if (f==m_current_file && f.getLastModificationTime()==m_current_file_date)

  685.     //    return String();

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

  687. 	if (ai != nullptr)

  688. 	{

  689. 		if (ai->numChannels > 32)

  690. 		{

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

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

  693. 		}

  694. 		if (ai->bitsPerSample>32)

  695. 		{

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

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

  698. 		}

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

  700. 		ScopedLock locker(m_cs);

  701. 		m_stretch_source->setAudioFile(f);

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

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

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

  705. 		m_current_file = f;

  706.         m_current_file_date = m_current_file.getLastModificationTime();

  707. 		m_using_memory_buffer = false;

  708. 		return String();

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

  710. 

  711. 	}

  712. 	

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

  714. }

  715. 

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

  717. {

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

  719. }

  720. 

  721. double PaulstretchpluginAudioProcessor::getPreBufferingPercent()

  722. {

  723. 	if (m_buffering_source==nullptr)

  724. 		return 0.0;

  725. 	return m_buffering_source->getPercentReady();

  726. }

  727. 

  728. void PaulstretchpluginAudioProcessor::timerCallback(int id)

  729. {

  730. 	if (id == 1)

  731. 	{

  732. 		bool capture = getParameter(cpi_capture_enabled);

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

  734. 		{

  735. 			m_max_reclen = *getFloatParameter(cpi_max_capture_len);

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

  737. 		}

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

  739. 		{

  740. 			setRecordingEnabled(true);

  741. 			return;

  742. 		}

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

  744. 		{

  745. 			setRecordingEnabled(false);

  746. 			return;

  747. 		}

  748. 		if (m_cur_num_out_chans != *m_outchansparam)

  749. 		{

  750. 			jassert(m_curmaxblocksize > 0);

  751. 			ScopedLock locker(m_cs);

  752. 			m_prebuffering_inited = false;

  753. 			m_cur_num_out_chans = *m_outchansparam;

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

  755. 			String err;

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

  757. 				m_cur_num_out_chans, m_curmaxblocksize, err);

  758. 			m_prebuffering_inited = true;

  759. 		}

  760. 	}

  761. }

  762. 

  763. void PaulstretchpluginAudioProcessor::finishRecording(int lenrecording)

  764. {

  765. 	m_is_recording = false;

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

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

  768. 	auto ed = dynamic_cast<PaulstretchpluginAudioProcessorEditor*>(getActiveEditor());

  769. 	if (ed)

  770. 	{

  771. 		//ed->setAudioBuffer(&m_recbuffer, getSampleRate(), lenrecording);

  772. 	}

  773. }

  774. 

  775. //==============================================================================

  776. // This creates new instances of the plugin..

  777. AudioProcessor* JUCE_CALLTYPE createPluginFilter()

  778. {

  779.     return new PaulstretchpluginAudioProcessor();

  780. }