diff --git a/plugins/community/repos/Bidoo/Makefile b/plugins/community/repos/Bidoo/Makefile index 09c71872..53d77d15 100644 --- a/plugins/community/repos/Bidoo/Makefile +++ b/plugins/community/repos/Bidoo/Makefile @@ -1,15 +1,10 @@ RACK_DIR ?= ../.. SLUG = Bidoo -VERSION = 0.6.7 +VERSION = 0.6.10 DISTRIBUTABLES += $(wildcard LICENSE*) res -FLAGS += -DUSE_KISS_FFT -Idep/include -I./src/dep/audiofile -I./src/dep/filters -I./src/dep/freeverb \ - -I./src/dep/gist/libs/kiss_fft130 -I./src/dep/gist/src -I./src/dep/minimp3\ - -I./src/dep/gist/src/mfcc -I./src/dep/gist/src/core -I./src/dep/gist/src/fft \ - -I./src/dep/gist/src/onset-detection-functions -I./src/dep/gist/src/pitch +FLAGS += -Idep/include -I./src/dep/dr_wav -I./src/dep/filters -I./src/dep/freeverb -I./src/dep/minimp3 -SOURCES = $(wildcard src/*.cpp src/dep/audiofile/*cpp src/dep/filters/*cpp src/dep/freeverb/*cpp src/dep/gist/src/*cpp \ - src/dep/gist/libs/kiss_fft130/*c src/dep/gist/src/mfcc/*cpp src/dep/gist/src/core/*cpp src/dep/gist/src/fft/*cpp \ - src/dep/gist/src/onset-detection-functions/*cpp src/dep/gist/src/pitch/*cpp) +SOURCES = $(wildcard src/*.cpp src/dep/filters/*cpp src/dep/freeverb/*cpp) include $(RACK_DIR)/plugin.mk diff --git a/plugins/community/repos/Bidoo/README.md b/plugins/community/repos/Bidoo/README.md index deac72b8..15244565 100644 --- a/plugins/community/repos/Bidoo/README.md +++ b/plugins/community/repos/Bidoo/README.md @@ -1,7 +1,7 @@ # Bidoo's plugins for [VCVRack](https://vcvrack.com) -![Version](https://img.shields.io/badge/version-0.6.6-green.svg?style=flat-square) +![Version](https://img.shields.io/badge/version-0.6.10-green.svg?style=flat-square) ![License](https://img.shields.io/badge/license-BSD3-blue.svg?style=flat-square) ![Language](https://img.shields.io/badge/language-C++-yellow.svg?style=flat-square) @@ -13,6 +13,20 @@ You can find information on that plugins pack in the [wiki](https://github.com/s ## Last changes +21/08/2018 => 0.6.10 + +rabBIT redesign + +20/08/2018 => 0.6.9 + +rabBIT is a 8 bit reducer/reverser + +09/07/2018 + +Changed the way wav files are loaded and saved => OUAIve and cANARd. Changed the way onsets are detected in cANARd. Fix play mode saving on close for OUAIve. + +This version is compliant with the last version I have of Rack SDK so maybe my pack will be available thru Rack again in 0.6.2. + 13/05/2018 => 0.6.6 antN goes away from mpg123 and is based now on minimp3 so maybe my pack will be available thru Rack again. diff --git a/plugins/community/repos/Bidoo/make.objects b/plugins/community/repos/Bidoo/make.objects index 64ec7bee..5c9ce177 100644 --- a/plugins/community/repos/Bidoo/make.objects +++ b/plugins/community/repos/Bidoo/make.objects @@ -4,41 +4,39 @@ ALL_OBJ= \ src/BAR.o \ src/Bidoo.o \ src/BORDL.o \ + src/CANARD.o \ src/CHUTE.o \ src/CLACOS.o \ + src/DFUZE.o \ src/dep/filters/biquad.o \ src/dep/filters/smbPitchShift.o \ src/DTROY.o \ src/DUKE.o \ src/FORK.o \ - src/HORUS.o \ src/LATE.o \ src/LIMBO.o \ src/LOURDE.o \ src/MOIRE.o \ src/MU.o \ + src/OUAIVE.o \ src/PERCO.o \ + src/RABBIT.o \ src/SIGMA.o \ src/TIARE.o \ src/TOCANTE.o \ src/VOID.o \ src/ZINC.o \ - src/dep/kiss_fft130/kiss_fft.o + src/dep/freeverb/allpass.o \ + src/dep/freeverb/comb.o \ + src/dep/freeverb/revmodel.o # src/dep/minimp3/minimp3_test.o -# already used in AS modules -DUPLICATE_OBJ__REMOVED__= \ - src/DFUZE.o \ - src/dep/freeverb/allpass.o \ - src/dep/freeverb/comb.o \ - src/dep/freeverb/revmodel.o +# src/dep/kiss_fft130/kiss_fft.o GPL_OBJ__REMOVED__= \ - src/CANARD.o \ - src/OUAIVE.o \ src/dep/audiofile/AudioFile.o \ src/dep/gist/src/core/CoreFrequencyDomainFeatures.o \ src/dep/gist/src/core/CoreTimeDomainFeatures.o \ diff --git a/plugins/community/repos/Bidoo/res/HORUS.svg b/plugins/community/repos/Bidoo/res/HORUS.svg deleted file mode 100644 index bcbd2a91..00000000 --- a/plugins/community/repos/Bidoo/res/HORUS.svg +++ /dev/null @@ -1,186 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/plugins/community/repos/Bidoo/res/HORUStemp.svg b/plugins/community/repos/Bidoo/res/HORUStemp.svg deleted file mode 100644 index c619930f..00000000 --- a/plugins/community/repos/Bidoo/res/HORUStemp.svg +++ /dev/null @@ -1,208 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - bId°° - horUS - - - OUT - - - - IN - - diff --git a/plugins/community/repos/Bidoo/res/RABBIT.svg b/plugins/community/repos/Bidoo/res/RABBIT.svg new file mode 100644 index 00000000..095341af --- /dev/null +++ b/plugins/community/repos/Bidoo/res/RABBIT.svg @@ -0,0 +1,224 @@ + + + + + + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/plugins/community/repos/Bidoo/res/RABBITtemp.svg b/plugins/community/repos/Bidoo/res/RABBITtemp.svg new file mode 100644 index 00000000..407c4cb1 --- /dev/null +++ b/plugins/community/repos/Bidoo/res/RABBITtemp.svg @@ -0,0 +1,230 @@ + + + + + + + + + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + bId°° + rabBIT + + + OUT + + + + IN + + + OFF + REV + diff --git a/plugins/community/repos/Bidoo/res/RADAR.svg b/plugins/community/repos/Bidoo/res/RADAR.svg deleted file mode 100644 index 26dfa5df..00000000 --- a/plugins/community/repos/Bidoo/res/RADAR.svg +++ /dev/null @@ -1,602 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - TRIG - - - POS - - - OUT - GATE - L - R - diff --git a/plugins/community/repos/Bidoo/res/RADARtemp.svg b/plugins/community/repos/Bidoo/res/RADARtemp.svg deleted file mode 100644 index 0789c119..00000000 --- a/plugins/community/repos/Bidoo/res/RADARtemp.svg +++ /dev/null @@ -1,728 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - RAdAR - - - - TRIG - - - POS - - - OUT - GATE - L - R - diff --git a/plugins/community/repos/Bidoo/res/SONAR.svg b/plugins/community/repos/Bidoo/res/SONAR.svg deleted file mode 100644 index d61b0056..00000000 --- a/plugins/community/repos/Bidoo/res/SONAR.svg +++ /dev/null @@ -1,134 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/plugins/community/repos/Bidoo/res/SONARtemp.svg b/plugins/community/repos/Bidoo/res/SONARtemp.svg deleted file mode 100644 index 108cd2ae..00000000 --- a/plugins/community/repos/Bidoo/res/SONARtemp.svg +++ /dev/null @@ -1,153 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - bId°° - SONar - - diff --git a/plugins/community/repos/Bidoo/src/Bidoo.cpp b/plugins/community/repos/Bidoo/src/Bidoo.cpp index e37089aa..efc3d767 100644 --- a/plugins/community/repos/Bidoo/src/Bidoo.cpp +++ b/plugins/community/repos/Bidoo/src/Bidoo.cpp @@ -8,8 +8,8 @@ RACK_PLUGIN_MODEL_DECLARE(Bidoo, CHUTE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, LATE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, LOURDE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, ACNE); -// RACK_PLUGIN_MODEL_DECLARE(Bidoo, OUAIVE); // GPL -// RACK_PLUGIN_MODEL_DECLARE(Bidoo, CANARD); // GPL +RACK_PLUGIN_MODEL_DECLARE(Bidoo, OUAIVE); +RACK_PLUGIN_MODEL_DECLARE(Bidoo, CANARD); RACK_PLUGIN_MODEL_DECLARE(Bidoo, DUKE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, MOIRE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, FORK); @@ -20,10 +20,10 @@ RACK_PLUGIN_MODEL_DECLARE(Bidoo, LIMBO); RACK_PLUGIN_MODEL_DECLARE(Bidoo, PERCO); RACK_PLUGIN_MODEL_DECLARE(Bidoo, BAR); RACK_PLUGIN_MODEL_DECLARE(Bidoo, ZINC); -// RACK_PLUGIN_MODEL_DECLARE(Bidoo, DFUZE); // clashes with AS modules -//RACK_PLUGIN_MODEL_DECLARE(Bidoo, HORUS); +RACK_PLUGIN_MODEL_DECLARE(Bidoo, DFUZE); RACK_PLUGIN_MODEL_DECLARE(Bidoo, VOID); RACK_PLUGIN_MODEL_DECLARE(Bidoo, SIGMA); +RACK_PLUGIN_MODEL_DECLARE(Bidoo, RABBIT); RACK_PLUGIN_INIT(Bidoo) { RACK_PLUGIN_INIT_ID(); @@ -36,8 +36,8 @@ RACK_PLUGIN_INIT(Bidoo) { RACK_PLUGIN_MODEL_ADD(Bidoo, LATE); RACK_PLUGIN_MODEL_ADD(Bidoo, LOURDE); RACK_PLUGIN_MODEL_ADD(Bidoo, ACNE); - // RACK_PLUGIN_MODEL_ADD(Bidoo, OUAIVE); // GPL - // RACK_PLUGIN_MODEL_ADD(Bidoo, CANARD); // GPL + RACK_PLUGIN_MODEL_ADD(Bidoo, OUAIVE); + RACK_PLUGIN_MODEL_ADD(Bidoo, CANARD); RACK_PLUGIN_MODEL_ADD(Bidoo, DUKE); RACK_PLUGIN_MODEL_ADD(Bidoo, MOIRE); RACK_PLUGIN_MODEL_ADD(Bidoo, FORK); @@ -48,8 +48,8 @@ RACK_PLUGIN_INIT(Bidoo) { RACK_PLUGIN_MODEL_ADD(Bidoo, PERCO); RACK_PLUGIN_MODEL_ADD(Bidoo, BAR); RACK_PLUGIN_MODEL_ADD(Bidoo, ZINC); - // RACK_PLUGIN_MODEL_ADD(Bidoo, DFUZE); // clashes with AS modules - //RACK_PLUGIN_MODEL_ADD(Bidoo, HORUS); + RACK_PLUGIN_MODEL_ADD(Bidoo, DFUZE); RACK_PLUGIN_MODEL_ADD(Bidoo, VOID); RACK_PLUGIN_MODEL_ADD(Bidoo, SIGMA); + RACK_PLUGIN_MODEL_ADD(Bidoo, RABBIT); } diff --git a/plugins/community/repos/Bidoo/src/CANARD.cpp b/plugins/community/repos/Bidoo/src/CANARD.cpp index e6d491f1..882c996d 100644 --- a/plugins/community/repos/Bidoo/src/CANARD.cpp +++ b/plugins/community/repos/Bidoo/src/CANARD.cpp @@ -2,14 +2,13 @@ #include "dsp/digital.hpp" #include "BidooComponents.hpp" #include "osdialog.h" -#include "dep/audiofile/AudioFile.h" +#include "dep/dr_wav/dr_wav.h" #include #include "cmath" #include // setprecision #include // stringstream #include #include "window.hpp" -#include "Gist.h" using namespace std; @@ -57,7 +56,10 @@ struct CANARD : Module { bool play = false; bool record = false; - AudioFile playBuffer, recordBuffer; + unsigned int channels = 2; + unsigned int sampleRate = 0; + drwav_uint64 totalSampleCount = 0; + vector> playBuffer, recordBuffer; float samplePos = 0.0f, sampleStart = 0.0f, loopLength = 0.0f, fadeLenght = 0.0f, fadeCoeff = 1.0f, speedFactor = 1.0f; size_t prevPlayedSlice = 0; size_t playedSlice = 0; @@ -86,16 +88,12 @@ struct CANARD : Module { bool newStop = false; CANARD() : Module(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS) { - recordBuffer.setBitDepth(16); - recordBuffer.setSampleRate(engineGetSampleRate()); - recordBuffer.setNumChannels(2); - recordBuffer.samples[0].resize(0); - recordBuffer.samples[1].resize(0); - playBuffer.setBitDepth(16); - playBuffer.setSampleRate(engineGetSampleRate()); - playBuffer.setNumChannels(2); - playBuffer.samples[0].resize(0); - playBuffer.samples[1].resize(0); + playBuffer.resize(2); + playBuffer[0].resize(0); + playBuffer[1].resize(0); + recordBuffer.resize(2); + recordBuffer[0].resize(0); + recordBuffer[1].resize(0); } void step() override; @@ -108,7 +106,6 @@ struct CANARD : Module { json_t *rootJ = json_object(); // lastPath json_object_set_new(rootJ, "lastPath", json_string(lastPath.c_str())); - json_t *slicesJ = json_array(); for (size_t i = 0; i0) { + if (totalSampleCount>0) { json_t *slicesJ = json_object_get(rootJ, "slices"); if (slicesJ) { size_t i; @@ -143,17 +140,30 @@ struct CANARD : Module { void CANARD::loadSample(std::string path) { loading = true; - playBuffer.setNumChannels(1); - if (playBuffer.load(path.c_str())) { + unsigned int c; + unsigned int sr; + drwav_uint64 sc; + float* pSampleData; + pSampleData = drwav_open_and_read_file_f32(path.c_str(), &c, &sr, &sc); + if (pSampleData != NULL) { lastPath = path; waveFileName = stringFilename(path); waveExtension = stringExtension(path); + channels = c; + sampleRate = sr; slices.clear(); slices.push_back(0); - if (playBuffer.getNumChannels() == 1) { - playBuffer.setNumChannels(2); - playBuffer.samples[1] = playBuffer.samples[0]; + playBuffer[0].clear(); + playBuffer[1].clear(); + for (unsigned int i=0; i < sc; i = i + c) { + playBuffer[0].push_back(pSampleData[i]); + if (channels == 2) + playBuffer[1].push_back((float)pSampleData[i+1]); + else + playBuffer[1].push_back((float)pSampleData[i]); } + totalSampleCount = playBuffer[0].size(); + drwav_free(pSampleData); } loading = false; } @@ -162,15 +172,15 @@ void CANARD::calcLoop() { prevPlayedSlice = index; index = 0; int sliceStart = 0;; - int sliceEnd = playBuffer.getNumSamplesPerChannel() > 0 ? playBuffer.getNumSamplesPerChannel() - 1 : 0; + int sliceEnd = totalSampleCount > 0 ? totalSampleCount - 1 : 0; if ((params[MODE_PARAM].value == 1) && (slices.size()>0)) { index = round(clamp(params[SLICE_PARAM].value + inputs[SLICE_INPUT].value, 0.0f,10.0f)*(slices.size()-1)/10); sliceStart = slices[index]; - sliceEnd = (index < (slices.size() - 1)) ? (slices[index+1] - 1) : (playBuffer.getNumSamplesPerChannel() - 1); + sliceEnd = (index < (slices.size() - 1)) ? (slices[index+1] - 1) : (totalSampleCount - 1); } - if (playBuffer.getNumSamplesPerChannel() > 0) { + if (totalSampleCount > 0) { sampleStart = rescale(clamp(inputs[SAMPLE_START_INPUT].value + params[SAMPLE_START_PARAM].value, 0.0f, 10.0f), 0.0f, 10.0f, sliceStart, sliceEnd); loopLength = clamp(rescale(clamp(inputs[LOOP_LENGTH_INPUT].value + params[LOOP_LENGTH_PARAM].value, 0.0f, 10.0f), 0.0f, 10.0f, 0.0f, sliceEnd - sliceStart + 1),1.0f,sliceEnd-sampleStart+1); fadeLenght = rescale(clamp(inputs[FADE_INPUT].value + params[FADE_PARAM].value, 0.0f, 10.0f), 0.0f, 10.0f,0.0f, floor(loopLength/2)); @@ -200,8 +210,8 @@ void CANARD::step() { if (clearTrigger.process(inputs[CLEAR_INPUT].value + params[CLEAR_PARAM].value)) { mylock.lock(); - playBuffer.samples[0].clear(); - playBuffer.samples[1].clear(); + playBuffer[0].clear(); + playBuffer[1].clear(); slices.clear(); mylock.unlock(); lastPath = ""; @@ -214,18 +224,19 @@ void CANARD::step() { if ((size_t)selected<(slices.size()-1)) { nbSample = slices[selected + 1] - slices[selected] - 1; mylock.lock(); - playBuffer.samples[0].erase(playBuffer.samples[0].begin() + slices[selected], playBuffer.samples[0].begin() + slices[selected + 1]-1); - playBuffer.samples[1].erase(playBuffer.samples[1].begin() + slices[selected], playBuffer.samples[1].begin() + slices[selected + 1]-1); + playBuffer[0].erase(playBuffer[0].begin() + slices[selected], playBuffer[0].begin() + slices[selected + 1]-1); + playBuffer[1].erase(playBuffer[1].begin() + slices[selected], playBuffer[1].begin() + slices[selected + 1]-1); mylock.unlock(); } else { - nbSample = playBuffer.getNumSamplesPerChannel() - slices[selected]; + nbSample = totalSampleCount - slices[selected]; mylock.lock(); - playBuffer.samples[0].erase(playBuffer.samples[0].begin() + slices[selected], playBuffer.samples[0].end()); - playBuffer.samples[1].erase(playBuffer.samples[1].begin() + slices[selected], playBuffer.samples[1].end()); + playBuffer[0].erase(playBuffer[0].begin() + slices[selected], playBuffer[0].end()); + playBuffer[1].erase(playBuffer[1].begin() + slices[selected], playBuffer[1].end()); mylock.unlock(); } slices.erase(slices.begin()+selected); + totalSampleCount = playBuffer[0].size(); for (size_t i = selected; i < slices.size(); i++) { slices[i] = slices[i]-nbSample; @@ -270,7 +281,14 @@ void CANARD::step() { mylock.lock(); slices.clear(); slices.push_back(0); - playBuffer.setAudioBuffer(recordBuffer.samples); + playBuffer.resize(2); + playBuffer[0].resize((int)recordBuffer[0].size()); + playBuffer[1].resize((int)recordBuffer[0].size()); + for (int i = 0; i < (int)recordBuffer[0].size(); i++) { + playBuffer[0][i] = recordBuffer[0][i]; + playBuffer[1][i] = recordBuffer[1][i]; + } + totalSampleCount = playBuffer[0].size(); mylock.unlock(); lastPath = ""; waveFileName = ""; @@ -278,14 +296,15 @@ void CANARD::step() { } else { mylock.lock(); - slices.push_back(playBuffer.getNumSamplesPerChannel() > 0 ? (playBuffer.getNumSamplesPerChannel()-1) : 0); - playBuffer.samples[0].insert(playBuffer.samples[0].end(), recordBuffer.samples[0].begin(), recordBuffer.samples[0].end()); - playBuffer.samples[1].insert(playBuffer.samples[1].end(), recordBuffer.samples[1].begin(), recordBuffer.samples[1].end()); + slices.push_back(totalSampleCount > 0 ? (totalSampleCount-1) : 0); + playBuffer[0].insert(playBuffer[0].end(), recordBuffer[0].begin(), recordBuffer[0].end()); + playBuffer[1].insert(playBuffer[1].end(), recordBuffer[1].begin(), recordBuffer[1].end()); + totalSampleCount = playBuffer[0].size(); mylock.unlock(); } mylock.lock(); - recordBuffer.samples[0].resize(0); - recordBuffer.samples[1].resize(0); + recordBuffer[0].resize(0); + recordBuffer[1].resize(0); mylock.unlock(); lights[REC_LIGHT].value = 0.0f; } @@ -294,8 +313,8 @@ void CANARD::step() { if (record) { mylock.lock(); - recordBuffer.samples[0].push_back(inputs[INL_INPUT].value/10); - recordBuffer.samples[1].push_back(inputs[INR_INPUT].value/10); + recordBuffer[0].push_back(inputs[INL_INPUT].value/10); + recordBuffer[1].push_back(inputs[INR_INPUT].value/10); mylock.unlock(); } @@ -389,7 +408,7 @@ void CANARD::step() { if (play) { newStop = true; - if (samplePos1000) { if ((samplePos-sampleStart)slices.size()>0) { refX = e.pos.x; - refIdx = ((e.pos.x - zoomLeftAnchor)/zoomWidth)*(float)module->playBuffer.getNumSamplesPerChannel(); + refIdx = ((e.pos.x - zoomLeftAnchor)/zoomWidth)*(float)module->totalSampleCount; module->addSliceMarker = refIdx; auto lower = std::lower_bound(module->slices.begin(), module->slices.end(), refIdx); module->selected = distance(module->slices.begin(),lower-1); @@ -478,8 +497,8 @@ struct CANARDDisplay : OpaqueWidget { void draw(NVGcontext *vg) override { module->mylock.lock(); - std::vector vL(module->playBuffer.samples[0]); - std::vector vR(module->playBuffer.samples[1]); + std::vector vL(module->playBuffer[0]); + std::vector vR(module->playBuffer[1]); std::vector s(module->slices); module->mylock.unlock(); size_t nbSample = vL.size(); @@ -490,7 +509,7 @@ struct CANARDDisplay : OpaqueWidget { { nvgBeginPath(vg); nvgStrokeWidth(vg, 2); - if (module->playBuffer.getNumSamplesPerChannel()>0) { + if (module->totalSampleCount>0) { nvgMoveTo(vg, module->samplePos * zoomWidth / nbSample + zoomLeftAnchor, 0); nvgLineTo(vg, module->samplePos * zoomWidth / nbSample + zoomLeftAnchor, 2*height+10); } @@ -674,7 +693,7 @@ CANARDWidget::CANARDWidget(CANARD *module) : ModuleWidget(module) { addInput(Port::create(Vec(portX0[1]-4, 277), Port::INPUT, module, CANARD::FADE_INPUT)); addInput(Port::create(Vec(portX0[2]-4, 277), Port::INPUT, module, CANARD::SLICE_INPUT)); addInput(Port::create(Vec(portX0[3]-4, 277), Port::INPUT, module, CANARD::CLEAR_INPUT)); - addParam(ParamWidget::create(Vec(portX0[4]-1, 280), module, CANARD::THRESHOLD_PARAM, 0.0001f, 0.05f, 0.05f)); + addParam(ParamWidget::create(Vec(portX0[4]-1, 280), module, CANARD::THRESHOLD_PARAM, 0.01f, 10.0f, 1.0f)); addParam(ParamWidget::create(Vec(90, 325), module, CANARD::MODE_PARAM, 0.0f, 1.0f, 0.0f)); @@ -714,22 +733,33 @@ struct CANARDTransientDetect : MenuItem { void onAction(EventAction &e) override { canardModule->slices.clear(); canardModule->slices.push_back(0); - int i = 0; - int size = 256; - Gist gist = Gist(size,engineGetSampleRate()); + unsigned int i = 0; + unsigned int size = 256; vector::const_iterator first; vector::const_iterator last; - while (i+sizeplayBuffer.getNumSamplesPerChannel()) { - first = canardModule->playBuffer.samples[0].begin() + i; - last = canardModule->playBuffer.samples[0].begin() + i + size; + float prevNrgy = 0.0f; + while (i+sizetotalSampleCount) { + first = canardModule->playBuffer[0].begin() + i; + last = canardModule->playBuffer[0].begin() + i + size; vector newVec(first, last); - gist.processAudioFrame(newVec); - if (((gist.energyDifference()/size)>canardModule->params[CANARD::THRESHOLD_PARAM].value) - && ((gist.complexSpectralDifference()/size)>canardModule->params[CANARD::THRESHOLD_PARAM].value) - && ((gist.zeroCrossingRate()/size)>canardModule->params[CANARD::THRESHOLD_PARAM].value)) { - canardModule->slices.push_back(i); + float nrgy = 0.0f; + float zcRate = 0.0f; + unsigned int zcIdx = 0; + bool first = true; + for (unsigned int k = 0; k < size; k++) { + nrgy += 100*newVec[k]*newVec[k]/size; + if (newVec[k]==0.0f) { + zcRate += 1; + if (first) { + zcIdx = k; + first = false; + } + } } + if ((nrgy > canardModule->params[CANARD::THRESHOLD_PARAM].value) && (nrgy > 10*prevNrgy)) + canardModule->slices.push_back(i+zcIdx); i+=size; + prevNrgy = nrgy; } } }; @@ -748,20 +778,34 @@ struct CANARDLoadSample : MenuItem { }; struct CANARDSaveSample : MenuItem { - CANARDWidget *canardWidget; - CANARD *canardModule; - void onAction(EventAction &e) override { - std::string dir = canardModule->lastPath.empty() ? assetLocal("") : stringDirectory(canardModule->lastPath); - char *path = osdialog_file(OSDIALOG_SAVE, dir.c_str(), (canardModule->waveFileName).c_str(), NULL); - if (path) { - canardModule->lastPath = path; - canardModule->waveFileName = stringDirectory(path); - canardModule->waveExtension = stringExtension(path); - canardModule->playBuffer.setSampleRate(engineGetSampleRate()); - canardModule->playBuffer.save(path); - free(path); - } - } + CANARDWidget *canardWidget; + CANARD *canardModule; + void onAction(EventAction &e) override { + std::string dir = canardModule->lastPath.empty() ? assetLocal("") : stringDirectory(canardModule->lastPath); + std::string fileName = canardModule->waveFileName.empty() ? "temp.wav" : canardModule->waveFileName; + char *path = osdialog_file(OSDIALOG_SAVE, dir.c_str(), (fileName).c_str(), NULL); + if (path) { + canardModule->lastPath = path; + canardModule->waveFileName = stringDirectory(path); + canardModule->waveExtension = stringExtension(path); + drwav_data_format format; + format.container = drwav_container_riff; + format.format = DR_WAVE_FORMAT_PCM; + format.channels = 2; + format.sampleRate = engineGetSampleRate(); + format.bitsPerSample = 32; + drwav* pWav = drwav_open_file_write(path, &format); + int *pSamples = new int[2*canardModule->totalSampleCount]; + for (unsigned int i = 0; i < canardModule->totalSampleCount; i++) { + pSamples[2*i]= floor(canardModule->playBuffer[0][i]*2147483647); + pSamples[2*i+1]= floor(canardModule->playBuffer[1][i]*2147483647); + } + drwav_write(pWav, 2*canardModule->totalSampleCount, pSamples); + drwav_close(pWav); + free(path); + delete [] pSamples; + } + } }; Menu *CANARDWidget::createContextMenu() { @@ -775,7 +819,7 @@ Menu *CANARDWidget::createContextMenu() { MenuLabel *spacerLabel; - if ((canardModule->selected>=0) || (canardModule->playBuffer.getNumSamplesPerChannel()>=0)) { + if ((canardModule->selected>=0) || (canardModule->totalSampleCount>0)) { spacerLabel = new MenuLabel(); menu->addChild(spacerLabel); } @@ -788,7 +832,7 @@ Menu *CANARDWidget::createContextMenu() { menu->addChild(deleteItem); } - if (canardModule->playBuffer.getNumSamplesPerChannel()>=0) { + if (canardModule->totalSampleCount>0) { CANARDAddSliceMarker *addSliceItem = new CANARDAddSliceMarker(); addSliceItem->text = "Add slice marker"; addSliceItem->canardWidget = this; @@ -831,6 +875,6 @@ Menu *CANARDWidget::createContextMenu() { using namespace rack_plugin_Bidoo; RACK_PLUGIN_MODEL_INIT(Bidoo, CANARD) { - Model *modelCANARD = Model::create("Bidoo","cANARd", "cANARd sampler", SAMPLER_TAG); + Model *modelCANARD = Model::create("Bidoo","cANARd", "cANARd sampler", SAMPLER_TAG, GRANULAR_TAG, RECORDING_TAG); return modelCANARD; } diff --git a/plugins/community/repos/Bidoo/src/DFUZE.cpp b/plugins/community/repos/Bidoo/src/DFUZE.cpp new file mode 100644 index 00000000..4b36aee5 --- /dev/null +++ b/plugins/community/repos/Bidoo/src/DFUZE.cpp @@ -0,0 +1,147 @@ +#include "Bidoo.hpp" +#include "BidooComponents.hpp" +#include "dsp/ringbuffer.hpp" +#include "dep/freeverb/revmodel.hpp" +#include "dep/filters/smbPitchShift.hpp" +#include "dsp/digital.hpp" + +using namespace std; + +namespace rack_plugin_Bidoo { + +struct DFUZE : Module { + enum ParamIds { + SIZE_PARAM, + DAMP_PARAM, + FREEZE_PARAM, + WIDTH_PARAM, + DRY_PARAM, + WET_PARAM, + SHIMM_PARAM, + NUM_PARAMS + }; + enum InputIds { + IN_L_INPUT, + IN_R_INPUT, + SIZE_INPUT, + DAMP_INPUT, + FREEZE_INPUT, + WIDTH_INPUT, + SHIMM_INPUT, + NUM_INPUTS + }; + enum OutputIds { + OUT_L_OUTPUT, + OUT_R_OUTPUT, + NUM_OUTPUTS + }; + enum LightIds { + NUM_LIGHTS + }; + DoubleRingBuffer in_L_Buffer, in_R_Buffer; + DoubleRingBuffer pin_L_Buffer, pin_R_Buffer; + revmodel revprocessor; + SchmittTrigger freezeTrigger; + bool freeze = false; + float sr = engineGetSampleRate(); + + DFUZE() : Module(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS) { + + } + + void step() override; +}; + +void DFUZE::step() { + float outL = 0.0f, outR = 0.0f; + float wOutL = 0.0f, wOutR = 0.0f; + float inL = 0.0f, inR = 0.0f; + + // if (sr != engineGetSampleRate()) { + // revprocessor.setsamplerate(engineGetSampleRate()); + // in_L_Buffer.clear(); + // in_R_Buffer.clear(); + // pin_L_Buffer.clear(); + // pin_R_Buffer.clear(); + // sr = engineGetSampleRate(); + // } + + revprocessor.setdamp(clamp(params[DAMP_PARAM].value+inputs[DAMP_INPUT].value,0.0f,1.0f)); + revprocessor.setroomsize(clamp(params[SIZE_PARAM].value+inputs[SIZE_INPUT].value,0.0f,1.0f)); + revprocessor.setwet(clamp(params[WET_PARAM].value,0.0f,1.0f)); + revprocessor.setdry(clamp(params[DRY_PARAM].value,0.0f,1.0f)); + revprocessor.setwidth(clamp(params[WIDTH_PARAM].value+inputs[WIDTH_INPUT].value,0.0f,1.0f)); + + if (freezeTrigger.process(params[FREEZE_PARAM].value + inputs[FREEZE_INPUT].value )) freeze = !freeze; + + revprocessor.setmode(freeze?1.0:0.0); + + inL = inputs[IN_L_INPUT].value; + inR = inputs[IN_R_INPUT].value; + + if (pin_L_Buffer.size()>0) { + revprocessor.process(inL, inR, params[SHIMM_PARAM].value * (*pin_L_Buffer.startData()) * 5, clamp(params[SHIMM_PARAM].value+inputs[SHIMM_INPUT].value,0.0f,0.08f) * (*pin_R_Buffer.startData()) * 5, outL, outR, wOutL, wOutR); + pin_L_Buffer.startIncr(1); + pin_R_Buffer.startIncr(1); + } + else { + revprocessor.process(inL, inR, 0.0f, 0.0f, outL, outR, wOutL, wOutR); + } + + in_L_Buffer.push(wOutL/10); + in_R_Buffer.push(wOutR/10); + + if (in_L_Buffer.full()) { + smbPitchShift(2.0f, in_L_Buffer.size(), 2048, 4, engineGetSampleRate(), in_L_Buffer.startData(), pin_L_Buffer.endData()); + smbPitchShift(2.0f, in_R_Buffer.size(), 2048, 4, engineGetSampleRate(), in_R_Buffer.startData(), pin_R_Buffer.endData()); + pin_L_Buffer.endIncr(in_L_Buffer.size()); + pin_R_Buffer.endIncr(in_L_Buffer.size()); + in_L_Buffer.clear(); + in_R_Buffer.clear(); + } + + outputs[OUT_L_OUTPUT].value = outL; + outputs[OUT_R_OUTPUT].value = outR; +} + + + +struct DFUZEWidget : ModuleWidget { + DFUZEWidget(DFUZE *module) : ModuleWidget(module) { + setPanel(SVG::load(assetPlugin(plugin, "res/DFUZE.svg"))); + + addChild(Widget::create(Vec(RACK_GRID_WIDTH, 0))); + addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, 0))); + addChild(Widget::create(Vec(RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); + addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); + + addParam(ParamWidget::create(Vec(13, 50), module, DFUZE::SIZE_PARAM, 0.0f, 1.0f, 0.5f)); + addParam(ParamWidget::create(Vec(13, 95), module, DFUZE::DAMP_PARAM, 0.0f, 1.0f, 0.5f)); + addParam(ParamWidget::create(Vec(13, 140), module, DFUZE::WIDTH_PARAM, 0.0f, 1.0f, 0.5f)); + addParam(ParamWidget::create(Vec(13, 185), module, DFUZE::DRY_PARAM, 0.0f, 1.0f, 0.0f)); + addParam(ParamWidget::create(Vec(63, 185), module, DFUZE::WET_PARAM, 0.0f, 1.0f, 1.0f)); + addParam(ParamWidget::create(Vec(13, 230), module, DFUZE::SHIMM_PARAM, 0.0f, 0.08f, 0.0f)); + addParam(ParamWidget::create(Vec(13, 276), module, DFUZE::FREEZE_PARAM, 0.0f, 10.0f, 0.0f)); + + addInput(Port::create(Vec(65.0f, 52.0f), Port::INPUT, module, DFUZE::SIZE_INPUT)); + addInput(Port::create(Vec(65.0f, 97.0f), Port::INPUT, module, DFUZE::DAMP_INPUT)); + addInput(Port::create(Vec(65.0f, 142.0f), Port::INPUT, module, DFUZE::WIDTH_INPUT)); + addInput(Port::create(Vec(65.0f, 232.0f), Port::INPUT, module, DFUZE::SHIMM_INPUT)); + addInput(Port::create(Vec(65.0f, 277.0f), Port::INPUT, module, DFUZE::FREEZE_INPUT)); + + //Changed ports opposite way around + addInput(Port::create(Vec(24.0f, 319.0f), Port::INPUT, module, DFUZE::IN_L_INPUT)); + addInput(Port::create(Vec(24.0f, 339.0f), Port::INPUT, module, DFUZE::IN_R_INPUT)); + addOutput(Port::create(Vec(78.0f, 319.0f),Port::OUTPUT, module, DFUZE::OUT_L_OUTPUT)); + addOutput(Port::create(Vec(78.0f, 339.0f),Port::OUTPUT, module, DFUZE::OUT_R_OUTPUT)); + } +}; + +} // namespace rack_plugin_Bidoo + +using namespace rack_plugin_Bidoo; + +RACK_PLUGIN_MODEL_INIT(Bidoo, DFUZE) { + Model *modelDFUZE = Model::create("Bidoo", "dFUZE", "dFUZE reverberator", REVERB_TAG, EFFECT_TAG); + return modelDFUZE; +} diff --git a/plugins/community/repos/Bidoo/src/HORUS.cpp b/plugins/community/repos/Bidoo/src/HORUS.cpp deleted file mode 100644 index ac91e617..00000000 --- a/plugins/community/repos/Bidoo/src/HORUS.cpp +++ /dev/null @@ -1,238 +0,0 @@ -// #include "Bidoo.hpp" -// #include "BidooComponents.hpp" -// #include "dsp/samplerate.hpp" -// #include "dsp/decimator.hpp" -// #include "dsp/filter.hpp" -// #include "dsp/ringbuffer.hpp" -// #include "dsp/digital.hpp" -// #include -// -// #define BOTTOM_FREQ 100 -// #define BSZ 65536 -// #define ROUND(n) ((int)((double)(n)+0.5)) -// #define PIN(n,min,max) ((n) > (max) ? max : ((n) < (min) ? (min) : (n))) -// #define MODF(n,i,f) ((i) = (int)(n), (f) = (n) - (double)(i)) -// -// enum -// { -// kMixMono, -// kMixWetOnly, -// kMixWetLeft, -// kMixWetRight, -// kMixWetLeft75, -// kMixWetRight75, -// kMixStereo -// }; -// -// #define NUM_MIX_MODES 7 -// #define NUM_DELAYS 11 -// -// using namespace std; -// -// struct HORUS : Module { -// enum ParamIds { -// RATE_PARAM, -// WIDTH_PARAM, -// FEEDBACK_PARAM, -// DELAY_PARAM, -// MIXMODE_PARAM, -// NUM_PARAMS -// }; -// enum InputIds { -// L_INPUT, -// R_INPUT, -// NUM_INPUTS -// }; -// enum OutputIds { -// L_OUTPUT, -// R_OUTPUT, -// NUM_OUTPUTS -// }; -// enum LightIds { -// NUM_LIGHTS -// }; -// -// void setRate (float v); -// void setWidth (float v); -// void setFeedback (float v); -// void setDelay (float v); -// void setMixMode (float v); -// void setSweep(void); -// -// float _paramSweepRate = 0.2f; -// float _paramWidth = 0.3f; -// float _paramFeedback = 0.0f; -// float _paramDelay = 0.2f; -// float _paramMixMode = 0.0f; -// double _sweepRate = 0.2; -// double _feedback = 0.0; -// double _feedbackPhase = 1.0; -// int _sweepSamples = 0; -// int _delaySamples = 22; -// double _minSweepSamples; -// double _maxSweepSamples; -// int _mixMode = 0; -// double *_buf = new double[BSZ]; -// int _fp = 0; -// double _step; -// double _sweep = 0.0; -// -// -// double _mixLeftWet = 0.5f; -// double _mixLeftDry = 0.5f; -// double _mixRightWet = 0.5f; -// double _mixRightDry = 0.5f; -// -// -// HORUS() : Module(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS) { -// } -// -// ~HORUS() { -// if( _buf ) -// delete[] _buf; -// } -// -// void step() override; -// }; -// -// void HORUS::setSweep() -// { -// _step = (double)(_sweepSamples * 2.0 * _sweepRate) / (double)engineGetSampleRate(); -// if( _step <= 1.0 ) -// { -// printf( "_sweepSamples: %i\n", _sweepSamples ); -// printf( "_sweepRate: %f\n", _sweepRate ); -// printf( "engineGetSampleRate: %f\n", engineGetSampleRate() ); -// printf( "_step out of range: %f\n", _step ); -// } -// -// _minSweepSamples = _delaySamples; -// _maxSweepSamples = _delaySamples + _sweepSamples; -// -// _sweep = (_minSweepSamples + _maxSweepSamples) / 2; -// } -// -// void HORUS::setRate (float rate) -// { -// _paramSweepRate = rate; -// _sweepRate = pow(10.0,(double)_paramSweepRate); -// _sweepRate -= 1.0; -// _sweepRate *= 1.1f; -// _sweepRate += 0.1f; -// } -// -// void HORUS::setWidth (float v) -// { -// _paramWidth = v; -// _sweepSamples = ROUND(v * 0.05 * engineGetSampleRate()); -// } -// -// void HORUS::setDelay (float v) -// { -// _paramDelay = v; -// double delay = pow(10.0, (double)v * 2.0)/1000.0; -// _delaySamples = ROUND(delay * engineGetSampleRate()); -// } -// -// void HORUS::setFeedback(float v) -// { -// _paramFeedback = v; -// _feedback = v; -// } -// -// void HORUS::setMixMode (float v) -// { -// _paramMixMode = v; -// _mixMode = (int)(v * NUM_MIX_MODES); -// switch(_mixMode) -// { -// case kMixMono: -// default: -// _mixLeftWet = _mixRightWet = 1.0; -// _mixLeftDry = _mixRightDry = 1.0f; -// _feedbackPhase = 1.0; -// break; -// case kMixWetOnly: -// _mixLeftWet = _mixRightWet = 1.0f; -// _mixLeftDry = _mixRightDry = 1.0; -// _feedbackPhase = -1.0; -// break; -// case kMixWetLeft: -// _mixLeftWet = 1.0f; -// _mixLeftDry = 0.0f; -// _mixRightWet = 0.0f; -// _mixRightDry = 1.0f; -// break; -// case kMixWetRight: -// _mixLeftWet = 0.0f; -// _mixLeftDry = 1.0f; -// _mixRightWet = 1.0f; -// _mixRightDry = 0.0f; -// break; -// case kMixStereo: -// _mixLeftWet = 1.0f; -// _mixLeftDry = 1.0f; -// _mixRightWet = -1.0f; -// _mixRightDry = 1.0f; -// break; -// } -// } -// -// -// void HORUS::step() { -// setRate(params[RATE_PARAM].value); -// setWidth(params[WIDTH_PARAM].value); -// setFeedback(params[FEEDBACK_PARAM].value); -// setDelay(params[DELAY_PARAM].value); -// setMixMode(params[MIXMODE_PARAM].value); -// setSweep(); -// -// float inval = (inputs[L_INPUT].value + inputs[R_INPUT].value) / 2.0f; -// _buf[_fp] = inval; -// _fp = (_fp + 1) & (BSZ-1); -// -// int ep1, ep2; -// double w1, w2; -// double ep = _fp - _sweep; -// MODF(ep, ep1, w2); -// ep1 &= (BSZ-1); -// ep2 = ep1 + 1; -// ep2 &= (BSZ-1); -// w1 = 1.0 - w2; -// double outval = _buf[ep1] * w1 + _buf[ep2] * w2; -// -// outputs[L_OUTPUT].value = (float)PIN(_mixLeftDry * inval + _mixLeftWet * outval, -10, 10); -// outputs[R_OUTPUT].value = (float)PIN(_mixRightDry * inval + _mixRightWet * outval,-10, 10); -// -// _sweep += _step; -// if( _sweep >= _maxSweepSamples || _sweep <= _minSweepSamples ) -// { -// _step = -_step; -// } -// } -// -// struct HORUSWidget : ModuleWidget { -// HORUSWidget(HORUS *module) : ModuleWidget(module) { -// setPanel(SVG::load(assetPlugin(plugin, "res/HORUS.svg"))); -// -// addChild(Widget::create(Vec(RACK_GRID_WIDTH, 0))); -// addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, 0))); -// addChild(Widget::create(Vec(RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); -// addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); -// -// addParam(ParamWidget::create(Vec(13, 50), module, HORUS::RATE_PARAM, 0.0f, 150.0f, 0.0f)); -// addParam(ParamWidget::create(Vec(13, 100), module, HORUS::WIDTH_PARAM, 0.0f, 5.0f, 0.0f)); -// addParam(ParamWidget::create(Vec(13, 150), module, HORUS::DELAY_PARAM, 0.0f, 1.0f, 0.0f)); -// addParam(ParamWidget::create(Vec(13, 200), module, HORUS::FEEDBACK_PARAM, 0.0f, 1.0f, 0.0f)); -// addParam(ParamWidget::create(Vec(13, 250), module, HORUS::MIXMODE_PARAM, 0.0f, 1.0f, 0.0f)); -// -// addInput(Port::create(Vec(24.0f, 319.0f), Port::INPUT, module, HORUS::L_INPUT)); -// addInput(Port::create(Vec(24.0f, 339.0f), Port::INPUT, module, HORUS::R_INPUT)); -// addOutput(Port::create(Vec(78.0f, 319.0f),Port::OUTPUT, module, HORUS::L_OUTPUT)); -// addOutput(Port::create(Vec(78.0f, 339.0f),Port::OUTPUT, module, HORUS::R_OUTPUT)); -// } -// }; -// -// -// -// Model *modelHORUS = Model::create("Bidoo", "horUS", "horUS chorus", CHORUS_TAG); diff --git a/plugins/community/repos/Bidoo/src/OUAIVE.cpp b/plugins/community/repos/Bidoo/src/OUAIVE.cpp index c8d76735..76ad052c 100644 --- a/plugins/community/repos/Bidoo/src/OUAIVE.cpp +++ b/plugins/community/repos/Bidoo/src/OUAIVE.cpp @@ -2,7 +2,8 @@ #include "dsp/digital.hpp" #include "BidooComponents.hpp" #include "osdialog.h" -#include "dep/audiofile/AudioFile.h" +#define DR_WAV_IMPLEMENTATION +#include "dep/dr_wav/dr_wav.h" #include #include "cmath" #include // setprecision @@ -39,7 +40,10 @@ struct OUAIVE : Module { bool play = false; string lastPath; - AudioFile audioFile; + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalSampleCount; + float* pSampleData; float samplePos = 0.0f; vector displayBuffL; vector displayBuffR; @@ -55,10 +59,10 @@ struct OUAIVE : Module { string displayReadMode = ""; string displaySlices = ""; string displaySpeed; - SchmittTrigger playTrigger; SchmittTrigger trigModeTrigger; SchmittTrigger readModeTrigger; + std::mutex mylock; OUAIVE() : Module(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS) { } @@ -74,6 +78,7 @@ struct OUAIVE : Module { // lastPath json_object_set_new(rootJ, "lastPath", json_string(lastPath.c_str())); json_object_set_new(rootJ, "trigMode", json_integer(trigMode)); + json_object_set_new(rootJ, "readMode", json_integer(readMode)); return rootJ; } @@ -88,26 +93,34 @@ struct OUAIVE : Module { if (trigModeJ) { trigMode = json_integer_value(trigModeJ); } + json_t *readModeJ = json_object_get(rootJ, "readMode"); + if (readModeJ) { + readMode = json_integer_value(readModeJ); + } } }; void OUAIVE::loadSample(std::string path) { - if (audioFile.load (path.c_str())) { - fileLoaded = true; + mylock.lock(); + fileLoaded = false; + drwav_free(pSampleData); + pSampleData = drwav_open_and_read_file_f32(path.c_str(), &channels, &sampleRate, &totalSampleCount); + if (pSampleData == NULL) { + fileLoaded = false; + } + else { vector().swap(displayBuffL); vector().swap(displayBuffR); - for (int i=0; i < audioFile.getNumSamplesPerChannel(); i = i + floor(audioFile.getNumSamplesPerChannel()/125)) { - displayBuffL.push_back(audioFile.samples[0][i]); - if (audioFile.getNumChannels() == 2) - displayBuffR.push_back(audioFile.samples[1][i]); + for (unsigned int i=0; i < totalSampleCount; i = i + floor(totalSampleCount/125)) { + displayBuffL.push_back(pSampleData[i]); + if (channels == 2) + displayBuffR.push_back(pSampleData[i+1]); } fileDesc = (stringFilename(path)).substr(0,20) + ((stringFilename(path)).length() >=20 ? "...\n" : "\n"); - fileDesc += std::to_string(audioFile.getSampleRate()) + " Hz " + std::to_string(audioFile.getBitDepth()) + " bit\n"; - fileDesc += std::to_string(roundf(audioFile.getLengthInSeconds() * 100) / 100) + " s\n"; - } - else { - fileLoaded = false; + fileDesc += std::to_string(sampleRate) + " Hz\n"; + fileLoaded = true; } + mylock.unlock(); } void OUAIVE::step() { @@ -152,14 +165,14 @@ void OUAIVE::step() { if (fileLoaded) { - sliceLength = clamp(audioFile.getNumSamplesPerChannel() / nbSlices, 1, audioFile.getNumSamplesPerChannel()); + sliceLength = clamp(totalSampleCount / nbSlices, 1, totalSampleCount); if ((trigMode == 0) && (playTrigger.process(inputs[GATE_INPUT].value))) { play = true; - samplePos = clamp((int)(inputs[POS_INPUT].value*audioFile.getNumSamplesPerChannel()/10), 0 , audioFile.getNumSamplesPerChannel() -1); + samplePos = clamp((int)(inputs[POS_INPUT].value * totalSampleCount / 10), 0 , totalSampleCount - 1); } else if (trigMode == 1) { play = (inputs[GATE_INPUT].value > 0); - samplePos = clamp((int)(inputs[POS_INPUT].value*audioFile.getNumSamplesPerChannel()/10), 0 , audioFile.getNumSamplesPerChannel() -1); + samplePos = clamp((int)(inputs[POS_INPUT].value * totalSampleCount / 10), 0 , totalSampleCount - 1); } else if ((trigMode == 2) && (playTrigger.process(inputs[GATE_INPUT].value))) { play = true; if (inputs[POS_INPUT].active) @@ -167,61 +180,62 @@ void OUAIVE::step() { else sliceIndex = (sliceIndex+1)%nbSlices; if (readMode != 1) - samplePos = clamp(sliceIndex*sliceLength, 0, audioFile.getNumSamplesPerChannel()); + samplePos = clamp(sliceIndex*sliceLength, 0, totalSampleCount); else - samplePos = clamp((sliceIndex + 1) * sliceLength - 1, 0 , audioFile.getNumSamplesPerChannel()); + samplePos = clamp((sliceIndex + 1) * sliceLength - 1, 0 , totalSampleCount); } - if ((play) && (samplePos>=0) && (samplePos < audioFile.getNumSamplesPerChannel())) { + if ((play) && (samplePos>=0) && (samplePos < totalSampleCount)) { + mylock.lock(); //calulate outputs - if (audioFile.getNumChannels() == 1) { - outputs[OUTL_OUTPUT].value = 5.0f * audioFile.samples[0][floor(samplePos)]; - outputs[OUTR_OUTPUT].value = 5.0f * audioFile.samples[0][floor(samplePos)]; + if (channels == 1) { + outputs[OUTL_OUTPUT].value = 10.0f * pSampleData[(unsigned int)floor(samplePos)]; + outputs[OUTR_OUTPUT].value = 10.0f * pSampleData[(unsigned int)floor(samplePos)]; } - else if (audioFile.getNumChannels() == 2) { + else if (channels == 2) { if (outputs[OUTL_OUTPUT].active && outputs[OUTR_OUTPUT].active) { - outputs[OUTL_OUTPUT].value = 5.0f * audioFile.samples[0][floor(samplePos)]; - outputs[OUTR_OUTPUT].value = 5.0f * audioFile.samples[1][floor(samplePos)]; + outputs[OUTL_OUTPUT].value = 10.0f * pSampleData[(unsigned int)floor(samplePos)]; + outputs[OUTR_OUTPUT].value = 10.0f * pSampleData[(unsigned int)floor(samplePos)+1]; } else { - outputs[OUTL_OUTPUT].value = 5.0f * (audioFile.samples[0][floor(samplePos)] + audioFile.samples[1][floor(samplePos)]) / 2; - outputs[OUTR_OUTPUT].value = 5.0f * (audioFile.samples[0][floor(samplePos)] + audioFile.samples[1][floor(samplePos)]) / 2; + outputs[OUTL_OUTPUT].value = 10.0f * (pSampleData[(unsigned int)floor(samplePos)] + pSampleData[(unsigned int)floor(samplePos)+1]) / 2; + outputs[OUTR_OUTPUT].value = 10.0f * (pSampleData[(unsigned int)floor(samplePos)] + pSampleData[(unsigned int)floor(samplePos)+1]) / 2; } } - + mylock.unlock(); //shift samplePos if (trigMode == 0) { if (readMode != 1) - samplePos = samplePos + speed; + samplePos = samplePos + speed * channels; else - samplePos = samplePos - speed; + samplePos = samplePos - speed * channels; //manage eof readMode - if ((readMode == 0) && (samplePos >= audioFile.getNumSamplesPerChannel())) + if ((readMode == 0) && (samplePos >= totalSampleCount)) play = false; else if ((readMode == 1) && (samplePos <=0)) play = false; - else if ((readMode == 2) && (samplePos >= audioFile.getNumSamplesPerChannel())) - samplePos = clamp((int)(inputs[POS_INPUT].value*audioFile.getNumSamplesPerChannel()/10), 0 , audioFile.getNumSamplesPerChannel() -1); + else if ((readMode == 2) && (samplePos >= totalSampleCount)) + samplePos = clamp((int)(inputs[POS_INPUT].value * totalSampleCount / 10), 0 , totalSampleCount -1); } else if (trigMode == 2) { if (readMode != 1) - samplePos = samplePos + speed; + samplePos = samplePos + speed * channels; else - samplePos = samplePos - speed; + samplePos = samplePos - speed * channels; //update diplay slices displaySlices = "|" + std::to_string(nbSlices) + "|"; //manage eof readMode - if ((readMode == 0) && ((samplePos >= (sliceIndex+1) * sliceLength) || (samplePos >= audioFile.getNumSamplesPerChannel()))) + if ((readMode == 0) && ((samplePos >= (sliceIndex+1) * sliceLength) || (samplePos >= totalSampleCount))) play = false; - if ((readMode == 1) && ((samplePos <= (sliceIndex) * sliceLength) || (samplePos <=0))) + if ((readMode == 1) && ((samplePos <= (sliceIndex) * sliceLength) || (samplePos <= 0))) play = false; - if ((readMode == 2) && ((samplePos >= (sliceIndex+1) * sliceLength) || (samplePos >= audioFile.getNumSamplesPerChannel()))) - samplePos = clamp(sliceIndex*sliceLength, 0 , audioFile.getNumSamplesPerChannel()); + if ((readMode == 2) && ((samplePos >= (sliceIndex+1) * sliceLength) || (samplePos >= totalSampleCount))) + samplePos = clamp(sliceIndex*sliceLength, 0 , totalSampleCount); } } - else if (samplePos == audioFile.getNumSamplesPerChannel()) + else if (samplePos == totalSampleCount) play = false; } } @@ -266,13 +280,13 @@ struct OUAIVEDisplay : TransparentWidget { { nvgBeginPath(vg); nvgStrokeWidth(vg, 2); - nvgMoveTo(vg, (int)(module->samplePos * 125 / module->audioFile.getNumSamplesPerChannel()) , 70); - nvgLineTo(vg, (int)(module->samplePos * 125 / module->audioFile.getNumSamplesPerChannel()) , 150); + nvgMoveTo(vg, (int)(module->samplePos * 125 / module->totalSampleCount) , 70); + nvgLineTo(vg, (int)(module->samplePos * 125 / module->totalSampleCount) , 150); nvgClosePath(vg); } nvgStroke(vg); - if (module->audioFile.getNumChannels() == 1) { + if (module->channels == 1) { // Draw ref line nvgStrokeColor(vg, nvgRGBA(0xff, 0xff, 0xff, 0x30)); nvgStrokeWidth(vg, 1); @@ -378,8 +392,8 @@ struct OUAIVEDisplay : TransparentWidget { { nvgBeginPath(vg); nvgStrokeWidth(vg, 1); - nvgMoveTo(vg, (int)(i * module->sliceLength * 125 / module->audioFile.getNumSamplesPerChannel()) , 70); - nvgLineTo(vg, (int)(i * module->sliceLength * 125 / module->audioFile.getNumSamplesPerChannel()) , 150); + nvgMoveTo(vg, (int)(i * module->sliceLength * 125 / module->totalSampleCount) , 70); + nvgLineTo(vg, (int)(i * module->sliceLength * 125 / module->totalSampleCount) , 150); nvgClosePath(vg); } nvgStroke(vg); @@ -468,6 +482,6 @@ Menu *OUAIVEWidget::createContextMenu() { using namespace rack_plugin_Bidoo; RACK_PLUGIN_MODEL_INIT(Bidoo, OUAIVE) { - Model *modelOUAIVE = Model::create("Bidoo","OUAIve", "OUAIve player", SAMPLER_TAG); + Model *modelOUAIVE = Model::create("Bidoo","OUAIve", "OUAIve player", SAMPLER_TAG, GRANULAR_TAG); return modelOUAIVE; } diff --git a/plugins/community/repos/Bidoo/src/RABBIT.cpp b/plugins/community/repos/Bidoo/src/RABBIT.cpp new file mode 100644 index 00000000..a25ee416 --- /dev/null +++ b/plugins/community/repos/Bidoo/src/RABBIT.cpp @@ -0,0 +1,154 @@ +#include "Bidoo.hpp" +#include "BidooComponents.hpp" +#include "dsp/digital.hpp" + +using namespace std; + +namespace rack_plugin_Bidoo { + +struct RABBIT : Module { + enum ParamIds { + BITOFF_PARAM, + BITREV_PARAM = BITOFF_PARAM + 8, + NUM_PARAMS = BITREV_PARAM + 8 + }; + enum InputIds { + L_INPUT, + R_INPUT, + BITOFF_INPUT, + BITREV_INPUT = BITOFF_INPUT + 8, + NUM_INPUTS = BITREV_INPUT + 8 + }; + enum OutputIds { + L_OUTPUT, + R_OUTPUT, + NUM_OUTPUTS + }; + enum LightIds { + BITOFF_LIGHTS, + BITREV_LIGHTS = BITOFF_LIGHTS + 8, + NUM_LIGHTS = BITREV_LIGHTS + 8 + }; + + SchmittTrigger bitOffTrigger[8], bitRevTrigger[8]; + + bool bitOff[8]; + bool bitRev[8]; + + RABBIT() : Module(NUM_PARAMS, NUM_INPUTS, NUM_OUTPUTS, NUM_LIGHTS) { + memset(&bitOff,0,8*sizeof(bool)); + memset(&bitRev,0,8*sizeof(bool)); + } + + ~RABBIT() { + } + + json_t *toJson() override { + json_t *rootJ = json_object(); + for (int i=0; i<8; i++) { + json_object_set_new(rootJ, ("bitOff" + to_string(i)).c_str(), json_boolean(bitOff[i])); + json_object_set_new(rootJ, ("bitRev" + to_string(i)).c_str(), json_boolean(bitRev[i])); + } + return rootJ; + } + + void fromJson(json_t *rootJ) override { + for (int i=0; i<8; i++) { + json_t *jbitOff = json_object_get(rootJ, ("bitOff" + to_string(i)).c_str()); + if (jbitOff) { + bitOff[i] = json_is_true(jbitOff) ? 1 : 0; + } + json_t *jbitRev = json_object_get(rootJ, ("bitRev" + to_string(i)).c_str()); + if (jbitRev) { + bitRev[i] = json_is_true(jbitRev) ? 1 : 0; + } + } + } + + void step() override; +}; + + +void RABBIT::step() { + float in_L = clamp(inputs[L_INPUT].value,-10.0f,10.0f); + float in_R = clamp(inputs[R_INPUT].value,-10.0f,10.0f); + + in_L = roundf(clamp(in_L / 20.0f + 0.5f, 0.0f, 1.0f) * 255); + in_R = roundf(clamp(in_R / 20.0f + 0.5f, 0.0f, 1.0f) * 255); + + unsigned char red_L = in_L; + unsigned char red_R = in_R; + + for (int i = 0 ; i < 8 ; i++ ) { + + if (bitOffTrigger[i].process(params[BITOFF_PARAM+i].value + inputs[BITOFF_INPUT+i].value)) + { + bitOff[i] = !bitOff[i]; + } + + if (bitRevTrigger[i].process(params[BITREV_PARAM+i].value + inputs[BITREV_INPUT+i].value)) + { + bitRev[i] = !bitRev[i]; + } + + if (bitOff[i]) { + red_L &= ~(1 << i); + red_R &= ~(1 << i); + } + else { + if (bitRev[i]) { + red_L ^= ~(1 << i); + red_R ^= ~(1 << i); + } + } + + lights[BITOFF_LIGHTS + i].value = bitOff[i] ? 1.0f : 0.0f; + lights[BITREV_LIGHTS + i].value = bitRev[i] ? 1.0f : 0.0f; + } + + outputs[L_OUTPUT].value = clamp(((((float)red_L/255.0f))-0.5f)*20.0f,-10.0f,10.0f); + outputs[R_OUTPUT].value = clamp(((((float)red_R/255.0f))-0.5f)*20.0f,-10.0f,10.0f); +} + +template +struct RabbitLight : BASE { + RabbitLight() { + this->box.size = mm2px(Vec(6.0f, 6.0f)); + } +}; + +struct RABBITWidget : ModuleWidget { + RABBITWidget(RABBIT *module) : ModuleWidget(module) { + setPanel(SVG::load(assetPlugin(plugin, "res/RABBIT.svg"))); + + addChild(Widget::create(Vec(RACK_GRID_WIDTH, 0))); + addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, 0))); + addChild(Widget::create(Vec(RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); + addChild(Widget::create(Vec(box.size.x - 2 * RACK_GRID_WIDTH, RACK_GRID_HEIGHT - RACK_GRID_WIDTH))); + + for (int i = 0; i<8; i++) { + addParam(ParamWidget::create(Vec(27.0f, 50.0f + 32.0f * i), module, RABBIT::BITOFF_PARAM + i, 0.0f, 1.0f, 0.0f)); + addChild(ModuleLightWidget::create>(Vec(29.0f, 52.0f + 32.0f * i), module, RABBIT::BITOFF_LIGHTS + i)); + + addInput(Port::create(Vec(8.0f, 54.0f + 32.0f * i), Port::INPUT, module, RABBIT::BITOFF_INPUT + i)); + addInput(Port::create(Vec(83.0f, 54.0f + 32.0f * i), Port::INPUT, module, RABBIT::BITREV_INPUT + i)); + + addParam(ParamWidget::create(Vec(57.0f, 50.0f + 32.0f * i), module, RABBIT::BITREV_PARAM + i, 0.0f, 1.0f, 0.0f)); + addChild(ModuleLightWidget::create>(Vec(59.0f, 52.0f + 32.0f * i), module, RABBIT::BITREV_LIGHTS + i)); + } + + addInput(Port::create(Vec(24.0f, 319.0f), Port::INPUT, module, RABBIT::L_INPUT)); + addInput(Port::create(Vec(24.0f, 339.0f), Port::INPUT, module, RABBIT::R_INPUT)); + addOutput(Port::create(Vec(78.0f, 319.0f),Port::OUTPUT, module, RABBIT::L_OUTPUT)); + addOutput(Port::create(Vec(78.0f, 339.0f),Port::OUTPUT, module, RABBIT::R_OUTPUT)); + } +}; + +} // namespace rack_plugin_Bidoo + +using namespace rack_plugin_Bidoo; + +RACK_PLUGIN_MODEL_INIT(Bidoo, RABBIT) { + Model *modelRABBIT = Model::create("Bidoo", "rabBIT", "rabBIT bit crusher", EFFECT_TAG, DIGITAL_TAG, DISTORTION_TAG); + return modelRABBIT; +} diff --git a/plugins/community/repos/Bidoo/src/ZINC.cpp b/plugins/community/repos/Bidoo/src/ZINC.cpp index 345fda6e..e48857b5 100644 --- a/plugins/community/repos/Bidoo/src/ZINC.cpp +++ b/plugins/community/repos/Bidoo/src/ZINC.cpp @@ -158,6 +158,6 @@ void ZINCWidget::step() { using namespace rack_plugin_Bidoo; RACK_PLUGIN_MODEL_INIT(Bidoo, ZINC) { - Model *modelZINC = Model::create("Bidoo", "ziNC", "ziNC vocoder", EFFECT_TAG); + Model *modelZINC = Model::create("Bidoo", "ziNC", "ziNC vocoder", EFFECT_TAG, VOCODER_TAG); return modelZINC; } diff --git a/plugins/community/repos/Bidoo/src/dep/dr_wav/dr_wav.h b/plugins/community/repos/Bidoo/src/dep/dr_wav/dr_wav.h new file mode 100644 index 00000000..748294eb --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/dr_wav/dr_wav.h @@ -0,0 +1,3724 @@ +// WAV audio loader and writer. Public domain. See "unlicense" statement at the end of this file. +// dr_wav - v0.8.1 - 2018-06-29 +// +// David Reid - mackron@gmail.com + +// USAGE +// +// This is a single-file library. To use it, do something like the following in one .c file. +// #define DR_WAV_IMPLEMENTATION +// #include "dr_wav.h" +// +// You can then #include this file in other parts of the program as you would with any other header file. Do something +// like the following to read audio data: +// +// drwav wav; +// if (!drwav_init_file(&wav, "my_song.wav")) { +// // Error opening WAV file. +// } +// +// drwav_int32* pDecodedInterleavedSamples = malloc(wav.totalSampleCount * sizeof(drwav_int32)); +// size_t numberOfSamplesActuallyDecoded = drwav_read_s32(&wav, wav.totalSampleCount, pDecodedInterleavedSamples); +// +// ... +// +// drwav_uninit(&wav); +// +// You can also use drwav_open() to allocate and initialize the loader for you: +// +// drwav* pWav = drwav_open_file("my_song.wav"); +// if (pWav == NULL) { +// // Error opening WAV file. +// } +// +// ... +// +// drwav_close(pWav); +// +// If you just want to quickly open and read the audio data in a single operation you can do something like this: +// +// unsigned int channels; +// unsigned int sampleRate; +// drwav_uint64 totalSampleCount; +// float* pSampleData = drwav_open_and_read_file_s32("my_song.wav", &channels, &sampleRate, &totalSampleCount); +// if (pSampleData == NULL) { +// // Error opening and reading WAV file. +// } +// +// ... +// +// drwav_free(pSampleData); +// +// The examples above use versions of the API that convert the audio data to a consistent format (32-bit signed PCM, in +// this case), but you can still output the audio data in its internal format (see notes below for supported formats): +// +// size_t samplesRead = drwav_read(&wav, wav.totalSampleCount, pDecodedInterleavedSamples); +// +// You can also read the raw bytes of audio data, which could be useful if dr_wav does not have native support for +// a particular data format: +// +// size_t bytesRead = drwav_read_raw(&wav, bytesToRead, pRawDataBuffer); +// +// +// dr_wav has seamless support the Sony Wave64 format. The decoder will automatically detect it and it should Just Work +// without any manual intervention. +// +// +// dr_wav can also be used to output WAV files. This does not currently support compressed formats. To use this, look at +// drwav_open_write(), drwav_open_file_write(), etc. Use drwav_write() to write samples, or drwav_write_raw() to write +// raw data in the "data" chunk. +// +// drwav_data_format format; +// format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64. +// format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes. +// format.channels = 2; +// format.sampleRate = 44100; +// format.bitsPerSample = 16; +// drwav* pWav = drwav_open_file_write("data/recording.wav", &format); +// +// ... +// +// drwav_uint64 samplesWritten = drwav_write(pWav, sampleCount, pSamples); +// +// +// +// OPTIONS +// #define these options before including this file. +// +// #define DR_WAV_NO_CONVERSION_API +// Disables conversion APIs such as drwav_read_f32() and drwav_s16_to_f32(). +// +// #define DR_WAV_NO_STDIO +// Disables drwav_open_file(), drwav_open_file_write(), etc. +// +// +// +// QUICK NOTES +// - Samples are always interleaved. +// - The default read function does not do any data conversion. Use drwav_read_f32() to read and convert audio data +// to IEEE 32-bit floating point samples, drwav_read_s32() to read samples as signed 32-bit PCM and drwav_read_s16() +// to read samples as signed 16-bit PCM. Tested and supported internal formats include the following: +// - Unsigned 8-bit PCM +// - Signed 12-bit PCM +// - Signed 16-bit PCM +// - Signed 24-bit PCM +// - Signed 32-bit PCM +// - IEEE 32-bit floating point +// - IEEE 64-bit floating point +// - A-law and u-law +// - Microsoft ADPCM +// - IMA ADPCM (DVI, format code 0x11) +// - dr_wav will try to read the WAV file as best it can, even if it's not strictly conformant to the WAV format. + + +#ifndef dr_wav_h +#define dr_wav_h + +#include + +#if defined(_MSC_VER) && _MSC_VER < 1600 +typedef signed char drwav_int8; +typedef unsigned char drwav_uint8; +typedef signed short drwav_int16; +typedef unsigned short drwav_uint16; +typedef signed int drwav_int32; +typedef unsigned int drwav_uint32; +typedef signed __int64 drwav_int64; +typedef unsigned __int64 drwav_uint64; +#else +#include +typedef int8_t drwav_int8; +typedef uint8_t drwav_uint8; +typedef int16_t drwav_int16; +typedef uint16_t drwav_uint16; +typedef int32_t drwav_int32; +typedef uint32_t drwav_uint32; +typedef int64_t drwav_int64; +typedef uint64_t drwav_uint64; +#endif +typedef drwav_uint8 drwav_bool8; +typedef drwav_uint32 drwav_bool32; +#define DRWAV_TRUE 1 +#define DRWAV_FALSE 0 + +#ifdef __cplusplus +extern "C" { +#endif + +// Common data formats. +#define DR_WAVE_FORMAT_PCM 0x1 +#define DR_WAVE_FORMAT_ADPCM 0x2 +#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3 +#define DR_WAVE_FORMAT_ALAW 0x6 +#define DR_WAVE_FORMAT_MULAW 0x7 +#define DR_WAVE_FORMAT_DVI_ADPCM 0x11 +#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE + +typedef enum +{ + drwav_seek_origin_start, + drwav_seek_origin_current +} drwav_seek_origin; + +typedef enum +{ + drwav_container_riff, + drwav_container_w64 +} drwav_container; + +// Callback for when data is read. Return value is the number of bytes actually read. +// +// pUserData [in] The user data that was passed to drwav_init(), drwav_open() and family. +// pBufferOut [out] The output buffer. +// bytesToRead [in] The number of bytes to read. +// +// Returns the number of bytes actually read. +// +// A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +// either the entire bytesToRead is filled or you have reached the end of the stream. +typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +// Callback for when data is written. Returns value is the number of bytes actually written. +// +// pUserData [in] The user data that was passed to drwav_init_write(), drwav_open_write() and family. +// pData [out] A pointer to the data to write. +// bytesToWrite [in] The number of bytes to write. +// +// Returns the number of bytes actually written. +// +// If the return value differs from bytesToWrite, it indicates an error. +typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite); + +// Callback for when data needs to be seeked. +// +// pUserData [in] The user data that was passed to drwav_init(), drwav_open() and family. +// offset [in] The number of bytes to move, relative to the origin. Will never be negative. +// origin [in] The origin of the seek - the current position or the start of the stream. +// +// Returns whether or not the seek was successful. +// +// Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which +// will be either drwav_seek_origin_start or drwav_seek_origin_current. +typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin); + +// Structure for internal use. Only used for loaders opened with drwav_open_memory(). +typedef struct +{ + const drwav_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drwav__memory_stream; + +// Structure for internal use. Only used for writers opened with drwav_open_memory_write(). +typedef struct +{ + void** ppData; + size_t* pDataSize; + size_t dataSize; + size_t dataCapacity; + size_t currentWritePos; +} drwav__memory_stream_write; + +typedef struct +{ + drwav_container container; // RIFF, W64. + drwav_uint32 format; // DR_WAVE_FORMAT_* + drwav_uint32 channels; + drwav_uint32 sampleRate; + drwav_uint32 bitsPerSample; +} drwav_data_format; + +typedef struct +{ + // The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications + // that require support for data formats not natively supported by dr_wav. + drwav_uint16 formatTag; + + // The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. + drwav_uint16 channels; + + // The sample rate. Usually set to something like 44100. + drwav_uint32 sampleRate; + + // Average bytes per second. You probably don't need this, but it's left here for informational purposes. + drwav_uint32 avgBytesPerSec; + + // Block align. This is equal to the number of channels * bytes per sample. + drwav_uint16 blockAlign; + + // Bits per sample. + drwav_uint16 bitsPerSample; + + // The size of the extended data. Only used internally for validation, but left here for informational purposes. + drwav_uint16 extendedSize; + + // The number of valid bits per sample. When is equal to WAVE_FORMAT_EXTENSIBLE, + // is always rounded up to the nearest multiple of 8. This variable contains information about exactly how + // many bits a valid per sample. Mainly used for informational purposes. + drwav_uint16 validBitsPerSample; + + // The channel mask. Not used at the moment. + drwav_uint32 channelMask; + + // The sub-format, exactly as specified by the wave file. + drwav_uint8 subFormat[16]; +} drwav_fmt; + +typedef struct +{ + // A pointer to the function to call when more data is needed. + drwav_read_proc onRead; + + // A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. + drwav_write_proc onWrite; + + // A pointer to the function to call when the wav file needs to be seeked. + drwav_seek_proc onSeek; + + // The user data to pass to callbacks. + void* pUserData; + + + // Whether or not the WAV file is formatted as a standard RIFF file or W64. + drwav_container container; + + + // Structure containing format information exactly as specified by the wav file. + drwav_fmt fmt; + + // The sample rate. Will be set to something like 44100. + drwav_uint32 sampleRate; + + // The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. + drwav_uint16 channels; + + // The bits per sample. Will be set to something like 16, 24, etc. + drwav_uint16 bitsPerSample; + + // The number of bytes per sample. + drwav_uint16 bytesPerSample; + + // Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). + drwav_uint16 translatedFormatTag; + + // The total number of samples making up the audio data. Use * to calculate + // the required size of a buffer to hold the entire audio data. + drwav_uint64 totalSampleCount; + + + // The size in bytes of the data chunk. + drwav_uint64 dataChunkDataSize; + + // The position in the stream of the first byte of the data chunk. This is used for seeking. + drwav_uint64 dataChunkDataPos; + + // The number of bytes remaining in the data chunk. + drwav_uint64 bytesRemaining; + + + // Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always + // set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation. + drwav_uint64 dataChunkDataSizeTargetWrite; + + // Keeps track of whether or not the wav writer was initialized in sequential mode. + drwav_bool32 isSequentialWrite; + + + // A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_open_memory(). + drwav__memory_stream memoryStream; + drwav__memory_stream_write memoryStreamWrite; + + // Generic data for compressed formats. This data is shared across all block-compressed formats. + struct + { + drwav_uint64 iCurrentSample; // The index of the next sample that will be read by drwav_read_*(). This is used with "totalSampleCount" to ensure we don't read excess samples at the end of the last block. + } compressed; + + // Microsoft ADPCM specific data. + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_uint16 predictor[2]; + drwav_int32 delta[2]; + drwav_int32 cachedSamples[4]; // Samples are stored in this cache during decoding. + drwav_uint32 cachedSampleCount; + drwav_int32 prevSamples[2][2]; // The previous 2 samples for each channel (2 channels at most). + } msadpcm; + + // IMA ADPCM specific data. + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_int32 predictor[2]; + drwav_int32 stepIndex[2]; + drwav_int32 cachedSamples[16]; // Samples are stored in this cache during decoding. + drwav_uint32 cachedSampleCount; + } ima; +} drwav; + + +// Initializes a pre-allocated drwav object. +// +// onRead [in] The function to call when data needs to be read from the client. +// onSeek [in] The function to call when the read position of the client data needs to move. +// pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +// +// Returns true if successful; false otherwise. +// +// Close the loader with drwav_uninit(). +// +// This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +// to open the stream from a file or from a block of memory respectively. +// +// If you want dr_wav to manage the memory allocation for you, consider using drwav_open() instead. This will allocate +// a drwav object on the heap and return a pointer to it. +// +// See also: drwav_init_file(), drwav_init_memory(), drwav_uninit() +drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData); + +// Initializes a pre-allocated drwav object for writing. +// +// onWrite [in] The function to call when data needs to be written. +// onSeek [in] The function to call when the write position needs to move. +// pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. +// +// Returns true if successful; false otherwise. +// +// Close the writer with drwav_uninit(). +// +// This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +// to open the stream from a file or from a block of memory respectively. +// +// If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform +// a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek. +// +// If you want dr_wav to manage the memory allocation for you, consider using drwav_open() instead. This will allocate +// a drwav object on the heap and return a pointer to it. +// +// See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit() +drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData); + +// Uninitializes the given drwav object. +// +// Use this only for objects initialized with drwav_init(). +void drwav_uninit(drwav* pWav); + + +// Opens a wav file using the given callbacks. +// +// onRead [in] The function to call when data needs to be read from the client. +// onSeek [in] The function to call when the read position of the client data needs to move. +// pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +// +// Returns null on error. +// +// Close the loader with drwav_close(). +// +// You can also use drwav_open_file() and drwav_open_memory() to open the stream from a file or from a block of +// memory respectively. +// +// This is different from drwav_init() in that it will allocate the drwav object for you via DRWAV_MALLOC() before +// initializing it. +// +// See also: drwav_open_file(), drwav_open_memory(), drwav_close() +drwav* drwav_open(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData); + +// Opens a wav file for writing using the given callbacks. +// +// onWrite [in] The function to call when data needs to be written. +// onSeek [in] The function to call when the write position needs to move. +// pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. +// +// Returns null on error. +// +// Close the loader with drwav_close(). +// +// You can also use drwav_open_file_write() and drwav_open_memory_write() to open the stream from a file or from a block +// of memory respectively. +// +// This is different from drwav_init_write() in that it will allocate the drwav object for you via DRWAV_MALLOC() before +// initializing it. +// +// See also: drwav_open_file_write(), drwav_open_memory_write(), drwav_close() +drwav* drwav_open_write(const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav* drwav_open_write_sequential(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData); + +// Uninitializes and deletes the the given drwav object. +// +// Use this only for objects created with drwav_open(). +void drwav_close(drwav* pWav); + + +// Reads raw audio data. +// +// This is the lowest level function for reading audio data. It simply reads the given number of +// bytes of the raw internal sample data. +// +// Consider using drwav_read_s16(), drwav_read_s32() or drwav_read_f32() for reading sample data in +// a consistent format. +// +// Returns the number of bytes actually read. +size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut); + +// Reads a chunk of audio data in the native internal format. +// +// This is typically the most efficient way to retrieve audio data, but it does not do any format +// conversions which means you'll need to convert the data manually if required. +// +// If the return value is less than it means the end of the file has been reached or +// you have requested more samples than can possibly fit in the output buffer. +// +// This function will only work when sample data is of a fixed size and uncompressed. If you are +// using a compressed format consider using drwav_read_raw() or drwav_read_s16/s32/f32/etc(). +drwav_uint64 drwav_read(drwav* pWav, drwav_uint64 samplesToRead, void* pBufferOut); + +// Seeks to the given sample. +// +// Returns true if successful; false otherwise. +drwav_bool32 drwav_seek_to_sample(drwav* pWav, drwav_uint64 sample); + + +// Writes raw audio data. +// +// Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error. +size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData); + +// Writes audio data based on sample counts. +// +// Returns the number of samples written. +drwav_uint64 drwav_write(drwav* pWav, drwav_uint64 samplesToWrite, const void* pData); + + + +//// Conversion Utilities //// +#ifndef DR_WAV_NO_CONVERSION_API + +// Reads a chunk of audio data and converts it to signed 16-bit PCM samples. +// +// Returns the number of samples actually read. +// +// If the return value is less than it means the end of the file has been reached. +drwav_uint64 drwav_read_s16(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); + +// Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. +void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. +void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. +void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount); + +// Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. +void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount); + +// Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. +void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount); + +// Low-level function for converting A-law samples to signed 16-bit PCM samples. +void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting u-law samples to signed 16-bit PCM samples. +void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +// Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples. +// +// Returns the number of samples actually read. +// +// If the return value is less than it means the end of the file has been reached. +drwav_uint64 drwav_read_f32(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut); + +// Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. +void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. +void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount); + +// Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. +void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. +void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount); + +// Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. +void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount); + +// Low-level function for converting A-law samples to IEEE 32-bit floating point samples. +void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting u-law samples to IEEE 32-bit floating point samples. +void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +// Reads a chunk of audio data and converts it to signed 32-bit PCM samples. +// +// Returns the number of samples actually read. +// +// If the return value is less than it means the end of the file has been reached. +drwav_uint64 drwav_read_s32(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut); + +// Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. +void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. +void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount); + +// Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. +void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. +void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount); + +// Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. +void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount); + +// Low-level function for converting A-law samples to signed 32-bit PCM samples. +void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +// Low-level function for converting u-law samples to signed 32-bit PCM samples. +void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +#endif //DR_WAV_NO_CONVERSION_API + + +//// High-Level Convenience Helpers //// + +#ifndef DR_WAV_NO_STDIO + +// Helper for initializing a wave file using stdio. +// +// This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +// objects because the operating system may restrict the number of file handles an application can have open at +// any given time. +drwav_bool32 drwav_init_file(drwav* pWav, const char* filename); + +// Helper for initializing a wave file for writing using stdio. +// +// This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +// objects because the operating system may restrict the number of file handles an application can have open at +// any given time. +drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat); +drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +// Helper for opening a wave file using stdio. +// +// This holds the internal FILE object until drwav_close() is called. Keep this in mind if you're caching drwav +// objects because the operating system may restrict the number of file handles an application can have open at +// any given time. +drwav* drwav_open_file(const char* filename); + +// Helper for opening a wave file for writing using stdio. +// +// This holds the internal FILE object until drwav_close() is called. Keep this in mind if you're caching drwav +// objects because the operating system may restrict the number of file handles an application can have open at +// any given time. +drwav* drwav_open_file_write(const char* filename, const drwav_data_format* pFormat); +drwav* drwav_open_file_write_sequential(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +#endif //DR_WAV_NO_STDIO + +// Helper for initializing a loader from a pre-allocated memory buffer. +// +// This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +// the lifetime of the drwav object. +// +// The buffer should contain the contents of the entire wave file, not just the sample data. +drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize); + +// Helper for initializing a writer which outputs data to a memory buffer. +// +// dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). +// +// The buffer will remain allocated even after drwav_uninit() is called. Indeed, the buffer should not be +// considered valid until after drwav_uninit() has been called anyway. +drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat); +drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +// Helper for opening a loader from a pre-allocated memory buffer. +// +// This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +// the lifetime of the drwav object. +// +// The buffer should contain the contents of the entire wave file, not just the sample data. +drwav* drwav_open_memory(const void* data, size_t dataSize); + +// Helper for opening a writer which outputs data to a memory buffer. +// +// dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). +// +// The buffer will remain allocated even after drwav_close() is called. Indeed, the buffer should not be +// considered valid until after drwav_close() has been called anyway. +drwav* drwav_open_memory_write(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat); +drwav* drwav_open_memory_write_sequential(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + + +#ifndef DR_WAV_NO_CONVERSION_API +// Opens and reads a wav file in a single operation. +drwav_int16* drwav_open_and_read_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_and_read_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_and_read_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#ifndef DR_WAV_NO_STDIO +// Opens and decodes a wav file in a single operation. +drwav_int16* drwav_open_and_read_file_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_and_read_file_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_and_read_file_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#endif + +// Opens and decodes a wav file from a block of memory in a single operation. +drwav_int16* drwav_open_and_read_memory_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +float* drwav_open_and_read_memory_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +drwav_int32* drwav_open_and_read_memory_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount); +#endif + +// Frees data that was allocated internally by dr_wav. +void drwav_free(void* pDataReturnedByOpenAndRead); + +#ifdef __cplusplus +} +#endif +#endif // dr_wav_h + + +///////////////////////////////////////////////////// +// +// IMPLEMENTATION +// +///////////////////////////////////////////////////// + +#ifdef DR_WAV_IMPLEMENTATION +#include +#include // For memcpy(), memset() +#include // For INT_MAX + +#ifndef DR_WAV_NO_STDIO +#include +#endif + +// Standard library stuff. +#ifndef DRWAV_ASSERT +#include +#define DRWAV_ASSERT(expression) assert(expression) +#endif +#ifndef DRWAV_MALLOC +#define DRWAV_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRWAV_REALLOC +#define DRWAV_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRWAV_FREE +#define DRWAV_FREE(p) free((p)) +#endif +#ifndef DRWAV_COPY_MEMORY +#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRWAV_ZERO_MEMORY +#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif + +#define drwav_countof(x) (sizeof(x) / sizeof(x[0])) +#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) +#define drwav_min(a, b) (((a) < (b)) ? (a) : (b)) +#define drwav_max(a, b) (((a) > (b)) ? (a) : (b)) +#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x)))) + +#define drwav_assert DRWAV_ASSERT +#define drwav_copy_memory DRWAV_COPY_MEMORY +#define drwav_zero_memory DRWAV_ZERO_MEMORY + + +#define DRWAV_MAX_SIMD_VECTOR_SIZE 64 // 64 for AVX-512 in the future. + +#ifdef _MSC_VER +#define DRWAV_INLINE __forceinline +#else +#ifdef __GNUC__ +#define DRWAV_INLINE inline __attribute__((always_inline)) +#else +#define DRWAV_INLINE inline +#endif +#endif + +// I couldn't figure out where SIZE_MAX was defined for VC6. If anybody knows, let me know. +#if defined(_MSC_VER) && _MSC_VER <= 1200 + #if defined(_WIN64) + #define SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define SIZE_MAX 0xFFFFFFFF + #endif +#endif + +static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; // 66666972-912E-11CF-A5D6-28DB04C10000 +static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; // 65766177-ACF3-11D3-8CD1-00C04F8EDB8A +static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; // 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A +static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; // 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A +static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; // 74636166-ACF3-11D3-8CD1-00C04F8EDB8A +static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; // 61746164-ACF3-11D3-8CD1-00C04F8EDB8A + +static DRWAV_INLINE drwav_bool32 drwav__guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + const drwav_uint32* a32 = (const drwav_uint32*)a; + const drwav_uint32* b32 = (const drwav_uint32*)b; + + return + a32[0] == b32[0] && + a32[1] == b32[1] && + a32[2] == b32[2] && + a32[3] == b32[3]; +} + +static DRWAV_INLINE drwav_bool32 drwav__fourcc_equal(const unsigned char* a, const char* b) +{ + return + a[0] == b[0] && + a[1] == b[1] && + a[2] == b[2] && + a[3] == b[3]; +} + + + +static DRWAV_INLINE int drwav__is_little_endian() +{ + int n = 1; + return (*(char*)&n) == 1; +} + +static DRWAV_INLINE unsigned short drwav__bytes_to_u16(const unsigned char* data) +{ + if (drwav__is_little_endian()) { + return (data[0] << 0) | (data[1] << 8); + } else { + return (data[1] << 0) | (data[0] << 8); + } +} + +static DRWAV_INLINE short drwav__bytes_to_s16(const unsigned char* data) +{ + return (short)drwav__bytes_to_u16(data); +} + +static DRWAV_INLINE unsigned int drwav__bytes_to_u32(const unsigned char* data) +{ + if (drwav__is_little_endian()) { + return (data[0] << 0) | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + } else { + return (data[3] << 0) | (data[2] << 8) | (data[1] << 16) | (data[0] << 24); + } +} + +static DRWAV_INLINE drwav_uint64 drwav__bytes_to_u64(const unsigned char* data) +{ + if (drwav__is_little_endian()) { + return + ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) | + ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56); + } else { + return + ((drwav_uint64)data[7] << 0) | ((drwav_uint64)data[6] << 8) | ((drwav_uint64)data[5] << 16) | ((drwav_uint64)data[4] << 24) | + ((drwav_uint64)data[3] << 32) | ((drwav_uint64)data[2] << 40) | ((drwav_uint64)data[1] << 48) | ((drwav_uint64)data[0] << 56); + } +} + +static DRWAV_INLINE void drwav__bytes_to_guid(const unsigned char* data, drwav_uint8* guid) +{ + for (int i = 0; i < 16; ++i) { + guid[i] = data[i]; + } +} + + +static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag) +{ + return + formatTag == DR_WAVE_FORMAT_ADPCM || + formatTag == DR_WAVE_FORMAT_DVI_ADPCM; +} + + +drwav_uint64 drwav_read_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +drwav_uint64 drwav_read_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); +drwav* drwav_open_write__internal(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData); + +typedef struct +{ + union + { + drwav_uint8 fourcc[4]; + drwav_uint8 guid[16]; + } id; + + // The size in bytes of the chunk. + drwav_uint64 sizeInBytes; + + // RIFF = 2 byte alignment. + // W64 = 8 byte alignment. + unsigned int paddingSize; + +} drwav__chunk_header; + +static drwav_bool32 drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav__chunk_header* pHeaderOut) +{ + if (container == drwav_container_riff) { + if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) { + return DRWAV_FALSE; + } + + unsigned char sizeInBytes[4]; + if (onRead(pUserData, sizeInBytes, 4) != 4) { + return DRWAV_FALSE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u32(sizeInBytes); + pHeaderOut->paddingSize = (unsigned int)(pHeaderOut->sizeInBytes % 2); + *pRunningBytesReadOut += 8; + } else { + if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) { + return DRWAV_FALSE; + } + + unsigned char sizeInBytes[8]; + if (onRead(pUserData, sizeInBytes, 8) != 8) { + return DRWAV_FALSE; + } + + pHeaderOut->sizeInBytes = drwav__bytes_to_u64(sizeInBytes) - 24; // <-- Subtract 24 because w64 includes the size of the header. + pHeaderOut->paddingSize = (unsigned int)(pHeaderOut->sizeInBytes % 8); + *pRunningBytesReadOut += 24; + } + + return DRWAV_TRUE; +} + +static drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + drwav_uint64 bytesRemainingToSeek = offset; + while (bytesRemainingToSeek > 0) { + if (bytesRemainingToSeek > 0x7FFFFFFF) { + if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek -= 0x7FFFFFFF; + } else { + if (!onSeek(pUserData, (int)bytesRemainingToSeek, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek = 0; + } + } + + return DRWAV_TRUE; +} + + +static drwav_bool32 drwav__read_fmt(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_fmt* fmtOut) +{ + drwav__chunk_header header; + if (!drwav__read_chunk_header(onRead, pUserData, container, pRunningBytesReadOut, &header)) { + return DRWAV_FALSE; + } + + + // Skip non-fmt chunks. + if ((container == drwav_container_riff && !drwav__fourcc_equal(header.id.fourcc, "fmt ")) || (container == drwav_container_w64 && !drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT))) { + if (!drwav__seek_forward(onSeek, header.sizeInBytes + header.paddingSize, pUserData)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.sizeInBytes + header.paddingSize; + + return drwav__read_fmt(onRead, onSeek, pUserData, container, pRunningBytesReadOut, fmtOut); + } + + + // Validation. + if (container == drwav_container_riff) { + if (!drwav__fourcc_equal(header.id.fourcc, "fmt ")) { + return DRWAV_FALSE; + } + } else { + if (!drwav__guid_equal(header.id.guid, drwavGUID_W64_FMT)) { + return DRWAV_FALSE; + } + } + + + unsigned char fmt[16]; + if (onRead(pUserData, fmt, sizeof(fmt)) != sizeof(fmt)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += sizeof(fmt); + + fmtOut->formatTag = drwav__bytes_to_u16(fmt + 0); + fmtOut->channels = drwav__bytes_to_u16(fmt + 2); + fmtOut->sampleRate = drwav__bytes_to_u32(fmt + 4); + fmtOut->avgBytesPerSec = drwav__bytes_to_u32(fmt + 8); + fmtOut->blockAlign = drwav__bytes_to_u16(fmt + 12); + fmtOut->bitsPerSample = drwav__bytes_to_u16(fmt + 14); + + fmtOut->extendedSize = 0; + fmtOut->validBitsPerSample = 0; + fmtOut->channelMask = 0; + memset(fmtOut->subFormat, 0, sizeof(fmtOut->subFormat)); + + if (header.sizeInBytes > 16) { + unsigned char fmt_cbSize[2]; + if (onRead(pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) { + return DRWAV_FALSE; // Expecting more data. + } + *pRunningBytesReadOut += sizeof(fmt_cbSize); + + int bytesReadSoFar = 18; + + fmtOut->extendedSize = drwav__bytes_to_u16(fmt_cbSize); + if (fmtOut->extendedSize > 0) { + // Simple validation. + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + if (fmtOut->extendedSize != 22) { + return DRWAV_FALSE; + } + } + + if (fmtOut->formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + unsigned char fmtext[22]; + if (onRead(pUserData, fmtext, fmtOut->extendedSize) != fmtOut->extendedSize) { + return DRWAV_FALSE; // Expecting more data. + } + + fmtOut->validBitsPerSample = drwav__bytes_to_u16(fmtext + 0); + fmtOut->channelMask = drwav__bytes_to_u32(fmtext + 2); + drwav__bytes_to_guid(fmtext + 6, fmtOut->subFormat); + } else { + if (!onSeek(pUserData, fmtOut->extendedSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + } + *pRunningBytesReadOut += fmtOut->extendedSize; + + bytesReadSoFar += fmtOut->extendedSize; + } + + // Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. + if (!onSeek(pUserData, (int)(header.sizeInBytes - bytesReadSoFar), drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += (header.sizeInBytes - bytesReadSoFar); + } + + if (header.paddingSize > 0) { + if (!onSeek(pUserData, header.paddingSize, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + *pRunningBytesReadOut += header.paddingSize; + } + + return DRWAV_TRUE; +} + + +#ifndef DR_WAV_NO_STDIO +FILE* drwav_fopen(const char* filePath, const char* openMode) +{ + FILE* pFile; +#if defined(_MSC_VER) && _MSC_VER >= 1400 + if (fopen_s(&pFile, filePath, openMode) != 0) { + return DRWAV_FALSE; + } +#else + pFile = fopen(filePath, openMode); + if (pFile == NULL) { + return DRWAV_FALSE; + } +#endif + + return pFile; +} + +static size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite) +{ + return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData); +} + +static drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin) +{ + return fseek((FILE*)pUserData, offset, (origin == drwav_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0; +} + +drwav_bool32 drwav_init_file(drwav* pWav, const char* filename) +{ + FILE* pFile = drwav_fopen(filename, "rb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + return drwav_init(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, (void*)pFile); +} + + +drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + FILE* pFile = drwav_fopen(filename, "wb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile); +} + +drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE); +} + +drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE); +} + +drwav* drwav_open_file(const char* filename) +{ + FILE* pFile = drwav_fopen(filename, "rb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + drwav* pWav = drwav_open(drwav__on_read_stdio, drwav__on_seek_stdio, (void*)pFile); + if (pWav == NULL) { + fclose(pFile); + return NULL; + } + + return pWav; +} + + +drwav* drwav_open_file_write__internal(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + FILE* pFile = drwav_fopen(filename, "wb"); + if (pFile == NULL) { + return DRWAV_FALSE; + } + + drwav* pWav = drwav_open_write__internal(pFormat, totalSampleCount, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile); + if (pWav == NULL) { + fclose(pFile); + return NULL; + } + + return pWav; +} + +drwav* drwav_open_file_write(const char* filename, const drwav_data_format* pFormat) +{ + return drwav_open_file_write__internal(filename, pFormat, 0, DRWAV_FALSE); +} + +drwav* drwav_open_file_write_sequential(const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_open_file_write__internal(filename, pFormat, totalSampleCount, DRWAV_TRUE); +} +#endif //DR_WAV_NO_STDIO + + +static size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drwav__memory_stream* memory = (drwav__memory_stream*)pUserData; + drwav_assert(memory != NULL); + drwav_assert(memory->dataSize >= memory->currentReadPos); + + size_t bytesRemaining = memory->dataSize - memory->currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRWAV_COPY_MEMORY(pBufferOut, memory->data + memory->currentReadPos, bytesToRead); + memory->currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav__memory_stream* memory = (drwav__memory_stream*)pUserData; + drwav_assert(memory != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (memory->currentReadPos + offset > memory->dataSize) { + offset = (int)(memory->dataSize - memory->currentReadPos); // Trying to seek too far forward. + } + } else { + if (memory->currentReadPos < (size_t)-offset) { + offset = -(int)memory->currentReadPos; // Trying to seek too far backwards. + } + } + + // This will never underflow thanks to the clamps above. + memory->currentReadPos += offset; + } else { + if ((drwav_uint32)offset <= memory->dataSize) { + memory->currentReadPos = offset; + } else { + memory->currentReadPos = memory->dataSize; // Trying to seek too far forward. + } + } + + return DRWAV_TRUE; +} + +static size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite) +{ + drwav__memory_stream_write* memory = (drwav__memory_stream_write*)pUserData; + drwav_assert(memory != NULL); + drwav_assert(memory->dataCapacity >= memory->currentWritePos); + + size_t bytesRemaining = memory->dataCapacity - memory->currentWritePos; + if (bytesRemaining < bytesToWrite) { + // Need to reallocate. + size_t newDataCapacity = (memory->dataCapacity == 0) ? 256 : memory->dataCapacity * 2; + + // If doubling wasn't enough, just make it the minimum required size to write the data. + if ((newDataCapacity - memory->currentWritePos) < bytesToWrite) { + newDataCapacity = memory->currentWritePos + bytesToWrite; + } + + void* pNewData = DRWAV_REALLOC(*memory->ppData, newDataCapacity); + if (pNewData == NULL) { + return 0; + } + + *memory->ppData = pNewData; + memory->dataCapacity = newDataCapacity; + } + + drwav_uint8* pDataOut = (drwav_uint8*)(*memory->ppData); + DRWAV_COPY_MEMORY(pDataOut + memory->currentWritePos, pDataIn, bytesToWrite); + + memory->currentWritePos += bytesToWrite; + if (memory->dataSize < memory->currentWritePos) { + memory->dataSize = memory->currentWritePos; + } + + *memory->pDataSize = memory->dataSize; + + return bytesToWrite; +} + +static drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav__memory_stream_write* memory = (drwav__memory_stream_write*)pUserData; + drwav_assert(memory != NULL); + + if (origin == drwav_seek_origin_current) { + if (offset > 0) { + if (memory->currentWritePos + offset > memory->dataSize) { + offset = (int)(memory->dataSize - memory->currentWritePos); // Trying to seek too far forward. + } + } else { + if (memory->currentWritePos < (size_t)-offset) { + offset = -(int)memory->currentWritePos; // Trying to seek too far backwards. + } + } + + // This will never underflow thanks to the clamps above. + memory->currentWritePos += offset; + } else { + if ((drwav_uint32)offset <= memory->dataSize) { + memory->currentWritePos = offset; + } else { + memory->currentWritePos = memory->dataSize; // Trying to seek too far forward. + } + } + + return DRWAV_TRUE; +} + +drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + drwav__memory_stream memoryStream; + drwav_zero_memory(&memoryStream, sizeof(memoryStream)); + memoryStream.data = (const unsigned char*)data; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + + if (!drwav_init(pWav, drwav__on_read_memory, drwav__on_seek_memory, (void*)&memoryStream)) { + return DRWAV_FALSE; + } + + pWav->memoryStream = memoryStream; + pWav->pUserData = &pWav->memoryStream; + return DRWAV_TRUE; +} + + +drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + if (ppData == NULL) { + return DRWAV_FALSE; + } + + *ppData = NULL; // Important because we're using realloc()! + *pDataSize = 0; + + drwav__memory_stream_write memoryStreamWrite; + drwav_zero_memory(&memoryStreamWrite, sizeof(memoryStreamWrite)); + memoryStreamWrite.ppData = ppData; + memoryStreamWrite.pDataSize = pDataSize; + memoryStreamWrite.dataSize = 0; + memoryStreamWrite.dataCapacity = 0; + memoryStreamWrite.currentWritePos = 0; + + if (!drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, (void*)&memoryStreamWrite)) { + return DRWAV_FALSE; + } + + pWav->memoryStreamWrite = memoryStreamWrite; + pWav->pUserData = &pWav->memoryStreamWrite; + return DRWAV_TRUE; +} + +drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE); +} + +drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE); +} + + +drwav* drwav_open_memory(const void* data, size_t dataSize) +{ + if (data == NULL || dataSize == 0) { + return NULL; + } + + drwav__memory_stream memoryStream; + drwav_zero_memory(&memoryStream, sizeof(memoryStream)); + memoryStream.data = (const unsigned char*)data; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + + drwav* pWav = drwav_open(drwav__on_read_memory, drwav__on_seek_memory, (void*)&memoryStream); + if (pWav == NULL) { + return NULL; + } + + pWav->memoryStream = memoryStream; + pWav->pUserData = &pWav->memoryStream; + return pWav; +} + + +drwav* drwav_open_memory_write__internal(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential) +{ + if (ppData == NULL) { + return NULL; + } + + *ppData = NULL; // Important because we're using realloc()! + *pDataSize = 0; + + drwav__memory_stream_write memoryStreamWrite; + drwav_zero_memory(&memoryStreamWrite, sizeof(memoryStreamWrite)); + memoryStreamWrite.ppData = ppData; + memoryStreamWrite.pDataSize = pDataSize; + memoryStreamWrite.dataSize = 0; + memoryStreamWrite.dataCapacity = 0; + memoryStreamWrite.currentWritePos = 0; + + drwav* pWav = drwav_open_write__internal(pFormat, totalSampleCount, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, (void*)&memoryStreamWrite); + if (pWav == NULL) { + return NULL; + } + + pWav->memoryStreamWrite = memoryStreamWrite; + pWav->pUserData = &pWav->memoryStreamWrite; + return pWav; +} + +drwav* drwav_open_memory_write(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat) +{ + return drwav_open_memory_write__internal(ppData, pDataSize, pFormat, 0, DRWAV_FALSE); +} + +drwav* drwav_open_memory_write_sequential(void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + return drwav_open_memory_write__internal(ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE); +} + + +drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData) +{ + if (onRead == NULL || onSeek == NULL) { + return DRWAV_FALSE; + } + + drwav_zero_memory(pWav, sizeof(*pWav)); + + + // The first 4 bytes should be the RIFF identifier. + unsigned char riff[4]; + if (onRead(pUserData, riff, sizeof(riff)) != sizeof(riff)) { + return DRWAV_FALSE; // Failed to read data. + } + + // The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for + // w64 it will start with "riff". + if (drwav__fourcc_equal(riff, "RIFF")) { + pWav->container = drwav_container_riff; + } else if (drwav__fourcc_equal(riff, "riff")) { + pWav->container = drwav_container_w64; + + // Check the rest of the GUID for validity. + drwav_uint8 riff2[12]; + if (onRead(pUserData, riff2, sizeof(riff2)) != sizeof(riff2)) { + return DRWAV_FALSE; + } + + for (int i = 0; i < 12; ++i) { + if (riff2[i] != drwavGUID_W64_RIFF[i+4]) { + return DRWAV_FALSE; + } + } + } else { + return DRWAV_FALSE; // Unknown or unsupported container. + } + + + if (pWav->container == drwav_container_riff) { + // RIFF/WAVE + unsigned char chunkSizeBytes[4]; + if (onRead(pUserData, chunkSizeBytes, sizeof(chunkSizeBytes)) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + unsigned int chunkSize = drwav__bytes_to_u32(chunkSizeBytes); + if (chunkSize < 36) { + return DRWAV_FALSE; // Chunk size should always be at least 36 bytes. + } + + unsigned char wave[4]; + if (onRead(pUserData, wave, sizeof(wave)) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__fourcc_equal(wave, "WAVE")) { + return DRWAV_FALSE; // Expecting "WAVE". + } + + pWav->dataChunkDataPos = 4 + sizeof(chunkSizeBytes) + sizeof(wave); + } else { + // W64 + unsigned char chunkSize[8]; + if (onRead(pUserData, chunkSize, sizeof(chunkSize)) != sizeof(chunkSize)) { + return DRWAV_FALSE; + } + + if (drwav__bytes_to_u64(chunkSize) < 80) { + return DRWAV_FALSE; + } + + drwav_uint8 wave[16]; + if (onRead(pUserData, wave, sizeof(wave)) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav__guid_equal(wave, drwavGUID_W64_WAVE)) { + return DRWAV_FALSE; + } + + pWav->dataChunkDataPos = 16 + sizeof(chunkSize) + sizeof(wave); + } + + + // The next bytes should be the "fmt " chunk. + drwav_fmt fmt; + if (!drwav__read_fmt(onRead, onSeek, pUserData, pWav->container, &pWav->dataChunkDataPos, &fmt)) { + return DRWAV_FALSE; // Failed to read the "fmt " chunk. + } + + // Basic validation. + if (fmt.sampleRate == 0 || fmt.channels == 0 || fmt.bitsPerSample == 0 || fmt.blockAlign == 0) { + return DRWAV_FALSE; // Invalid channel count. Probably an invalid WAV file. + } + + + // Translate the internal format. + unsigned short translatedFormatTag = fmt.formatTag; + if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + translatedFormatTag = drwav__bytes_to_u16(fmt.subFormat + 0); + } + + + drwav_uint64 sampleCountFromFactChunk = 0; + + // The next chunk we care about is the "data" chunk. This is not necessarily the next chunk so we'll need to loop. + drwav_uint64 dataSize; + for (;;) + { + drwav__chunk_header header; + if (!drwav__read_chunk_header(onRead, pUserData, pWav->container, &pWav->dataChunkDataPos, &header)) { + return DRWAV_FALSE; + } + + dataSize = header.sizeInBytes; + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "data")) { + break; + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_DATA)) { + break; + } + } + + // Optional. Get the total sample count from the FACT chunk. This is useful for compressed formats. + if (pWav->container == drwav_container_riff) { + if (drwav__fourcc_equal(header.id.fourcc, "fact")) { + drwav_uint32 sampleCount; + if (onRead(pUserData, &sampleCount, 4) != 4) { + return DRWAV_FALSE; + } + pWav->dataChunkDataPos += 4; + dataSize -= 4; + + // The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this + // for Microsoft ADPCM formats. + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + sampleCountFromFactChunk = sampleCount; + } else { + sampleCountFromFactChunk = 0; + } + } + } else { + if (drwav__guid_equal(header.id.guid, drwavGUID_W64_FACT)) { + if (onRead(pUserData, &sampleCountFromFactChunk, 8) != 8) { + return DRWAV_FALSE; + } + pWav->dataChunkDataPos += 8; + dataSize -= 8; + } + } + + // If we get here it means we didn't find the "data" chunk. Seek past it. + + // Make sure we seek past the padding. + dataSize += header.paddingSize; + drwav__seek_forward(onSeek, dataSize, pUserData); + pWav->dataChunkDataPos += dataSize; + } + + // At this point we should be sitting on the first byte of the raw audio data. + + pWav->onRead = onRead; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->fmt = fmt; + pWav->sampleRate = fmt.sampleRate; + pWav->channels = fmt.channels; + pWav->bitsPerSample = fmt.bitsPerSample; + pWav->bytesPerSample = fmt.blockAlign / fmt.channels; + pWav->bytesRemaining = dataSize; + pWav->translatedFormatTag = translatedFormatTag; + pWav->dataChunkDataSize = dataSize; + + // The bytes per sample should never be 0 at this point. This would indicate an invalid WAV file. + if (pWav->bytesPerSample == 0) { + return DRWAV_FALSE; + } + + if (sampleCountFromFactChunk != 0) { + pWav->totalSampleCount = sampleCountFromFactChunk * fmt.channels; + } else { + pWav->totalSampleCount = dataSize / pWav->bytesPerSample; + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataSize / fmt.blockAlign; + pWav->totalSampleCount = (blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2; // x2 because two samples per byte. + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataSize / fmt.blockAlign; + pWav->totalSampleCount = ((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels); + } + } + + // The way we calculate the bytes per sample does not make sense for compressed formats so we just set it to 0. + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + pWav->bytesPerSample = 0; + } + + // Some formats only support a certain number of channels. + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + if (pWav->channels > 2) { + return DRWAV_FALSE; + } + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + // I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website), + // it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count + // from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct + // way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should + // always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my + // correctness tests against libsndfile, and is disabled by default. + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataSize / fmt.blockAlign; + pWav->totalSampleCount = (blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2; // x2 because two samples per byte. + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataSize / fmt.blockAlign; + pWav->totalSampleCount = ((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels); + } +#endif + + return DRWAV_TRUE; +} + + +drwav_uint32 drwav_riff_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= (0xFFFFFFFF - 36)) { + return 36 + (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFF; + } +} + +drwav_uint32 drwav_data_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= 0xFFFFFFFF) { + return (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFF; + } +} + +drwav_uint64 drwav_riff_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 80 + 24 + dataChunkSize; // +24 because W64 includes the size of the GUID and size fields. +} + +drwav_uint64 drwav_data_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 24 + dataChunkSize; // +24 because W64 includes the size of the GUID and size fields. +} + + +drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + if (pWav == NULL) { + return DRWAV_FALSE; + } + + if (onWrite == NULL) { + return DRWAV_FALSE; + } + + if (!isSequential && onSeek == NULL) { + return DRWAV_FALSE; // <-- onSeek is required when in non-sequential mode. + } + + + // Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. + if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) { + return DRWAV_FALSE; + } + if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) { + return DRWAV_FALSE; + } + + + drwav_zero_memory(pWav, sizeof(*pWav)); + pWav->onWrite = onWrite; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->fmt.formatTag = (drwav_uint16)pFormat->format; + pWav->fmt.channels = (drwav_uint16)pFormat->channels; + pWav->fmt.sampleRate = pFormat->sampleRate; + pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8); + pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8); + pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->fmt.extendedSize = 0; + pWav->isSequentialWrite = isSequential; + + + size_t runningPos = 0; + + // The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In + // sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non- + // sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek. + drwav_uint64 initialDataChunkSize = 0; + if (isSequential) { + initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8; + + // The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64 + // so for the sake of simplicity I'm not doing any validation for that. + if (pFormat->container == drwav_container_riff) { + if (initialDataChunkSize > (0xFFFFFFFF - 36)) { + return DRWAV_FALSE; // Not enough room to store every sample. + } + } + } + + pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize; + + + // "RIFF" chunk. + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeRIFF = 36 + (drwav_uint32)initialDataChunkSize; // +36 = "RIFF"+[RIFF Chunk Size]+"WAVE" + [sizeof "fmt " chunk] + runningPos += pWav->onWrite(pUserData, "RIFF", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeRIFF, 4); + runningPos += pWav->onWrite(pUserData, "WAVE", 4); + } else { + drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; // +24 because W64 includes the size of the GUID and size fields. + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_RIFF, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeRIFF, 8); + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_WAVE, 16); + } + + // "fmt " chunk. + drwav_uint64 chunkSizeFMT; + if (pFormat->container == drwav_container_riff) { + chunkSizeFMT = 16; + runningPos += pWav->onWrite(pUserData, "fmt ", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeFMT, 4); + } else { + chunkSizeFMT = 40; + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_FMT, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeFMT, 8); + } + + runningPos += pWav->onWrite(pUserData, &pWav->fmt.formatTag, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.channels, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.sampleRate, 4); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.avgBytesPerSec, 4); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.blockAlign, 2); + runningPos += pWav->onWrite(pUserData, &pWav->fmt.bitsPerSample, 2); + + pWav->dataChunkDataPos = runningPos; + + // "data" chunk. + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize; + runningPos += pWav->onWrite(pUserData, "data", 4); + runningPos += pWav->onWrite(pUserData, &chunkSizeDATA, 4); + } else { + drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; // +24 because W64 includes the size of the GUID and size fields. + runningPos += pWav->onWrite(pUserData, drwavGUID_W64_DATA, 16); + runningPos += pWav->onWrite(pUserData, &chunkSizeDATA, 8); + } + + + // Simple validation. + if (pFormat->container == drwav_container_riff) { + if (runningPos != 20 + chunkSizeFMT + 8) { + return DRWAV_FALSE; + } + } else { + if (runningPos != 40 + chunkSizeFMT + 24) { + return DRWAV_FALSE; + } + } + + + + // Set some properties for the client's convenience. + pWav->container = pFormat->container; + pWav->channels = (drwav_uint16)pFormat->channels; + pWav->sampleRate = pFormat->sampleRate; + pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->bytesPerSample = (drwav_uint16)(pFormat->bitsPerSample >> 3); + pWav->translatedFormatTag = (drwav_uint16)pFormat->format; + + return DRWAV_TRUE; +} + + +drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_init_write__internal(pWav, pFormat, 0, DRWAV_FALSE, onWrite, onSeek, pUserData); // DRWAV_FALSE = Not Sequential +} + +drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData) +{ + return drwav_init_write__internal(pWav, pFormat, totalSampleCount, DRWAV_TRUE, onWrite, NULL, pUserData); // DRWAV_TRUE = Sequential +} + +void drwav_uninit(drwav* pWav) +{ + if (pWav == NULL) { + return; + } + + // If the drwav object was opened in write mode we'll need to finalize a few things: + // - Make sure the "data" chunk is aligned to 16-bits for RIFF containers, or 64 bits for W64 containers. + // - Set the size of the "data" chunk. + if (pWav->onWrite != NULL) { + // Validation for sequential mode. + if (pWav->isSequentialWrite) { + drwav_assert(pWav->dataChunkDataSize == pWav->dataChunkDataSizeTargetWrite); + } + + // Padding. Do not adjust pWav->dataChunkDataSize - this should not include the padding. + drwav_uint32 paddingSize = 0; + if (pWav->container == drwav_container_riff) { + paddingSize = (drwav_uint32)(pWav->dataChunkDataSize % 2); + } else { + paddingSize = (drwav_uint32)(pWav->dataChunkDataSize % 8); + } + + if (paddingSize > 0) { + drwav_uint64 paddingData = 0; + pWav->onWrite(pWav->pUserData, &paddingData, paddingSize); + } + + + // Chunk sizes. When using sequential mode, these will have been filled in at initialization time. We only need + // to do this when using non-sequential mode. + if (pWav->onSeek && !pWav->isSequentialWrite) { + if (pWav->container == drwav_container_riff) { + // The "RIFF" chunk size. + if (pWav->onSeek(pWav->pUserData, 4, drwav_seek_origin_start)) { + drwav_uint32 riffChunkSize = drwav_riff_chunk_size_riff(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &riffChunkSize, 4); + } + + // the "data" chunk size. + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 4, drwav_seek_origin_start)) { + drwav_uint32 dataChunkSize = drwav_data_chunk_size_riff(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &dataChunkSize, 4); + } + } else { + // The "RIFF" chunk size. + if (pWav->onSeek(pWav->pUserData, 16, drwav_seek_origin_start)) { + drwav_uint64 riffChunkSize = drwav_riff_chunk_size_w64(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &riffChunkSize, 8); + } + + // The "data" chunk size. + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos + 16, drwav_seek_origin_start)) { + drwav_uint64 dataChunkSize = drwav_data_chunk_size_w64(pWav->dataChunkDataSize); + pWav->onWrite(pWav->pUserData, &dataChunkSize, 8); + } + } + } + } + +#ifndef DR_WAV_NO_STDIO + // If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file() + // was used by looking at the onRead and onSeek callbacks. + if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) { + fclose((FILE*)pWav->pUserData); + } +#endif +} + + +drwav* drwav_open(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData) +{ + drwav* pWav = (drwav*)DRWAV_MALLOC(sizeof(*pWav)); + if (pWav == NULL) { + return NULL; + } + + if (!drwav_init(pWav, onRead, onSeek, pUserData)) { + DRWAV_FREE(pWav); + return NULL; + } + + return pWav; +} + + +drwav* drwav_open_write__internal(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + drwav* pWav = (drwav*)DRWAV_MALLOC(sizeof(*pWav)); + if (pWav == NULL) { + return NULL; + } + + if (!drwav_init_write__internal(pWav, pFormat, totalSampleCount, isSequential, onWrite, onSeek, pUserData)) { + DRWAV_FREE(pWav); + return NULL; + } + + return pWav; +} + +drwav* drwav_open_write(const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData) +{ + return drwav_open_write__internal(pFormat, 0, DRWAV_FALSE, onWrite, onSeek, pUserData); +} + +drwav* drwav_open_write_sequential(const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData) +{ + return drwav_open_write__internal(pFormat, totalSampleCount, DRWAV_TRUE, onWrite, NULL, pUserData); +} + +void drwav_close(drwav* pWav) +{ + drwav_uninit(pWav); + DRWAV_FREE(pWav); +} + + +size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut) +{ + if (pWav == NULL || bytesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + if (bytesToRead > pWav->bytesRemaining) { + bytesToRead = (size_t)pWav->bytesRemaining; + } + + size_t bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead); + + pWav->bytesRemaining -= bytesRead; + return bytesRead; +} + +drwav_uint64 drwav_read(drwav* pWav, drwav_uint64 samplesToRead, void* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + // Cannot use this function for compressed formats. + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + // Don't try to read more samples than can potentially fit in the output buffer. + if (samplesToRead * pWav->bytesPerSample > SIZE_MAX) { + samplesToRead = SIZE_MAX / pWav->bytesPerSample; + } + + size_t bytesRead = drwav_read_raw(pWav, (size_t)(samplesToRead * pWav->bytesPerSample), pBufferOut); + return bytesRead / pWav->bytesPerSample; +} + +drwav_bool32 drwav_seek_to_first_sample(drwav* pWav) +{ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; // No seeking in write mode. + } + + if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, drwav_seek_origin_start)) { + return DRWAV_FALSE; + } + + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + pWav->compressed.iCurrentSample = 0; + } + + pWav->bytesRemaining = pWav->dataChunkDataSize; + return DRWAV_TRUE; +} + +drwav_bool32 drwav_seek_to_sample(drwav* pWav, drwav_uint64 sample) +{ + // Seeking should be compatible with wave files > 2GB. + + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; // No seeking in write mode. + } + + if (pWav == NULL || pWav->onSeek == NULL) { + return DRWAV_FALSE; + } + + // If there are no samples, just return DRWAV_TRUE without doing anything. + if (pWav->totalSampleCount == 0) { + return DRWAV_TRUE; + } + + // Make sure the sample is clamped. + if (sample >= pWav->totalSampleCount) { + sample = pWav->totalSampleCount - 1; + } + + + // For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need + // to seek back to the start. + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + // TODO: This can be optimized. + + // If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards, + // we first need to seek back to the start and then just do the same thing as a forward seek. + if (sample < pWav->compressed.iCurrentSample) { + if (!drwav_seek_to_first_sample(pWav)) { + return DRWAV_FALSE; + } + } + + if (sample > pWav->compressed.iCurrentSample) { + drwav_uint64 offset = sample - pWav->compressed.iCurrentSample; + + drwav_int16 devnull[2048]; + while (offset > 0) { + drwav_uint64 samplesToRead = offset; + if (samplesToRead > 2048) { + samplesToRead = 2048; + } + + drwav_uint64 samplesRead = 0; + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + samplesRead = drwav_read_s16__msadpcm(pWav, samplesToRead, devnull); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + samplesRead = drwav_read_s16__ima(pWav, samplesToRead, devnull); + } else { + assert(DRWAV_FALSE); // If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. + } + + if (samplesRead != samplesToRead) { + return DRWAV_FALSE; + } + + offset -= samplesRead; + } + } + } else { + drwav_uint64 totalSizeInBytes = pWav->totalSampleCount * pWav->bytesPerSample; + drwav_assert(totalSizeInBytes >= pWav->bytesRemaining); + + drwav_uint64 currentBytePos = totalSizeInBytes - pWav->bytesRemaining; + drwav_uint64 targetBytePos = sample * pWav->bytesPerSample; + + drwav_uint64 offset; + if (currentBytePos < targetBytePos) { + // Offset forwards. + offset = (targetBytePos - currentBytePos); + } else { + // Offset backwards. + if (!drwav_seek_to_first_sample(pWav)) { + return DRWAV_FALSE; + } + offset = targetBytePos; + } + + while (offset > 0) { + int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset); + if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) { + return DRWAV_FALSE; + } + + pWav->bytesRemaining -= offset32; + offset -= offset32; + } + } + + return DRWAV_TRUE; +} + + +size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData) +{ + if (pWav == NULL || bytesToWrite == 0 || pData == NULL) { + return 0; + } + + size_t bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite); + pWav->dataChunkDataSize += bytesWritten; + + return bytesWritten; +} + +drwav_uint64 drwav_write(drwav* pWav, drwav_uint64 samplesToWrite, const void* pData) +{ + if (pWav == NULL || samplesToWrite == 0 || pData == NULL) { + return 0; + } + + drwav_uint64 bytesToWrite = ((samplesToWrite * pWav->bitsPerSample) / 8); + if (bytesToWrite > SIZE_MAX) { + return 0; + } + + drwav_uint64 bytesWritten = 0; + const drwav_uint8* pRunningData = (const drwav_uint8*)pData; + while (bytesToWrite > 0) { + drwav_uint64 bytesToWriteThisIteration = bytesToWrite; + if (bytesToWriteThisIteration > SIZE_MAX) { + bytesToWriteThisIteration = SIZE_MAX; + } + + size_t bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample; +} + + + +drwav_uint64 drwav_read_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_assert(pWav != NULL); + drwav_assert(samplesToRead > 0); + drwav_assert(pBufferOut != NULL); + + // TODO: Lots of room for optimization here. + + drwav_uint64 totalSamplesRead = 0; + + while (samplesToRead > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + // If there are no cached samples we need to load a new block. + if (pWav->msadpcm.cachedSampleCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + // Mono. + drwav_uint8 header[7]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 1); + pWav->msadpcm.prevSamples[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 3); + pWav->msadpcm.prevSamples[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 5); + pWav->msadpcm.cachedSamples[2] = pWav->msadpcm.prevSamples[0][0]; + pWav->msadpcm.cachedSamples[3] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.cachedSampleCount = 2; + } else { + // Stereo. + drwav_uint8 header[14]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.predictor[1] = header[1]; + pWav->msadpcm.delta[0] = drwav__bytes_to_s16(header + 2); + pWav->msadpcm.delta[1] = drwav__bytes_to_s16(header + 4); + pWav->msadpcm.prevSamples[0][1] = (drwav_int32)drwav__bytes_to_s16(header + 6); + pWav->msadpcm.prevSamples[1][1] = (drwav_int32)drwav__bytes_to_s16(header + 8); + pWav->msadpcm.prevSamples[0][0] = (drwav_int32)drwav__bytes_to_s16(header + 10); + pWav->msadpcm.prevSamples[1][0] = (drwav_int32)drwav__bytes_to_s16(header + 12); + + pWav->msadpcm.cachedSamples[0] = pWav->msadpcm.prevSamples[0][0]; + pWav->msadpcm.cachedSamples[1] = pWav->msadpcm.prevSamples[1][0]; + pWav->msadpcm.cachedSamples[2] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.cachedSamples[3] = pWav->msadpcm.prevSamples[1][1]; + pWav->msadpcm.cachedSampleCount = 4; + } + } + + // Output anything that's cached. + while (samplesToRead > 0 && pWav->msadpcm.cachedSampleCount > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + pBufferOut[0] = (drwav_int16)pWav->msadpcm.cachedSamples[drwav_countof(pWav->msadpcm.cachedSamples) - pWav->msadpcm.cachedSampleCount]; + pWav->msadpcm.cachedSampleCount -= 1; + + pBufferOut += 1; + samplesToRead -= 1; + totalSamplesRead += 1; + pWav->compressed.iCurrentSample += 1; + } + + if (samplesToRead == 0) { + return totalSamplesRead; + } + + + // If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + // loop iteration which will trigger the loading of a new block. + if (pWav->msadpcm.cachedSampleCount == 0) { + if (pWav->msadpcm.bytesRemainingInBlock == 0) { + continue; + } else { + drwav_uint8 nibbles; + if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) { + return totalSamplesRead; + } + pWav->msadpcm.bytesRemainingInBlock -= 1; + + // TODO: Optimize away these if statements. + drwav_int32 nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; } + drwav_int32 nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; } + + static drwav_int32 adaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + static drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 }; + static drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 }; + + if (pWav->channels == 1) { + // Mono. + drwav_int32 newSample0; + newSample0 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample0; + + + drwav_int32 newSample1; + newSample1 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[0]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample1; + + + pWav->msadpcm.cachedSamples[2] = newSample0; + pWav->msadpcm.cachedSamples[3] = newSample1; + pWav->msadpcm.cachedSampleCount = 2; + } else { + // Stereo. + + // Left. + drwav_int32 newSample0; + newSample0 = ((pWav->msadpcm.prevSamples[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevSamples[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8; + if (pWav->msadpcm.delta[0] < 16) { + pWav->msadpcm.delta[0] = 16; + } + + pWav->msadpcm.prevSamples[0][0] = pWav->msadpcm.prevSamples[0][1]; + pWav->msadpcm.prevSamples[0][1] = newSample0; + + + // Right. + drwav_int32 newSample1; + newSample1 = ((pWav->msadpcm.prevSamples[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevSamples[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[1]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8; + if (pWav->msadpcm.delta[1] < 16) { + pWav->msadpcm.delta[1] = 16; + } + + pWav->msadpcm.prevSamples[1][0] = pWav->msadpcm.prevSamples[1][1]; + pWav->msadpcm.prevSamples[1][1] = newSample1; + + pWav->msadpcm.cachedSamples[2] = newSample0; + pWav->msadpcm.cachedSamples[3] = newSample1; + pWav->msadpcm.cachedSampleCount = 2; + } + } + } + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_assert(pWav != NULL); + drwav_assert(samplesToRead > 0); + drwav_assert(pBufferOut != NULL); + + // TODO: Lots of room for optimization here. + + drwav_uint64 totalSamplesRead = 0; + + while (samplesToRead > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + // If there are no cached samples we need to load a new block. + if (pWav->ima.cachedSampleCount == 0 && pWav->ima.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + // Mono. + drwav_uint8 header[4]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 1] = pWav->ima.predictor[0]; + pWav->ima.cachedSampleCount = 1; + } else { + // Stereo. + drwav_uint8 header[8]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->ima.predictor[0] = drwav__bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = header[2]; + pWav->ima.predictor[1] = drwav__bytes_to_s16(header + 4); + pWav->ima.stepIndex[1] = header[6]; + + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 2] = pWav->ima.predictor[0]; + pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - 1] = pWav->ima.predictor[1]; + pWav->ima.cachedSampleCount = 2; + } + } + + // Output anything that's cached. + while (samplesToRead > 0 && pWav->ima.cachedSampleCount > 0 && pWav->compressed.iCurrentSample < pWav->totalSampleCount) { + pBufferOut[0] = (drwav_int16)pWav->ima.cachedSamples[drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount]; + pWav->ima.cachedSampleCount -= 1; + + pBufferOut += 1; + samplesToRead -= 1; + totalSamplesRead += 1; + pWav->compressed.iCurrentSample += 1; + } + + if (samplesToRead == 0) { + return totalSamplesRead; + } + + // If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + // loop iteration which will trigger the loading of a new block. + if (pWav->ima.cachedSampleCount == 0) { + if (pWav->ima.bytesRemainingInBlock == 0) { + continue; + } else { + static drwav_int32 indexTable[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 + }; + + static drwav_int32 stepTable[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 + }; + + // From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the + // left channel, 4 bytes for the right channel. + pWav->ima.cachedSampleCount = 8 * pWav->channels; + for (drwav_uint32 iChannel = 0; iChannel < pWav->channels; ++iChannel) { + drwav_uint8 nibbles[4]; + if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) { + return totalSamplesRead; + } + pWav->ima.bytesRemainingInBlock -= 4; + + for (drwav_uint32 iByte = 0; iByte < 4; ++iByte) { + drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0); + drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4); + + drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]]; + drwav_int32 predictor = pWav->ima.predictor[iChannel]; + + drwav_int32 diff = step >> 3; + if (nibble0 & 1) diff += step >> 2; + if (nibble0 & 2) diff += step >> 1; + if (nibble0 & 4) diff += step; + if (nibble0 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedSamples[(drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount) + (iByte*2+0)*pWav->channels + iChannel] = predictor; + + + step = stepTable[pWav->ima.stepIndex[iChannel]]; + predictor = pWav->ima.predictor[iChannel]; + + diff = step >> 3; + if (nibble1 & 1) diff += step >> 2; + if (nibble1 & 2) diff += step >> 1; + if (nibble1 & 4) diff += step; + if (nibble1 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedSamples[(drwav_countof(pWav->ima.cachedSamples) - pWav->ima.cachedSampleCount) + (iByte*2+1)*pWav->channels + iChannel] = predictor; + } + } + } + } + } + + return totalSamplesRead; +} + + +#ifndef DR_WAV_NO_CONVERSION_API +static unsigned short g_drwavAlawTable[256] = { + 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580, + 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0, + 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600, + 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00, + 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58, + 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58, + 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960, + 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0, + 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80, + 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40, + 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00, + 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500, + 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8, + 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8, + 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0, + 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350 +}; + +static unsigned short g_drwavMulawTable[256] = { + 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84, + 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84, + 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004, + 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844, + 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64, + 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74, + 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C, + 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000, + 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C, + 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C, + 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC, + 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC, + 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C, + 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C, + 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084, + 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000 +}; + +static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavAlawTable[sampleIn]; +} + +static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavMulawTable[sampleIn]; +} + + + +static void drwav__pcm_to_s16(drwav_int16* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned short bytesPerSample) +{ + // Special case for 8-bit sample data because it's treated as unsigned. + if (bytesPerSample == 1) { + drwav_u8_to_s16(pOut, pIn, totalSampleCount); + return; + } + + + // Slightly more optimal implementation for common formats. + if (bytesPerSample == 2) { + for (unsigned int i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((drwav_int16*)pIn)[i]; + } + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s16(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount); + return; + } + + + // Anything more than 64 bits per sample is not supported. + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + // Generic, slow converter. + for (unsigned int i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int16)((drwav_int64)sample >> 48); + } +} + +static void drwav__ieee_to_s16(drwav_int16* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned short bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s16(pOut, (float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s16(pOut, (double*)pIn, totalSampleCount); + return; + } else { + // Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + +drwav_uint64 drwav_read_s16__pcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + // Fast path. + if (pWav->bytesPerSample == 2) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__ieee(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__alaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16__mulaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s16(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + // Don't try to read more samples than can potentially fit in the output buffer. + if (samplesToRead * sizeof(drwav_int16) > SIZE_MAX) { + samplesToRead = SIZE_MAX / sizeof(drwav_int16); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_s16__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_s16__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_s16__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_s16__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_s16__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_s16__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + for (size_t i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x - 128; + r = r << 8; + pOut[i] = (short)r; + } +} + +void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + for (size_t i = 0; i < sampleCount; ++i) { + int x = ((int)(((unsigned int)(((unsigned char*)pIn)[i*3+0]) << 8) | ((unsigned int)(((unsigned char*)pIn)[i*3+1]) << 16) | ((unsigned int)(((unsigned char*)pIn)[i*3+2])) << 24)) >> 8; + r = x >> 8; + pOut[i] = (short)r; + } +} + +void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + int r; + for (size_t i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x >> 16; + pOut[i] = (short)r; + } +} + +void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount) +{ + int r; + for (size_t i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + float c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5f); + r = r - 32768; + pOut[i] = (short)r; + } +} + +void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount) +{ + int r; + for (size_t i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + double c; + c = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); + c = c + 1; + r = (int)(c * 32767.5); + r = r - 32768; + pOut[i] = (short)r; + } +} + +void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + for (size_t i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__alaw_to_s16(pIn[i]); + } +} + +void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + for (size_t i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__mulaw_to_s16(pIn[i]); + } +} + + + +static void drwav__pcm_to_f32(float* pOut, const unsigned char* pIn, size_t sampleCount, unsigned short bytesPerSample) +{ + // Special case for 8-bit sample data because it's treated as unsigned. + if (bytesPerSample == 1) { + drwav_u8_to_f32(pOut, pIn, sampleCount); + return; + } + + // Slightly more optimal implementation for common formats. + if (bytesPerSample == 2) { + drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_f32(pOut, pIn, sampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount); + return; + } + + + // Anything more than 64 bits per sample is not supported. + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, sampleCount * sizeof(*pOut)); + return; + } + + + // Generic, slow converter. + for (unsigned int i = 0; i < sampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0); + } +} + +static void drwav__ieee_to_f32(float* pOut, const unsigned char* pIn, size_t sampleCount, unsigned short bytesPerSample) +{ + if (bytesPerSample == 4) { + for (unsigned int i = 0; i < sampleCount; ++i) { + *pOut++ = ((float*)pIn)[i]; + } + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_f32(pOut, (double*)pIn, sampleCount); + return; + } else { + // Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. + drwav_zero_memory(pOut, sampleCount * sizeof(*pOut)); + return; + } +} + + +drwav_uint64 drwav_read_f32__pcm(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + pBufferOut += samplesRead; + + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + // We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + // want to duplicate that code. + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)samplesRead); // <-- Safe cast because we're clamping to 2048. + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__ima(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + // We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + // want to duplicate that code. + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)samplesRead); // <-- Safe cast because we're clamping to 2048. + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__ieee(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + // Fast path. + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bytesPerSample == 4) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__alaw(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32__mulaw(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_f32(drwav* pWav, drwav_uint64 samplesToRead, float* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + // Don't try to read more samples than can potentially fit in the output buffer. + if (samplesToRead * sizeof(float) > SIZE_MAX) { + samplesToRead = SIZE_MAX / sizeof(float); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_f32__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_f32__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_f32__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_f32__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_f32__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_f32__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + // It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears + // libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note + // the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated + // correctness testing. This is disabled by default. + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 256.0f) * 2 - 1; + } +#else + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 255.0f) * 2 - 1; + } +#endif +} + +void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] / 32768.0f; + } +} + +void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + int sample32 = (int)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = (float)(sample32 / 2147483648.0); + } +} + +void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (float)(pIn[i] / 2147483648.0); + } +} + +void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (float)pIn[i]; + } +} + +void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f; + } +} + +void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f; + } +} + + + +static void drwav__pcm_to_s32(drwav_int32* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned short bytesPerSample) +{ + // Special case for 8-bit sample data because it's treated as unsigned. + if (bytesPerSample == 1) { + drwav_u8_to_s32(pOut, pIn, totalSampleCount); + return; + } + + // Slightly more optimal implementation for common formats. + if (bytesPerSample == 2) { + drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s32(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + for (unsigned int i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((drwav_int32*)pIn)[i]; + } + return; + } + + + // Anything more than 64 bits per sample is not supported. + if (bytesPerSample > 8) { + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + // Generic, slow converter. + for (unsigned int i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample && j < 8; j += 1) { + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + pIn += j; + *pOut++ = (drwav_int32)((drwav_int64)sample >> 32); + } +} + +static void drwav__ieee_to_s32(drwav_int32* pOut, const unsigned char* pIn, size_t totalSampleCount, unsigned short bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s32(pOut, (float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s32(pOut, (double*)pIn, totalSampleCount); + return; + } else { + // Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. + drwav_zero_memory(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + + +drwav_uint64 drwav_read_s32__pcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + // Fast path. + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bytesPerSample == 4) { + return drwav_read(pWav, samplesToRead, pBufferOut); + } + + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + // We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + // want to duplicate that code. + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)samplesRead); // <-- Safe cast because we're clamping to 2048. + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + // We're just going to borrow the implementation from the drwav_read_s16() since IMA-ADPCM is a little bit more complicated than other formats and I don't + // want to duplicate that code. + drwav_uint64 totalSamplesRead = 0; + drwav_int16 samples16[2048]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read_s16(pWav, drwav_min(samplesToRead, 2048), samples16); + if (samplesRead == 0) { + break; + } + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)samplesRead); // <-- Safe cast because we're clamping to 2048. + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__ieee(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, pWav->bytesPerSample); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__alaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32__mulaw(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + if (pWav->bytesPerSample == 0) { + return 0; + } + + drwav_uint64 totalSamplesRead = 0; + unsigned char sampleData[4096]; + while (samplesToRead > 0) { + drwav_uint64 samplesRead = drwav_read(pWav, drwav_min(samplesToRead, sizeof(sampleData)/pWav->bytesPerSample), sampleData); + if (samplesRead == 0) { + break; + } + + drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + pBufferOut += samplesRead; + samplesToRead -= samplesRead; + totalSamplesRead += samplesRead; + } + + return totalSamplesRead; +} + +drwav_uint64 drwav_read_s32(drwav* pWav, drwav_uint64 samplesToRead, drwav_int32* pBufferOut) +{ + if (pWav == NULL || samplesToRead == 0 || pBufferOut == NULL) { + return 0; + } + + // Don't try to read more samples than can potentially fit in the output buffer. + if (samplesToRead * sizeof(drwav_int32) > SIZE_MAX) { + samplesToRead = SIZE_MAX / sizeof(drwav_int32); + } + + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_s32__pcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_s32__msadpcm(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_s32__ieee(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_s32__alaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_s32__mulaw(pWav, samplesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_s32__ima(pWav, samplesToRead, pBufferOut); + } + + return 0; +} + +void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = ((int)pIn[i] - 128) << 24; + } +} + +void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] << 16; + } +} + +void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = sample32; + } +} + +void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)(2147483648.0 * pIn[i]); + } +} + +void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i = 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16; + } +} + +void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + if (pOut == NULL || pIn == NULL) { + return; + } + + for (size_t i= 0; i < sampleCount; ++i) { + *pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16; + } +} + + + +drwav_int16* drwav__read_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_assert(pWav != NULL); + + drwav_uint64 sampleDataSize = pWav->totalSampleCount * sizeof(drwav_int16); + if (sampleDataSize > SIZE_MAX) { + drwav_uninit(pWav); + return NULL; // File's too big. + } + + drwav_int16* pSampleData = (drwav_int16*)DRWAV_MALLOC((size_t)sampleDataSize); // <-- Safe cast due to the check above. + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; // Failed to allocate memory. + } + + drwav_uint64 samplesRead = drwav_read_s16(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; // There was an error reading the samples. + } + + drwav_uninit(pWav); + + if (sampleRate) *sampleRate = pWav->sampleRate; + if (channels) *channels = pWav->channels; + if (totalSampleCount) *totalSampleCount = pWav->totalSampleCount; + return pSampleData; +} + +float* drwav__read_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_assert(pWav != NULL); + + drwav_uint64 sampleDataSize = pWav->totalSampleCount * sizeof(float); + if (sampleDataSize > SIZE_MAX) { + drwav_uninit(pWav); + return NULL; // File's too big. + } + + float* pSampleData = (float*)DRWAV_MALLOC((size_t)sampleDataSize); // <-- Safe cast due to the check above. + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; // Failed to allocate memory. + } + + drwav_uint64 samplesRead = drwav_read_f32(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; // There was an error reading the samples. + } + + drwav_uninit(pWav); + + if (sampleRate) *sampleRate = pWav->sampleRate; + if (channels) *channels = pWav->channels; + if (totalSampleCount) *totalSampleCount = pWav->totalSampleCount; + return pSampleData; +} + +drwav_int32* drwav__read_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + drwav_assert(pWav != NULL); + + drwav_uint64 sampleDataSize = pWav->totalSampleCount * sizeof(drwav_int32); + if (sampleDataSize > SIZE_MAX) { + drwav_uninit(pWav); + return NULL; // File's too big. + } + + drwav_int32* pSampleData = (drwav_int32*)DRWAV_MALLOC((size_t)sampleDataSize); // <-- Safe cast due to the check above. + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; // Failed to allocate memory. + } + + drwav_uint64 samplesRead = drwav_read_s32(pWav, (size_t)pWav->totalSampleCount, pSampleData); + if (samplesRead != pWav->totalSampleCount) { + DRWAV_FREE(pSampleData); + drwav_uninit(pWav); + return NULL; // There was an error reading the samples. + } + + drwav_uninit(pWav); + + if (sampleRate) *sampleRate = pWav->sampleRate; + if (channels) *channels = pWav->channels; + if (totalSampleCount) *totalSampleCount = pWav->totalSampleCount; + return pSampleData; +} + + +drwav_int16* drwav_open_and_read_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_and_read_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_and_read_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init(&wav, onRead, onSeek, pUserData)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} + +#ifndef DR_WAV_NO_STDIO +drwav_int16* drwav_open_and_read_file_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_and_read_file_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_and_read_file_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_file(&wav, filename)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} +#endif + +drwav_int16* drwav_open_and_read_memory_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_s16(&wav, channels, sampleRate, totalSampleCount); +} + +float* drwav_open_and_read_memory_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_f32(&wav, channels, sampleRate, totalSampleCount); +} + +drwav_int32* drwav_open_and_read_memory_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalSampleCount) +{ + if (sampleRate) *sampleRate = 0; + if (channels) *channels = 0; + if (totalSampleCount) *totalSampleCount = 0; + + drwav wav; + if (!drwav_init_memory(&wav, data, dataSize)) { + return NULL; + } + + return drwav__read_and_close_s32(&wav, channels, sampleRate, totalSampleCount); +} +#endif //DR_WAV_NO_CONVERSION_API + + +void drwav_free(void* pDataReturnedByOpenAndRead) +{ + DRWAV_FREE(pDataReturnedByOpenAndRead); +} + +#endif //DR_WAV_IMPLEMENTATION + + +// REVISION HISTORY +// +// v0.8.1 - 2018-06-29 +// - Add support for sequential writing APIs. +// - Disable seeking in write mode. +// - Fix bugs with Wave64. +// - Fix typos. +// +// v0.8 - 2018-04-27 +// - Bug fix. +// - Start using major.minor.revision versioning. +// +// v0.7f - 2018-02-05 +// - Restrict ADPCM formats to a maximum of 2 channels. +// +// v0.7e - 2018-02-02 +// - Fix a crash. +// +// v0.7d - 2018-02-01 +// - Fix a crash. +// +// v0.7c - 2018-02-01 +// - Set drwav.bytesPerSample to 0 for all compressed formats. +// - Fix a crash when reading 16-bit floating point WAV files. In this case dr_wav will output silence for +// all format conversion reading APIs (*_s16, *_s32, *_f32 APIs). +// - Fix some divide-by-zero errors. +// +// v0.7b - 2018-01-22 +// - Fix errors with seeking of compressed formats. +// - Fix compilation error when DR_WAV_NO_CONVERSION_API +// +// v0.7a - 2017-11-17 +// - Fix some GCC warnings. +// +// v0.7 - 2017-11-04 +// - Add writing APIs. +// +// v0.6 - 2017-08-16 +// - API CHANGE: Rename dr_* types to drwav_*. +// - Add support for custom implementations of malloc(), realloc(), etc. +// - Add support for Microsoft ADPCM. +// - Add support for IMA ADPCM (DVI, format code 0x11). +// - Optimizations to drwav_read_s16(). +// - Bug fixes. +// +// v0.5g - 2017-07-16 +// - Change underlying type for booleans to unsigned. +// +// v0.5f - 2017-04-04 +// - Fix a minor bug with drwav_open_and_read_s16() and family. +// +// v0.5e - 2016-12-29 +// - Added support for reading samples as signed 16-bit integers. Use the _s16() family of APIs for this. +// - Minor fixes to documentation. +// +// v0.5d - 2016-12-28 +// - Use drwav_int*/drwav_uint* sized types to improve compiler support. +// +// v0.5c - 2016-11-11 +// - Properly handle JUNK chunks that come before the FMT chunk. +// +// v0.5b - 2016-10-23 +// - A minor change to drwav_bool8 and drwav_bool32 types. +// +// v0.5a - 2016-10-11 +// - Fixed a bug with drwav_open_and_read() and family due to incorrect argument ordering. +// - Improve A-law and mu-law efficiency. +// +// v0.5 - 2016-09-29 +// - API CHANGE. Swap the order of "channels" and "sampleRate" parameters in drwav_open_and_read*(). Rationale for this is to +// keep it consistent with dr_audio and dr_flac. +// +// v0.4b - 2016-09-18 +// - Fixed a typo in documentation. +// +// v0.4a - 2016-09-18 +// - Fixed a typo. +// - Change date format to ISO 8601 (YYYY-MM-DD) +// +// v0.4 - 2016-07-13 +// - API CHANGE. Make onSeek consistent with dr_flac. +// - API CHANGE. Rename drwav_seek() to drwav_seek_to_sample() for clarity and consistency with dr_flac. +// - Added support for Sony Wave64. +// +// v0.3a - 2016-05-28 +// - API CHANGE. Return drwav_bool32 instead of int in onSeek callback. +// - Fixed a memory leak. +// +// v0.3 - 2016-05-22 +// - Lots of API changes for consistency. +// +// v0.2a - 2016-05-16 +// - Fixed Linux/GCC build. +// +// v0.2 - 2016-05-11 +// - Added support for reading data as signed 32-bit PCM for consistency with dr_flac. +// +// v0.1a - 2016-05-07 +// - Fixed a bug in drwav_open_file() where the file handle would not be closed if the loader failed to initialize. +// +// v0.1 - 2016-05-04 +// - Initial versioned release. + + +/* +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or +distribute this software, either in source code form or as a compiled +binary, for any purpose, commercial or non-commercial, and by any +means. + +In jurisdictions that recognize copyright laws, the author or authors +of this software dedicate any and all copyright interest in the +software to the public domain. We make this dedication for the benefit +of the public at large and to the detriment of our heirs and +successors. We intend this dedication to be an overt act of +relinquishment in perpetuity of all present and future rights to this +software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, +EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF +MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. +IN NO EVENT SHALL THE AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR +OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, +ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR +OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to +*/ diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/allpass.cpp b/plugins/community/repos/Bidoo/src/dep/freeverb/allpass.cpp new file mode 100644 index 00000000..7d9e9644 --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/allpass.cpp @@ -0,0 +1,54 @@ +// Allpass filter implementation +// +// Written by Jezar at Dreampoint, June 2000 +// http://www.dreampoint.co.uk +// This code is public domain + +#include "allpass.hpp" + +namespace rack_plugin_Bidoo { + +allpass::allpass() +{ + bufidx = 0; + buffer = 0; +}; + +allpass::~allpass() +{ + //if (buffer) delete buffer; +}; + +void allpass::setbuffer(float *buf, int size) +{ + buffer = buf; + bufsize = size; +} + +void allpass::changebuffer(float *buf, int size) + { + if (buffer) {delete buffer;} + buffer = new float[size]; + bufsize = size; + bufidx = 0; +} + +void allpass::mute() +{ + for (int i=0; i=bufsize) bufidx = 0; + + return output; +} + +} // namespace rack_plugin_Bidoo + +#endif diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/comb.cpp b/plugins/community/repos/Bidoo/src/dep/freeverb/comb.cpp new file mode 100644 index 00000000..f1d992ab --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/comb.cpp @@ -0,0 +1,69 @@ +// Comb filter implementation +// +// Written by Jezar at Dreampoint, June 2000 +// http://www.dreampoint.co.uk +// This code is public domain + +#include "comb.hpp" + +namespace rack_plugin_Bidoo { + +comb::comb() +{ + filterstore = 0; + bufidx = 0; + buffer = 0; +} + +comb::~comb() +{ + filterstore = 0; + bufidx = 0; + //if (buffer) delete buffer; +} + + +void comb::setbuffer(float *buf, int size) +{ + buffer = buf; + bufsize = size; +} + +void comb::changebuffer(float *buf, int size) + { + if (buffer) {delete buffer;} + buffer = new float[size]; + bufsize = size; + bufidx = 0; +} + +void comb::mute() +{ + for (int i=0; i=bufsize) bufidx = 0; + + return output; +} + +} // namespace rack_plugin_Bidoo + +#endif //_comb_ + +//ends diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/readme.txt b/plugins/community/repos/Bidoo/src/dep/freeverb/readme.txt new file mode 100644 index 00000000..57caed58 --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/readme.txt @@ -0,0 +1,67 @@ +Freeverb - Free, studio-quality reverb SOURCE CODE in the public domain +----------------------------------------------------------------------- + +Written by Jezar at Dreampoint - http://www.dreampoint.co.uk + + +Introduction +------------ + +Hello. + +I'll try to keep this "readme" reasonably small. There are few things in the world that I hate more than long "readme" files. Except "coding conventions" - but more on that later... + +In this zip file you will find two folders of C++ source code: + +"Components" - Contains files that should clean-compile ON ANY TYPE OF COMPUTER OR SYSTEM WHATSOEVER. It should not be necessary to make ANY changes to these files to get them to compile, except to make up for inadequacies of certain compilers. These files create three classes - a comb filter, an allpass filter, and a reverb model made up of a number of instances of the filters, with some features to control the filters at a macro level. You will need to link these classes into another program that interfaces with them. The files in the components drawer are completely independant, and can be built without dependancies on anything else. Because of the simple interface, it should be possible to interface these files to any system - VST, DirectX, anything - without changing them AT ALL. + +"FreeverbVST" - Contains a Steinberg VST implementation of this version of Freeverb, using the components in (surprise) the components folder. It was built on a PC but may compile properly for the Macintosh with no problems. I don't know - I don't have a Macintosh. If you've figured out how to compile the examples in the Steinberg VST Development Kit, then you should easilly figure out how to bring the files into a project and get it working in a few minutes. It should be very simple. + +Note that this version of Freeverb doesn't contain predelay, or any EQ. I thought that might make it difficult to understand the "reverb" part of the code. Once you figure out how Freeverb works, you should find it trivial to add such features with little CPU overhead. + +Also, the code in this version of Freeverb has been optimised. This has changed the sound *slightly*, but not significantly compared to how much processing power it saves. + +Finally, note that there is also a built copy of this version of Freeverb called "Freeverb3.dll" - this is a VST plugin for the PC. If you want a version for the Mac or anything else, then you'll need to build it yourself from the code. + + +Technical Explanation +--------------------- + +Freeverb is a simple implementation of the standard Schroeder/Moorer reverb model. I guess the only reason why it sounds better than other reverbs, is simply because I spent a long while doing listening tests in order to create the values found in "tuning.hh". It uses 8 comb filters on both the left and right channels), and you might possibly be able to get away with less if CPU power is a serious constraint for you. It then feeds the result of the reverb through 4 allpass filters on both the left and right channels. These "smooth" the sound. Adding more than four allpasses doesn't seem to add anything significant to the sound, and if you use less, the sound gets a bit "grainy". The filters on the right channel are slightly detuned compared to the left channel in order to create a stereo effect. + +Hopefully, you should find the code in the components drawer a model of brevity and clarity. Notice that I don't use any "coding conventions". Personally, I think that coding conventions suck. They are meant to make the code "clearer", but they inevitably do the complete opposite, making the code completely unfathomable. Anyone whose done Windows programming with its - frankly stupid - "Hungarian notation" will know exactly what I mean. Coding conventions typically promote issues that are irrelevant up to the status of appearing supremely important. It may have helped back people in the days when compilers where somewhat feeble in their type-safety, but not in the new millenium with advanced C++ compilers. + +Imagine if we rewrote the English language to conform to coding conventions. After all, The arguments should be just as valid for the English language as they are for a computer language. For example, we could put a lower-case "n" in front of every noun, a lower-case "p" in front of a persons name, a lower-case "v" in front of every verb, and a lower-case "a" in front of every adjective. Can you imagine what the English language would look like? All in the name of "clarity". It's just as stupid to do this for computer code as it would be to do it for the English language. I hope that the code for Freeverb in the components drawer demonstrates this, and helps start a movement back towards sanity in coding practices. + + +Background +---------- + +Why is the Freeverb code now public domain? Simple. I only intended to create Freeverb to provide me and my friends with studio-quality reverb for free. I never intended to make any money out of it. However, I simply do not have the time to develop it any further. I'm working on a "concept album" at the moment, and I'll never finish it if I spend any more time programming. + +In any case, I make more far money as a contract programmer - making Mobile Internet products - than I ever could writing plugins, so it simply doesn't make financial sense for me to spend any more time on it. + +Rather than give Freeverb to any particular individual or organisation to profit from it, I've decided to give it away to the internet community at large, so that quality, FREE (or at the very least, low-cost) reverbs can be developed for all platforms. + +Feel free to use the source code for Freeverb in any of your own products, whether they are also available for free, or even if they are commercial - I really don't mind. You may do with the code whatever you wish. If you use it in a product (whether commercial or not), it would be very nice of you, if you were to send me a copy of your product - although I appreciate that this isn't always possible in all circumstances. + +HOWEVER, please don't bug me with questions about how to use this code. I gave away Freeverb because I don't have time to maintain it. That means I *certainly* don't have time to answer questions about the source code, so please don't email questions to me. I *will* ignore them. If you can't figure the code for Freeverb out - then find somebody who can. I hope that either way, you enjoy experimenting with it. + + +Disclaimer +---------- + +This software and source code is given away for free, without any warranties of any kind. It has been given away to the internet community as a free gift, so please treat it in the same spirit. + + +I hope this code is useful and interesting to you all! +I hope you have lots of fun experimenting with it and make good products! + +Very best regards, +Jezar. +Technology Consultant +Dreampoint Design and Engineering +http://www.dreampoint.co.uk + + +//ends diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.cpp b/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.cpp new file mode 100644 index 00000000..f50f96de --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.cpp @@ -0,0 +1,349 @@ +// Reverb model implementation +// +// Written by Jezar at Dreampoint, June 2000 +// http://www.dreampoint.co.uk +// This code is public domain + +#include "revmodel.hpp" +#include + +namespace rack_plugin_Bidoo { + +revmodel::revmodel() +{ + // Tie the components to their buffers + combL[0].setbuffer(bufcombL1,combtuningL1); + combR[0].setbuffer(bufcombR1,combtuningR1); + combL[1].setbuffer(bufcombL2,combtuningL2); + combR[1].setbuffer(bufcombR2,combtuningR2); + combL[2].setbuffer(bufcombL3,combtuningL3); + combR[2].setbuffer(bufcombR3,combtuningR3); + combL[3].setbuffer(bufcombL4,combtuningL4); + combR[3].setbuffer(bufcombR4,combtuningR4); + combL[4].setbuffer(bufcombL5,combtuningL5); + combR[4].setbuffer(bufcombR5,combtuningR5); + combL[5].setbuffer(bufcombL6,combtuningL6); + combR[5].setbuffer(bufcombR6,combtuningR6); + combL[6].setbuffer(bufcombL7,combtuningL7); + combR[6].setbuffer(bufcombR7,combtuningR7); + combL[7].setbuffer(bufcombL8,combtuningL8); + combR[7].setbuffer(bufcombR8,combtuningR8); + allpassL[0].setbuffer(bufallpassL1,allpasstuningL1); + allpassR[0].setbuffer(bufallpassR1,allpasstuningR1); + allpassL[1].setbuffer(bufallpassL2,allpasstuningL2); + allpassR[1].setbuffer(bufallpassR2,allpasstuningR2); + allpassL[2].setbuffer(bufallpassL3,allpasstuningL3); + allpassR[2].setbuffer(bufallpassR3,allpasstuningR3); + allpassL[3].setbuffer(bufallpassL4,allpasstuningL4); + allpassR[3].setbuffer(bufallpassR4,allpasstuningR4); + + // Set default values + allpassL[0].setfeedback(0.5f); + allpassR[0].setfeedback(0.5f); + allpassL[1].setfeedback(0.5f); + allpassR[1].setfeedback(0.5f); + allpassL[2].setfeedback(0.5f); + allpassR[2].setfeedback(0.5f); + allpassL[3].setfeedback(0.5f); + allpassR[3].setfeedback(0.5f); + setwet(initialwet); + setroomsize(initialroom); + setdry(initialdry); + setdamp(initialdamp); + setwidth(initialwidth); + setmode(initialmode); + + // Buffer will be full of rubbish - so we MUST mute them + mute(); +} + +void revmodel::mute() +{ + if (getmode() >= freezemode) + return; + + for (int i=0;i 0) + { + outL = outR = 0; + input = (*inputL + *inputR) * gain; + + // Accumulate comb filters in parallel + for(int i=0; i 0) + { + outL = outR = 0; + input = (*inputL + *inputR) * gain; + + // Accumulate comb filters in parallel + for(int i=0; i= freezemode) + { + roomsize1 = 1; + damp1 = 0; + gain = muted; + } + else + { + roomsize1 = roomsize; + damp1 = damp; + gain = fixedgain; + } + + for(i=0; i= freezemode) + return 1; + else + return 0; +} + +void revmodel::setsamplerate(const float samplerate) { + + sampleRate = samplerate; + float coeff = sampleRate/44100.0; + + int mctL1 = round(coeff * combtuningL1); + int mctR1 = round(coeff * combtuningR1); + int mctL2 = round(coeff * combtuningL2); + int mctR2 = round(coeff * combtuningR2); + int mctL3 = round(coeff * combtuningL3); + int mctR3 = round(coeff * combtuningR3); + int mctL4 = round(coeff * combtuningL4); + int mctR4 = round(coeff * combtuningR4); + int mctL5 = round(coeff * combtuningL5); + int mctR5 = round(coeff * combtuningR5); + int mctL6 = round(coeff * combtuningL6); + int mctR6 = round(coeff * combtuningR6); + int mctL7 = round(coeff * combtuningL7); + int mctR7 = round(coeff * combtuningR7); + int mctL8 = round(coeff * combtuningL8); + int mctR8 = round(coeff * combtuningR8); + + int maptL1 = round(coeff * allpasstuningL1); + int maptR1 = round(coeff * allpasstuningR1); + int maptL2 = round(coeff * allpasstuningL2); + int maptR2 = round(coeff * allpasstuningR2); + int maptL3 = round(coeff * allpasstuningL3); + int maptR3 = round(coeff * allpasstuningR3); + int maptL4 = round(coeff * allpasstuningL4); + int maptR4 = round(coeff * allpasstuningR4); + + mute(); + + combL[0].changebuffer(bufcombL1,mctL1); + combR[0].changebuffer(bufcombR1,mctR1); + combL[1].changebuffer(bufcombL2,mctL2); + combR[1].changebuffer(bufcombR2,mctR2); + combL[2].changebuffer(bufcombL3,mctL3); + combR[2].changebuffer(bufcombR3,mctR3); + combL[3].changebuffer(bufcombL4,mctL4); + combR[3].changebuffer(bufcombR4,mctR4); + combL[4].changebuffer(bufcombL5,mctL5); + combR[4].changebuffer(bufcombR5,mctR5); + combL[5].changebuffer(bufcombL6,mctL6); + combR[5].changebuffer(bufcombR6,mctR6); + combL[6].changebuffer(bufcombL7,mctL7); + combR[6].changebuffer(bufcombR7,mctR7); + combL[7].changebuffer(bufcombL8,mctL8); + combR[7].changebuffer(bufcombR8,mctR8); + + allpassL[0].changebuffer(bufallpassL1,maptL1); + allpassR[0].changebuffer(bufallpassR1,maptR1); + allpassL[1].changebuffer(bufallpassL2,maptL2); + allpassR[1].changebuffer(bufallpassR2,maptR2); + allpassL[2].changebuffer(bufallpassL3,maptL3); + allpassR[2].changebuffer(bufallpassR3,maptR3); + allpassL[3].changebuffer(bufallpassL4,maptL4); + allpassR[3].changebuffer(bufallpassR4,maptR4); + + setwet(initialwet); + setroomsize(initialroom); + setdry(initialdry); + setdamp(initialdamp); + setwidth(initialwidth); + setmode(initialmode); + +} + +} // namespace rack_plugin_Bidoo + +//ends diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.hpp b/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.hpp new file mode 100644 index 00000000..43df0676 --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/revmodel.hpp @@ -0,0 +1,94 @@ +// Reverb model declaration +// +// Written by Jezar at Dreampoint, June 2000 +// http://www.dreampoint.co.uk +// This code is public domain + +#ifndef _revmodel_ +#define _revmodel_ + +#include "comb.hpp" +#include "allpass.hpp" +#include "tuning.hh" + +namespace rack_plugin_Bidoo { + +class revmodel +{ +public: + revmodel(); + void mute(); + void processmix(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip); + void processreplace(float *inputL, float *inputR, float *outputL, float *outputR, long numsamples, int skip); + void process(const float inL, const float inR, const float fbInL, const float fbInR, float &outputL, float &outputR, float &wOutputL, float &wOutputR); + void setroomsize(float value); + float getroomsize(); + void setdamp(float value); + float getdamp(); + void setwet(float value); + float getwet(); + void setdry(float value); + float getdry(); + void setwidth(float value); + float getwidth(); + void setmode(float value); + float getmode(); + void setsamplerate(const float samplerate); +private: + void update(); +private: + float gain; + float roomsize,roomsize1; + float damp,damp1; + float wet,wet1,wet2; + float dry; + float width; + float mode; + float sampleRate; + + // The following are all declared inline + // to remove the need for dynamic allocation + // with its subsequent error-checking messiness + + // Comb filters + comb combL[numcombs]; + comb combR[numcombs]; + + // Allpass filters + allpass allpassL[numallpasses]; + allpass allpassR[numallpasses]; + + // Buffers for the combs + float bufcombL1[combtuningL1]; + float bufcombR1[combtuningR1]; + float bufcombL2[combtuningL2]; + float bufcombR2[combtuningR2]; + float bufcombL3[combtuningL3]; + float bufcombR3[combtuningR3]; + float bufcombL4[combtuningL4]; + float bufcombR4[combtuningR4]; + float bufcombL5[combtuningL5]; + float bufcombR5[combtuningR5]; + float bufcombL6[combtuningL6]; + float bufcombR6[combtuningR6]; + float bufcombL7[combtuningL7]; + float bufcombR7[combtuningR7]; + float bufcombL8[combtuningL8]; + float bufcombR8[combtuningR8]; + + // Buffers for the allpasses + float bufallpassL1[allpasstuningL1]; + float bufallpassR1[allpasstuningR1]; + float bufallpassL2[allpasstuningL2]; + float bufallpassR2[allpasstuningR2]; + float bufallpassL3[allpasstuningL3]; + float bufallpassR3[allpasstuningR3]; + float bufallpassL4[allpasstuningL4]; + float bufallpassR4[allpasstuningR4]; +}; + +} // namespace rack_plugin_Bidoo + +#endif//_revmodel_ + +//ends diff --git a/plugins/community/repos/Bidoo/src/dep/freeverb/tuning.hh b/plugins/community/repos/Bidoo/src/dep/freeverb/tuning.hh new file mode 100644 index 00000000..922fe0ff --- /dev/null +++ b/plugins/community/repos/Bidoo/src/dep/freeverb/tuning.hh @@ -0,0 +1,64 @@ +// Reverb model tuning values +// +// Written by Jezar at Dreampoint, June 2000 +// http://www.dreampoint.co.uk +// This code is public domain + +#ifndef _tuning_ +#define _tuning_ + +namespace rack_plugin_Bidoo { + +const int numcombs = 8; +const int numallpasses = 4; +const float muted = 0; +const float fixedgain = 0.015f; +const float scalewet = 3; +const float scaledry = 2; +const float scaledamp = 0.4f; +const float scaleroom = 0.28f; +const float offsetroom = 0.7f; +const float initialroom = 0.5f; +const float initialdamp = 0.5f; +const float initialwet = 1/scalewet; +const float initialdry = 0; +const float initialwidth = 1; +const float initialmode = 0; +const float freezemode = 0.5f; +const int stereospread = 23; + +// These values assume 44.1KHz sample rate +// they will probably be OK for 48KHz sample rate +// but would need scaling for 96KHz (or other) sample rates. +// The values were obtained by listening tests. +const int combtuningL1 = 1116; +const int combtuningR1 = 1116+stereospread; +const int combtuningL2 = 1188; +const int combtuningR2 = 1188+stereospread; +const int combtuningL3 = 1277; +const int combtuningR3 = 1277+stereospread; +const int combtuningL4 = 1356; +const int combtuningR4 = 1356+stereospread; +const int combtuningL5 = 1422; +const int combtuningR5 = 1422+stereospread; +const int combtuningL6 = 1491; +const int combtuningR6 = 1491+stereospread; +const int combtuningL7 = 1557; +const int combtuningR7 = 1557+stereospread; +const int combtuningL8 = 1617; +const int combtuningR8 = 1617+stereospread; +const int allpasstuningL1 = 556; +const int allpasstuningR1 = 556+stereospread; +const int allpasstuningL2 = 441; +const int allpasstuningR2 = 441+stereospread; +const int allpasstuningL3 = 341; +const int allpasstuningR3 = 341+stereospread; +const int allpasstuningL4 = 225; +const int allpasstuningR4 = 225+stereospread; + +} // namespace rack_plugin_Bidoo + +#endif//_tuning_ + +//ends + diff --git a/vst2_bin/plugins/Bidoo/README.md b/vst2_bin/plugins/Bidoo/README.md index deac72b8..15244565 100644 --- a/vst2_bin/plugins/Bidoo/README.md +++ b/vst2_bin/plugins/Bidoo/README.md @@ -1,7 +1,7 @@ # Bidoo's plugins for [VCVRack](https://vcvrack.com) -![Version](https://img.shields.io/badge/version-0.6.6-green.svg?style=flat-square) +![Version](https://img.shields.io/badge/version-0.6.10-green.svg?style=flat-square) ![License](https://img.shields.io/badge/license-BSD3-blue.svg?style=flat-square) ![Language](https://img.shields.io/badge/language-C++-yellow.svg?style=flat-square) @@ -13,6 +13,20 @@ You can find information on that plugins pack in the [wiki](https://github.com/s ## Last changes +21/08/2018 => 0.6.10 + +rabBIT redesign + +20/08/2018 => 0.6.9 + +rabBIT is a 8 bit reducer/reverser + +09/07/2018 + +Changed the way wav files are loaded and saved => OUAIve and cANARd. Changed the way onsets are detected in cANARd. Fix play mode saving on close for OUAIve. + +This version is compliant with the last version I have of Rack SDK so maybe my pack will be available thru Rack again in 0.6.2. + 13/05/2018 => 0.6.6 antN goes away from mpg123 and is based now on minimp3 so maybe my pack will be available thru Rack again. diff --git a/vst2_bin/plugins/Bidoo/res/HORUS.svg b/vst2_bin/plugins/Bidoo/res/HORUS.svg deleted file mode 100644 index bcbd2a91..00000000 --- a/vst2_bin/plugins/Bidoo/res/HORUS.svg +++ /dev/null @@ -1,186 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/vst2_bin/plugins/Bidoo/res/HORUStemp.svg b/vst2_bin/plugins/Bidoo/res/HORUStemp.svg deleted file mode 100644 index c619930f..00000000 --- a/vst2_bin/plugins/Bidoo/res/HORUStemp.svg +++ /dev/null @@ -1,208 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - bId°° - horUS - - - OUT - - - - IN - - diff --git a/vst2_bin/plugins/Bidoo/res/RABBIT.svg b/vst2_bin/plugins/Bidoo/res/RABBIT.svg new file mode 100644 index 00000000..095341af --- /dev/null +++ b/vst2_bin/plugins/Bidoo/res/RABBIT.svg @@ -0,0 +1,224 @@ + + + + + + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/vst2_bin/plugins/Bidoo/res/RABBITtemp.svg b/vst2_bin/plugins/Bidoo/res/RABBITtemp.svg new file mode 100644 index 00000000..407c4cb1 --- /dev/null +++ b/vst2_bin/plugins/Bidoo/res/RABBITtemp.svg @@ -0,0 +1,230 @@ + + + + + + + + + + + + + + + + + + + + + image/svg+xml + + + + + + + + + + bId°° + rabBIT + + + OUT + + + + IN + + + OFF + REV + diff --git a/vst2_bin/plugins/Bidoo/res/RADAR.svg b/vst2_bin/plugins/Bidoo/res/RADAR.svg deleted file mode 100644 index 26dfa5df..00000000 --- a/vst2_bin/plugins/Bidoo/res/RADAR.svg +++ /dev/null @@ -1,602 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - TRIG - - - POS - - - OUT - GATE - L - R - diff --git a/vst2_bin/plugins/Bidoo/res/RADARtemp.svg b/vst2_bin/plugins/Bidoo/res/RADARtemp.svg deleted file mode 100644 index 0789c119..00000000 --- a/vst2_bin/plugins/Bidoo/res/RADARtemp.svg +++ /dev/null @@ -1,728 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - RAdAR - - - - TRIG - - - POS - - - OUT - GATE - L - R - diff --git a/vst2_bin/plugins/Bidoo/res/SONAR.svg b/vst2_bin/plugins/Bidoo/res/SONAR.svg deleted file mode 100644 index d61b0056..00000000 --- a/vst2_bin/plugins/Bidoo/res/SONAR.svg +++ /dev/null @@ -1,134 +0,0 @@ - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/vst2_bin/plugins/Bidoo/res/SONARtemp.svg b/vst2_bin/plugins/Bidoo/res/SONARtemp.svg deleted file mode 100644 index 108cd2ae..00000000 --- a/vst2_bin/plugins/Bidoo/res/SONARtemp.svg +++ /dev/null @@ -1,153 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - image/svg+xml - - - - - - - - - bId°° - SONar - -