Mercurial > hg > beaglert
diff examples/08-PureData/customRender/heavy/render.cpp @ 552:f8bb6186498d prerelease
added customRender example for predate
author | chnrx <chris.heinrichs@gmail.com> |
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date | Fri, 24 Jun 2016 16:22:17 +0100 |
parents | |
children | 5ef33a8c9702 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/examples/08-PureData/customRender/heavy/render.cpp Fri Jun 24 16:22:17 2016 +0100 @@ -0,0 +1,491 @@ +/* + * render.cpp + * + * Template render.cpp file for on-board heavy compiling + * + * N.B. this is currently *not* compatible with foleyDesigner source files! + * + * Created on: November 5, 2015 + * + * Christian Heinrichs + * + */ + +#include <Bela.h> +#include <Midi.h> +#include <Scope.h> +#include <cmath> +#include <Heavy_bela.h> +#include <string.h> +#include <stdlib.h> +#include <string.h> +#include <DigitalChannelManager.h> + +/* + * MODIFICATION + * ------------ + * Global variables for tremolo effect applied to libpd output + */ + +float gTremoloRate = 4.0; +float gPhase; + +/*********/ + +/* + * HEAVY CONTEXT & BUFFERS + */ + +Hv_bela *gHeavyContext; +float *gHvInputBuffers = NULL, *gHvOutputBuffers = NULL; +unsigned int gHvInputChannels = 0, gHvOutputChannels = 0; + +float gInverseSampleRate; + +/* + * HEAVY FUNCTIONS + */ + +// TODO: rename this +#define LIBPD_DIGITAL_OFFSET 11 // digitals are preceded by 2 audio and 8 analogs (even if using a different number of analogs) + +void printHook(double timestampSecs, const char *printLabel, const char *msgString, void *userData) { + rt_printf("Message from Heavy patch: [@ %.3f] %s: %s\n", timestampSecs, printLabel, msgString); +} + + +// digitals +static DigitalChannelManager dcm; + +void sendDigitalMessage(bool state, unsigned int delay, void* receiverName){ + hv_sendFloatToReceiver(gHeavyContext, hv_stringToHash((char*)receiverName), (float)state); +// rt_printf("%s: %d\n", (char*)receiverName, state); +} + +// TODO: turn them into hv hashes and adjust sendDigitalMessage accordingly +char hvDigitalInHashes[16][21]={ + {"bela_digitalIn11"},{"bela_digitalIn12"},{"bela_digitalIn13"},{"bela_digitalIn14"},{"bela_digitalIn15"}, + {"bela_digitalIn16"},{"bela_digitalIn17"},{"bela_digitalIn18"},{"bela_digitalIn19"},{"bela_digitalIn20"}, + {"bela_digitalIn21"},{"bela_digitalIn22"},{"bela_digitalIn23"},{"bela_digitalIn24"},{"bela_digitalIn25"}, + {"bela_digitalIn26"} +}; + +static void sendHook( + double timestamp, // in milliseconds + const char *receiverName, + const HvMessage *const m, + void *userData) { + + /* + * MODIFICATION + * ------------ + * Parse float sent to receiver 'tremoloRate' and assign it to a global variable + */ + + if(strncmp(receiverName, "tremoloRate", 11) == 0){ + float value = hv_msg_getFloat(m, 0); // see the Heavy C API documentation: https://enzienaudio.com/docs/index.html#8.c + gTremoloRate = value; + } + + /*********/ + + // Bela digital + + // Bela digital run-time messages + + // TODO: this first block is almost an exact copy of libpd's code, should we add this to the class? + // let's make this as optimized as possible for built-in digital Out parsing + // the built-in digital receivers are of the form "bela_digitalOutXX" where XX is between 11 and 26 + static int prefixLength = 15; // strlen("bela_digitalOut") + if(strncmp(receiverName, "bela_digitalOut", prefixLength)==0){ + if(receiverName[prefixLength] != 0){ //the two ifs are used instead of if(strlen(source) >= prefixLength+2) + if(receiverName[prefixLength + 1] != 0){ + // quickly convert the suffix to integer, assuming they are numbers, avoiding to call atoi + int receiver = ((receiverName[prefixLength] - 48) * 10); + receiver += (receiverName[prefixLength+1] - 48); + unsigned int channel = receiver - LIBPD_DIGITAL_OFFSET; // go back to the actual Bela digital channel number + bool value = hv_msg_getFloat(m, 0); + if(channel < 16){ //16 is the hardcoded value for the number of digital channels + dcm.setValue(channel, value); + } + } + } + } + + // Bela digital initialization messages + if(strcmp(receiverName, "bela_setDigital") == 0){ + // Third argument (optional) can be ~ or sig for signal-rate, message-rate otherwise. + // [in 14 ~( + // | + // [s bela_setDigital] + // is signal("sig" or "~") or message("message", default) rate + bool isMessageRate = true; // defaults to message rate + bool direction = 0; // initialize it just to avoid the compiler's warning + bool disable = false; + int numArgs = hv_msg_getNumElements(m); + if(numArgs < 2 || numArgs > 3 || !hv_msg_isSymbol(m, 0) || !hv_msg_isFloat(m, 1)) + return; + if(numArgs == 3 && !hv_msg_isSymbol(m,2)) + return; + char * symbol = hv_msg_getSymbol(m, 0); + + if(strcmp(symbol, "in") == 0){ + direction = INPUT; + } else if(strcmp(symbol, "out") == 0){ + direction = OUTPUT; + } else if(strcmp(symbol, "disable") == 0){ + disable = true; + } else { + return; + } + int channel = hv_msg_getFloat(m, 1) - LIBPD_DIGITAL_OFFSET; + if(disable == true){ + dcm.unmanage(channel); + return; + } + if(numArgs >= 3){ + char* s = hv_msg_getSymbol(m, 2); + if(strcmp(s, "~") == 0 || strncmp(s, "sig", 3) == 0){ + isMessageRate = false; + } + } + dcm.manage(channel, direction, isMessageRate); + } +} + + +/* + * SETUP, RENDER LOOP & CLEANUP + */ + +// leaving this here, trying to come up with a coherent interface with libpd. +// commenting them out so the compiler does not warn +// 2 audio + (up to)8 analog + (up to) 16 digital + 4 scope outputs +//static const unsigned int gChannelsInUse = 30; +//static unsigned int gAnalogChannelsInUse = 8; // hard-coded for the moment, TODO: get it at run-time from hv_context +//static const unsigned int gFirstAudioChannel = 0; +//static const unsigned int gFirstAnalogChannel = 2; +static const unsigned int gFirstDigitalChannel = 10; +static const unsigned int gFirstScopeChannel = 26; +static unsigned int gDigitalSigInChannelsInUse; +static unsigned int gDigitalSigOutChannelsInUse; + +// Bela Midi +Midi midi; +unsigned int hvMidiHashes[7]; +// Bela Scope +Scope scope; +unsigned int gScopeChannelsInUse; +float* gScopeOut; + + +bool setup(BelaContext *context, void *userData) { + if(context->audioInChannels != context->audioOutChannels || + context->analogInChannels != context->analogOutChannels){ + // It should actually work, but let's test it before releasing it! + printf("Error: TODO: a different number of channels for inputs and outputs is not yet supported\n"); + return false; + } + + /* + * MODIFICATION + * ------------ + * Initialise variables for tremolo effect + */ + + gPhase = 0.0; + + /*********/ + + /* HEAVY */ + hvMidiHashes[kmmNoteOn] = hv_stringToHash("__hv_notein"); +// hvMidiHashes[kmmNoteOff] = hv_stringToHash("noteoff"); // this is handled differently, see the render function + hvMidiHashes[kmmControlChange] = hv_stringToHash("__hv_ctlin"); + // Note that the ones below are not defined by Heavy, but they are here for (wishing) forward-compatibility + // You need to receive from the corresponding symbol in Pd and unpack the message, e.g.: + //[r __hv_pgmin] + //| + //[unpack f f] + //| | + //| [print pgmin_channel] + //[print pgmin_number] + hvMidiHashes[kmmProgramChange] = hv_stringToHash("__hv_pgmin"); + hvMidiHashes[kmmPolyphonicKeyPressure] = hv_stringToHash("__hv_polytouchin"); + hvMidiHashes[kmmChannelPressure] = hv_stringToHash("__hv_touchin"); + hvMidiHashes[kmmPitchBend] = hv_stringToHash("__hv_bendin"); + + gHeavyContext = hv_bela_new(context->audioSampleRate); + + gHvInputChannels = hv_getNumInputChannels(gHeavyContext); + gHvOutputChannels = hv_getNumOutputChannels(gHeavyContext); + + gScopeChannelsInUse = gHvOutputChannels > gFirstScopeChannel ? + gHvOutputChannels - gFirstScopeChannel : 0; + gDigitalSigInChannelsInUse = gHvInputChannels > gFirstDigitalChannel ? + gHvInputChannels - gFirstDigitalChannel : 0; + gDigitalSigOutChannelsInUse = gHvOutputChannels > gFirstDigitalChannel ? + gHvOutputChannels - gFirstDigitalChannel - gScopeChannelsInUse: 0; + + printf("Starting Heavy context with %d input channels and %d output channels\n", + gHvInputChannels, gHvOutputChannels); + printf("Channels in use:\n"); + printf("Digital in : %u, Digital out: %u\n", gDigitalSigInChannelsInUse, gDigitalSigOutChannelsInUse); + printf("Scope out: %u\n", gScopeChannelsInUse); + + if(gHvInputChannels != 0) { + gHvInputBuffers = (float *)calloc(gHvInputChannels * context->audioFrames,sizeof(float)); + } + if(gHvOutputChannels != 0) { + gHvOutputBuffers = (float *)calloc(gHvOutputChannels * context->audioFrames,sizeof(float)); + } + + gInverseSampleRate = 1.0 / context->audioSampleRate; + + // Set heavy print hook + hv_setPrintHook(gHeavyContext, printHook); + // Set heavy send hook + hv_setSendHook(gHeavyContext, sendHook); + + // TODO: change these hardcoded port values and actually change them in the Midi class + midi.readFrom(0); + midi.writeTo(0); + midi.enableParser(true); + + if(gScopeChannelsInUse > 0){ + // block below copy/pasted from libpd, except + scope.setup(gScopeChannelsInUse, context->audioSampleRate); + gScopeOut = new float[gScopeChannelsInUse]; + } + // Bela digital + dcm.setCallback(sendDigitalMessage); + if(context->digitalChannels > 0){ + for(unsigned int ch = 0; ch < context->digitalChannels; ++ch){ + dcm.setCallbackArgument(ch, hvDigitalInHashes[ch]); + } + } + // unlike libpd, no need here to bind the bela_digitalOut.. receivers + + return true; +} + + +void render(BelaContext *context, void *userData) +{ + { + int num; + while((num = midi.getParser()->numAvailableMessages()) > 0){ + static MidiChannelMessage message; + message = midi.getParser()->getNextChannelMessage(); + switch(message.getType()){ + case kmmNoteOn: { + //message.prettyPrint(); + int noteNumber = message.getDataByte(0); + int velocity = message.getDataByte(1); + int channel = message.getChannel(); + // rt_printf("message: noteNumber: %f, velocity: %f, channel: %f\n", noteNumber, velocity, channel); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmNoteOn], 0, "fff", + (float)noteNumber, (float)velocity, (float)channel+1); + break; + } + case kmmNoteOff: { + /* PureData does not seem to handle noteoff messages as per the MIDI specs, + * so that the noteoff velocity is ignored. Here we convert them to noteon + * with a velocity of 0. + */ + int noteNumber = message.getDataByte(0); + // int velocity = message.getDataByte(1); // would be ignored by Pd + int channel = message.getChannel(); + // note we are sending the below to hvHashes[kmmNoteOn] !! + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmNoteOn], 0, "fff", + (float)noteNumber, (float)0, (float)channel+1); + break; + } + case kmmControlChange: { + int channel = message.getChannel(); + int controller = message.getDataByte(0); + int value = message.getDataByte(1); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmControlChange], 0, "fff", + (float)value, (float)controller, (float)channel+1); + break; + } + case kmmProgramChange: { + int channel = message.getChannel(); + int program = message.getDataByte(0); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmProgramChange], 0, "ff", + (float)program, (float)channel+1); + break; + } + case kmmPolyphonicKeyPressure: { + //TODO: untested, I do not have anything with polyTouch... who does, anyhow? + int channel = message.getChannel(); + int pitch = message.getDataByte(0); + int value = message.getDataByte(1); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmPolyphonicKeyPressure], 0, "fff", + (float)channel+1, (float)pitch, (float)value); + break; + } + case kmmChannelPressure: + { + int channel = message.getChannel(); + int value = message.getDataByte(0); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmChannelPressure], 0, "ff", + (float)value, (float)channel+1); + break; + } + case kmmPitchBend: + { + int channel = message.getChannel(); + int value = ((message.getDataByte(1) << 7) | message.getDataByte(0)); + hv_vscheduleMessageForReceiver(gHeavyContext, hvMidiHashes[kmmPitchBend], 0, "ff", + (float)value, (float)channel+1); + break; + } + case kmmNone: + case kmmAny: + break; + } + } + } + + // De-interleave the data + if(gHvInputBuffers != NULL) { + for(unsigned int n = 0; n < context->audioFrames; n++) { + for(unsigned int ch = 0; ch < gHvInputChannels; ch++) { + if(ch >= context->audioInChannels+context->analogInChannels) { + // THESE ARE PARAMETER INPUT 'CHANNELS' USED FOR ROUTING + // 'sensor' outputs from routing channels of dac~ are passed through here + break; + } else { + // If more than 2 ADC inputs are used in the pd patch, route the analog inputs + // i.e. ADC3->analogIn0 etc. (first two are always audio inputs) + if(ch >= context->audioInChannels) { + int m = n/2; + float mIn = context->analogIn[m*context->analogInChannels + (ch-context->audioInChannels)]; + gHvInputBuffers[ch * context->audioFrames + n] = mIn; + } else { + gHvInputBuffers[ch * context->audioFrames + n] = context->audioIn[n * context->audioInChannels + ch]; + } + } + } + } + } + + // Bela digital in + // note: in multiple places below we assume that the number of digital frames is same as number of audio + // Bela digital in at message-rate + dcm.processInput(context->digital, context->digitalFrames); + + // Bela digital in at signal-rate + if(gDigitalSigInChannelsInUse > 0) + { + unsigned int j, k; + float *p0, *p1; + const unsigned int gLibpdBlockSize = context->audioFrames; + const unsigned int audioFrameBase = 0; + float* gInBuf = gHvInputBuffers; + // block below copy/pasted from libpd, except + // 16 has been replaced with gDigitalSigInChannelsInUse + for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) { + unsigned int digitalFrame = audioFrameBase + j; + for (k = 0, p1 = p0 + gLibpdBlockSize * gFirstDigitalChannel; + k < gDigitalSigInChannelsInUse; ++k, p1 += gLibpdBlockSize) { + if(dcm.isSignalRate(k) && dcm.isInput(k)){ // only process input channels that are handled at signal rate + *p1 = digitalRead(context, digitalFrame, k); + } + } + } + } + + + // replacement for bang~ object + //hv_vscheduleMessageForReceiver(gHeavyContext, "bela_bang", 0.0f, "b"); + + hv_bela_process_inline(gHeavyContext, gHvInputBuffers, gHvOutputBuffers, context->audioFrames); + + // Bela digital out + // Bela digital out at signal-rate + if(gDigitalSigOutChannelsInUse > 0) + { + unsigned int j, k; + float *p0, *p1; + const unsigned int gLibpdBlockSize = context->audioFrames; + const unsigned int audioFrameBase = 0; + float* gOutBuf = gHvOutputBuffers; + // block below copy/pasted from libpd, except + // context->digitalChannels has been replaced with gDigitalSigOutChannelsInUse + for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) { + unsigned int digitalFrame = (audioFrameBase + j); + for (k = 0, p1 = p0 + gLibpdBlockSize * gFirstDigitalChannel; + k < gDigitalSigOutChannelsInUse; k++, p1 += gLibpdBlockSize) { + if(dcm.isSignalRate(k) && dcm.isOutput(k)){ // only process output channels that are handled at signal rate + digitalWriteOnce(context, digitalFrame, k, *p1 > 0.5); + } + } + } + } + // Bela digital out at message-rate + dcm.processOutput(context->digital, context->digitalFrames); + + // Bela scope + if(gScopeChannelsInUse > 0) + { + unsigned int j, k; + float *p0, *p1; + const unsigned int gLibpdBlockSize = context->audioFrames; + float* gOutBuf = gHvOutputBuffers; + + // block below copy/pasted from libpd + for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) { + for (k = 0, p1 = p0 + gLibpdBlockSize * gFirstScopeChannel; k < gScopeChannelsInUse; k++, p1 += gLibpdBlockSize) { + gScopeOut[k] = *p1; + } + scope.log(gScopeOut); + } + } + + // Interleave the output data + if(gHvOutputBuffers != NULL) { + for(unsigned int n = 0; n < context->audioFrames; n++) { + + /* + * MODIFICATION + * ------------ + * Processing for tremolo effect while writing libpd output to Bela output buffer + */ + + // Generate a sinewave with frequency set by gTremoloRate + // and amplitude from -0.5 to 0.5 + float lfo = sinf(gPhase) * 0.5; + // Keep track and wrap the phase of the sinewave + gPhase += 2.0 * M_PI * gTremoloRate * gInverseSampleRate; + if(gPhase > 2.0 * M_PI) + gPhase -= 2.0 * M_PI; + + /*********/ + + for(unsigned int ch = 0; ch < gHvOutputChannels; ch++) { + if(ch <= context->audioOutChannels+context->analogOutChannels) { + if(ch >= context->audioOutChannels) { + int m = n/2; + context->analogOut[m * context->analogFrames + (ch-context->audioOutChannels)] = constrain(gHvOutputBuffers[ch*context->audioFrames + n],0.0,1.0); + } else { + context->audioOut[n * context->audioOutChannels + ch] = gHvOutputBuffers[ch * context->audioFrames + n] * lfo; // MODIFICATION (* lfo) + } + } + } + } + } + +} + + +void cleanup(BelaContext *context, void *userData) +{ + + hv_bela_free(gHeavyContext); + if(gHvInputBuffers != NULL) + free(gHvInputBuffers); + if(gHvOutputBuffers != NULL) + free(gHvOutputBuffers); + delete[] gScopeOut; +}