#include #include "teensy-speaker.h" #include "teensy-println.h" #include #include TeensyAudio audioDriver; //AudioMixer4 mixer2; //xy=280,253 AudioMixer4 mixer1; //xy=280,175 AudioOutputI2S i2s; //xy=452,189 AudioConnection patchCord1(audioDriver, 0, mixer1, 0); //AudioConnection patchCord2(audioDriver, 0, mixer2, 0); //AudioConnection patchCord3(mixer2, 0, i2s, 1); AudioConnection patchCord4(mixer1, 0, i2s, 0); #include "globals.h" #define HIGHVAL (0x4FFF) #define LOWVAL (-0x4FFF) // Ring buffer that we fill with 44.1kHz data #define BUFSIZE 4096 static volatile uint32_t bufIdx; // 0 .. BUFSIZE-1 static volatile uint32_t skippedSamples; // Who knows where this will // wind up (FIXME: eventual // rollover means we need a // way to purge the queue // when it's quiescent for // too long & restart all the // constants) static volatile uint8_t audioRunning = 0; // FIXME: needs constants abstracted static volatile uint32_t lastFilledTime = 0; #define SAMPLEBYTES sizeof(short) EXTMEM short soundBuf[BUFSIZE]; static bool toggleState = false; TeensySpeaker::TeensySpeaker(uint8_t sda, uint8_t scl) : PhysicalSpeaker() { toggleState = false; mixerValue = numMixed = 0; AudioMemory(8); } TeensySpeaker::~TeensySpeaker() { } void TeensySpeaker::begin() { mixer1.gain(0, 0.1f); // left channel toggleState = false; bufIdx = 0; skippedSamples = 0; audioRunning = 0; } void TeensySpeaker::toggle(uint32_t c) { // Figure out when the last time was that we put data in the audio buffer; // then figure out how many audio buffer cycles we have to fill from that // CPU time to this one. #if 1 __disable_irq(); // We expect to have filled to this cycle number... uint32_t expectedCycleNumber = (float)c * (float)AUDIO_SAMPLE_RATE_EXACT / (float)g_speed; // Dynamically initialize the lastFilledTime based on the start time of the // audio channel. if (lastFilledTime == 0) lastFilledTime = expectedCycleNumber; // and we have filled to cycle number lastFilledTime. So how many do // we need? This subtracts skippedSamples because those were filled // automatically by the audioCallback when we had no data. int32_t audioBufferSamples = expectedCycleNumber - lastFilledTime - skippedSamples; // If audioBufferSamples < 0, then we need to keep some // skippedSamples for later; otherwise we can keep moving forward. if (audioBufferSamples < 0) { skippedSamples = -audioBufferSamples; audioBufferSamples = 0; } else { // Otherwise we consumed them and can forget about it. skippedSamples = 0; } int32_t newIdx = bufIdx + audioBufferSamples; if (audioBufferSamples == 0) { // If the toggle wouldn't result in at least 1 buffer sample // change, then we'll blatantly skip it here. If this turns out to // be a problem, we could try setting audioBufferSamples++ and // then twiddle the lastFilledTime so it looks like it's more in // the future, but I suspect that would mean missing more future // events, just like we would have missed this one. // // But I think this is probably okay - because something that's // toggling the speaker fast enough that our 44k audio can't keep // up with the individual changes is likely to toggle again in a // moment without significant distortion? return; } if (newIdx >= BUFSIZE) { // Buffer overrun error. Shouldn't happen? newIdx = BUFSIZE - 1; } lastFilledTime = expectedCycleNumber; // Flip the toggle state toggleState = !toggleState; // Fill from bufIdx .. newIdx and set bufIdx to newIdx when done. if (newIdx > bufIdx) { long count = (long)newIdx - bufIdx; for (long i=0; idata; memset(bp, 0, AUDIO_BLOCK_SAMPLES * SAMPLEBYTES); transmit(block, 0); release(block); } return; } if (audioRunning == 1 && bufIdx >= AUDIO_BLOCK_SAMPLES) { // We have enough samples in the buffer to fill it, so we're fully // up and running. audioRunning = 2; } else if (audioRunning == 1) { // Still waiting for the first fill; return an empty buffer. block = allocate(); if (block) { bp = block->data; memset(bp, 0, AUDIO_BLOCK_SAMPLES * SAMPLEBYTES); transmit(block, 0); release(block); } return; } block = allocate(); if (block) { bp = block->data; static short lastKnownSample = 0; if (bufIdx >= AUDIO_BLOCK_SAMPLES) { memcpy(bp, (void *)soundBuf, AUDIO_BLOCK_SAMPLES * SAMPLEBYTES); lastKnownSample = bp[AUDIO_BLOCK_SAMPLES-1]; if (bufIdx > AUDIO_BLOCK_SAMPLES) { // move the remaining data down memcpy((void *)soundBuf, (void *)&soundBuf[AUDIO_BLOCK_SAMPLES], (bufIdx - AUDIO_BLOCK_SAMPLES + 1)*SAMPLEBYTES); bufIdx -= AUDIO_BLOCK_SAMPLES; } } else { if (bufIdx) { // partial buffer exists memcpy(bp, (void *)soundBuf, bufIdx * SAMPLEBYTES); // and it's a partial underrun. Track the number of samples we skipped // so we can keep the audio buffer in sync. skippedSamples += AUDIO_BLOCK_SAMPLES - bufIdx; for (int32_t i=0; i