ref: 0b2ada50ccd53135668f3023bf7c257a6ad34966
dir: /src/pt2_audio.c/
// for finding memory leaks in debug mode with Visual Studio
#if defined _DEBUG && defined _MSC_VER
#include <crtdbg.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <stdbool.h>
#include <math.h>
#include <SDL2/SDL.h>
#ifdef _WIN32
#include <io.h>
#else
#include <unistd.h>
#endif
#include <fcntl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <limits.h>
#include "pt2_audio.h"
#include "pt2_helpers.h"
#include "pt2_config.h"
#include "pt2_textout.h"
#include "pt2_scopes.h"
#include "pt2_visuals_sync.h"
#include "pt2_downsample2x.h"
#include "pt2_replayer.h"
#include "pt2_paula.h"
// cumulative mid/side normalization factor (1/sqrt(2))*(1/sqrt(2))
#define STEREO_NORM_FACTOR 0.5
#define INITIAL_DITHER_SEED 0x12345000
static uint8_t panningMode;
static int32_t randSeed = INITIAL_DITHER_SEED, stereoSeparation = 100;
static double *dMixBufferL, *dMixBufferR;
static double dPrngStateL, dPrngStateR, dSideFactor;
static SDL_AudioDeviceID dev;
audio_t audio; // globalized
void setAmigaFilterModel(uint8_t model)
{
if (audio.amigaModel == model)
return; // same state as before!
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
audio.amigaModel = model;
const int32_t paulaMixFrequency = audio.oversamplingFlag ? audio.outputRate*2 : audio.outputRate;
paulaSetup(paulaMixFrequency, audio.amigaModel);
if (audioWasntLocked)
unlockAudio();
}
void toggleAmigaFilterModel(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
audio.amigaModel ^= 1;
const int32_t paulaMixFrequency = audio.oversamplingFlag ? audio.outputRate*2 : audio.outputRate;
paulaSetup(paulaMixFrequency, audio.amigaModel);
if (audioWasntLocked)
unlockAudio();
if (audio.amigaModel == MODEL_A500)
displayMsg("AUDIO: AMIGA 500");
else
displayMsg("AUDIO: AMIGA 1200");
}
void setLEDFilter(bool state)
{
if (audio.ledFilterEnabled == state)
return; // same state as before!
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
audio.ledFilterEnabled = state;
paulaWriteByte(0xBFE001, audio.ledFilterEnabled << 1);
if (audioWasntLocked)
unlockAudio();
}
void toggleLEDFilter(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
audio.ledFilterEnabled ^= 1;
paulaWriteByte(0xBFE001, audio.ledFilterEnabled << 1);
if (audioWasntLocked)
unlockAudio();
}
void lockAudio(void)
{
if (dev != 0)
SDL_LockAudioDevice(dev);
audio.locked = true;
audio.resetSyncTickTimeFlag = true;
resetChSyncQueue();
}
void unlockAudio(void)
{
if (dev != 0)
SDL_UnlockAudioDevice(dev);
audio.resetSyncTickTimeFlag = true;
resetChSyncQueue();
audio.locked = false;
}
void resetAudioDithering(void)
{
randSeed = INITIAL_DITHER_SEED;
dPrngStateL = 0.0;
dPrngStateR = 0.0;
}
static inline int32_t random32(void)
{
// LCG random 32-bit generator (quite good and fast)
randSeed *= 134775813;
randSeed++;
return randSeed;
}
#define NORM_FACTOR 2.0 /* nominally correct, but can clip from high-pass filter overshoot */
static inline void processMixedSamplesAmigaPanning(int32_t i, int16_t *out)
{
int32_t smp32;
double dPrng;
double dL = dMixBufferL[i];
double dR = dMixBufferR[i];
// normalize w/ phase-inversion (A500/A1200 has a phase-inverted audio signal)
dL *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
dR *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dL = (dL + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dL;
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dR = (dR + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dR;
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamples(int32_t i, int16_t *out)
{
int32_t smp32;
double dPrng;
double dL = dMixBufferL[i];
double dR = dMixBufferR[i];
// apply stereo separation
const double dOldL = dL;
const double dOldR = dR;
double dMid = (dOldL + dOldR) * STEREO_NORM_FACTOR;
double dSide = (dOldL - dOldR) * dSideFactor;
dL = dMid + dSide;
dR = dMid - dSide;
// normalize w/ phase-inversion (A500/A1200 has a phase-inverted audio signal)
dL *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
dR *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dL = (dL + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dL;
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dR = (dR + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dR;
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamplesAmigaPanning_2x(int32_t i, int16_t *out) // 2x oversampling
{
int32_t smp32;
double dPrng, dL, dR;
// 2x downsampling (decimation)
const uint32_t offset1 = (i << 1) + 0;
const uint32_t offset2 = (i << 1) + 1;
dL = decimate2x_L(dMixBufferL[offset1], dMixBufferL[offset2]);
dR = decimate2x_R(dMixBufferR[offset1], dMixBufferR[offset2]);
// normalize w/ phase-inversion (A500/A1200 has a phase-inverted audio signal)
dL *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
dR *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dL = (dL + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dL;
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dR = (dR + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dR;
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
static inline void processMixedSamples_2x(int32_t i, int16_t *out) // 2x oversampling
{
int32_t smp32;
double dPrng, dL, dR;
// 2x downsampling (decimation)
const uint32_t offset1 = (i << 1) + 0;
const uint32_t offset2 = (i << 1) + 1;
dL = decimate2x_L(dMixBufferL[offset1], dMixBufferL[offset2]);
dR = decimate2x_R(dMixBufferR[offset1], dMixBufferR[offset2]);
// apply stereo separation
const double dOldL = dL;
const double dOldR = dR;
double dMid = (dOldL + dOldR) * STEREO_NORM_FACTOR;
double dSide = (dOldL - dOldR) * dSideFactor;
dL = dMid + dSide;
dR = dMid - dSide;
// normalize w/ phase-inversion (A500/A1200 has a phase-inverted audio signal)
dL *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
dR *= NORM_FACTOR * (-INT16_MAX / (double)PAULA_VOICES);
// left channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dL = (dL + dPrng) - dPrngStateL;
dPrngStateL = dPrng;
smp32 = (int32_t)dL;
CLAMP16(smp32);
out[0] = (int16_t)smp32;
// right channel - 1-bit triangular dithering (high-pass filtered)
dPrng = random32() * (0.5 / INT32_MAX); // -0.5..0.5
dR = (dR + dPrng) - dPrngStateR;
dPrngStateR = dPrng;
smp32 = (int32_t)dR;
CLAMP16(smp32);
out[1] = (int16_t)smp32;
}
void outputAudio(int16_t *target, int32_t numSamples)
{
if (audio.oversamplingFlag) // 2x oversampling
{
paulaGenerateSamples(dMixBufferL, dMixBufferR, numSamples*2);
// downsample, normalize and dither
int16_t out[2];
int16_t *outStream = target;
if (stereoSeparation == 100)
{
for (int32_t i = 0; i < numSamples; i++)
{
processMixedSamplesAmigaPanning_2x(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
else
{
for (int32_t i = 0; i < numSamples; i++)
{
processMixedSamples_2x(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
}
else
{
paulaGenerateSamples(dMixBufferL, dMixBufferR, numSamples);
// normalize and dither
int16_t out[2];
int16_t *outStream = target;
if (stereoSeparation == 100)
{
for (int32_t i = 0; i < numSamples; i++)
{
processMixedSamplesAmigaPanning(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
else
{
for (int32_t i = 0; i < numSamples; i++)
{
processMixedSamples(i, out);
*outStream++ = out[0];
*outStream++ = out[1];
}
}
}
}
static void SDLCALL audioCallback(void *userdata, Uint8 *stream, int len)
{
if (editor.mod2WavOngoing || editor.pat2SmpOngoing) // send silence to sound output device
{
memset(stream, 0, len);
return;
}
int16_t *streamOut = (int16_t *)stream;
uint32_t samplesLeft = (uint32_t)len >> 2;
while (samplesLeft > 0)
{
if (audio.tickSampleCounter64 <= 0)
{
// new replayer tick
if (editor.songPlaying)
{
intMusic(); // PT replayer ticker
fillVisualsSyncBuffer();
}
audio.tickSampleCounter64 += audio.samplesPerTick64;
}
const uint32_t remainingTickSamples = ((uint64_t)audio.tickSampleCounter64 + UINT32_MAX) >> 32; // ceil rounding (upwards)
uint32_t samplesTodo = samplesLeft;
if (samplesTodo > remainingTickSamples)
samplesTodo = remainingTickSamples;
outputAudio(streamOut, samplesTodo);
streamOut += samplesTodo << 1;
samplesLeft -= samplesTodo;
audio.tickSampleCounter64 -= (int64_t)samplesTodo << 32;
}
(void)userdata;
}
void audioSetStereoSeparation(uint8_t percentage) // 0..100 (percentage)
{
assert(percentage <= 100);
stereoSeparation = percentage;
dSideFactor = (percentage / 100.0) * STEREO_NORM_FACTOR;
}
void generateBpmTable(double dAudioFreq, bool vblankTimingFlag)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
for (int32_t bpm = 32; bpm <= 255; bpm++)
{
double dBpmHz;
if (vblankTimingFlag)
dBpmHz = AMIGA_PAL_VBLANK_HZ;
else
dBpmHz = ciaBpm2Hz(bpm);
const double dSamplesPerTick = dAudioFreq / dBpmHz;
// convert to rounded 32.32 fixed-point
const int32_t i = bpm - 32;
audio.samplesPerTickTable[i] = (int64_t)((dSamplesPerTick * (UINT32_MAX+1.0)) + 0.5);
}
audio.tickSampleCounter64 = 0;
if (audioWasntLocked)
unlockAudio();
}
static void generateTickLengthTable(bool vblankTimingFlag)
{
for (int32_t bpm = 32; bpm <= 255; bpm++)
{
double dHz;
if (vblankTimingFlag)
dHz = AMIGA_PAL_VBLANK_HZ;
else
dHz = ciaBpm2Hz(bpm);
// BPM -> Hz -> tick length for performance counter (syncing visuals to audio)
double dTimeInt;
double dTimeFrac = modf((double)hpcFreq.freq64 / dHz, &dTimeInt);
const int32_t timeInt = (int32_t)dTimeInt;
dTimeFrac = floor((dTimeFrac * (UINT32_MAX+1.0)) + 0.5); // fractional part (scaled to 0..2^32-1)
audio.tickLengthTable[bpm-32] = ((uint64_t)timeInt << 32) | (uint32_t)dTimeFrac;
}
}
void updateReplayerTimingMode(void)
{
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
const bool vblankTimingMode = (editor.timingMode == TEMPO_MODE_VBLANK);
generateBpmTable(audio.outputRate, vblankTimingMode);
generateTickLengthTable(vblankTimingMode);
if (audioWasntLocked)
unlockAudio();
}
bool setupAudio(void)
{
SDL_AudioSpec want, have;
want.freq = config.soundFrequency;
want.samples = (uint16_t)config.soundBufferSize;
want.format = AUDIO_S16;
want.channels = 2;
want.callback = audioCallback;
want.userdata = NULL;
dev = SDL_OpenAudioDevice(NULL, 0, &want, &have, 0);
if (dev == 0)
{
showErrorMsgBox("Unable to open audio device: %s", SDL_GetError());
return false;
}
// lower than this is not safe for the BLEP synthesis in the mixer
const int32_t minFreq = (int32_t)(PAULA_PAL_CLK / 113.0 / 2.0)+1; // /2 because we do 2x oversampling
if (have.freq < minFreq)
{
showErrorMsgBox("Unable to open audio: An audio rate below %dHz can't be used!", minFreq);
return false;
}
if (have.format != want.format)
{
showErrorMsgBox("Unable to open audio: The sample format (signed 16-bit) couldn't be used!");
return false;
}
audio.outputRate = have.freq;
audio.audioBufferSize = have.samples;
audio.oversamplingFlag = (audio.outputRate < 96000); // we do 2x oversampling if the audio output rate is below 96kHz
audio.amigaModel = config.amigaModel;
const int32_t paulaMixFrequency = audio.oversamplingFlag ? audio.outputRate*2 : audio.outputRate;
const uint32_t maxSamplesToMix = (int32_t)ceil(paulaMixFrequency / (REPLAYER_MIN_BPM / 2.5));
dMixBufferL = (double *)malloc((maxSamplesToMix + 1) * sizeof (double));
dMixBufferR = (double *)malloc((maxSamplesToMix + 1) * sizeof (double));
if (dMixBufferL == NULL || dMixBufferR == NULL)
{
// these two are free'd later
showErrorMsgBox("Out of memory!");
return false;
}
paulaSetup(paulaMixFrequency, audio.amigaModel);
audioSetStereoSeparation(config.stereoSeparation);
updateReplayerTimingMode(); // also generates the BPM table (audio.bpmTable)
setLEDFilter(false);
calcAudioLatencyVars(audio.audioBufferSize, audio.outputRate);
clearMixerDownsamplerStates();
audio.resetSyncTickTimeFlag = true;
audio.samplesPerTick64 = audio.samplesPerTickTable[125-32]; // BPM 125
audio.tickSampleCounter64 = 0; // zero tick sample counter so that it will instantly initiate a tick
SDL_PauseAudioDevice(dev, false);
return true;
}
void audioClose(void)
{
if (dev > 0)
{
SDL_PauseAudioDevice(dev, true);
SDL_CloseAudioDevice(dev);
dev = 0;
}
if (dMixBufferL != NULL)
{
free(dMixBufferL);
dMixBufferL = NULL;
}
if (dMixBufferR != NULL)
{
free(dMixBufferR);
dMixBufferR = NULL;
}
}
void toggleAmigaPanMode(void)
{
panningMode = (panningMode + 1) % 3;
const bool audioWasntLocked = !audio.locked;
if (audioWasntLocked)
lockAudio();
if (panningMode == 0)
audioSetStereoSeparation(config.stereoSeparation);
else if (panningMode == 1)
audioSetStereoSeparation(0);
else
audioSetStereoSeparation(100);
if (audioWasntLocked)
unlockAudio();
if (panningMode == 0)
displayMsg("CUSTOM PANNING");
else if (panningMode == 1)
displayMsg("CENTERED PANNING");
else
displayMsg("AMIGA PANNING");
}
uint16_t get16BitPeak(int16_t *sampleData, uint32_t sampleLength)
{
uint16_t samplePeak = 0;
for (uint32_t i = 0; i < sampleLength; i++)
{
uint16_t sample = ABS(sampleData[i]);
if (samplePeak < sample)
samplePeak = sample;
}
return samplePeak;
}
uint32_t get32BitPeak(int32_t *sampleData, uint32_t sampleLength)
{
uint32_t samplePeak = 0;
for (uint32_t i = 0; i < sampleLength; i++)
{
uint32_t sample = ABS(sampleData[i]);
if (samplePeak < sample)
samplePeak = sample;
}
return samplePeak;
}
float getFloatPeak(float *fSampleData, uint32_t sampleLength)
{
float fSamplePeak = 0.0f;
for (uint32_t i = 0; i < sampleLength; i++)
{
const float fSample = fabsf(fSampleData[i]);
if (fSamplePeak < fSample)
fSamplePeak = fSample;
}
return fSamplePeak;
}
double getDoublePeak(double *dSampleData, uint32_t sampleLength)
{
double dSamplePeak = 0.0;
for (uint32_t i = 0; i < sampleLength; i++)
{
const double dSample = fabs(dSampleData[i]);
if (dSamplePeak < dSample)
dSamplePeak = dSample;
}
return dSamplePeak;
}
void normalize16BitTo8Bit(int16_t *sampleData, uint32_t sampleLength)
{
const uint16_t samplePeak = get16BitPeak(sampleData, sampleLength);
if (samplePeak == 0 || samplePeak >= INT16_MAX)
return;
const double dGain = (double)INT16_MAX / samplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
{
const int32_t sample = (const int32_t)(sampleData[i] * dGain);
sampleData[i] = (int16_t)sample;
}
}
void normalize32BitTo8Bit(int32_t *sampleData, uint32_t sampleLength)
{
const uint32_t samplePeak = get32BitPeak(sampleData, sampleLength);
if (samplePeak == 0 || samplePeak >= INT32_MAX)
return;
const double dGain = (double)INT32_MAX / samplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
{
const int32_t sample = (const int32_t)(sampleData[i] * dGain);
sampleData[i] = (int32_t)sample;
}
}
void normalizeFloatTo8Bit(float *fSampleData, uint32_t sampleLength)
{
const float fSamplePeak = getFloatPeak(fSampleData, sampleLength);
if (fSamplePeak <= 0.0f)
return;
const float fGain = INT8_MAX / fSamplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
fSampleData[i] *= fGain;
}
void normalizeDoubleTo8Bit(double *dSampleData, uint32_t sampleLength)
{
const double dSamplePeak = getDoublePeak(dSampleData, sampleLength);
if (dSamplePeak <= 0.0)
return;
const double dGain = INT8_MAX / dSamplePeak;
for (uint32_t i = 0; i < sampleLength; i++)
dSampleData[i] *= dGain;
}