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Merge pull request #8950 from hrydgard/audio-linear-interpolation
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WIP: SasAudio: Implement linear interpolation
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@hrydgard
hrydgard authored Dec 20, 2016
2 parents e9bea75 + 1e09595 commit 8222a66
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Showing 2 changed files with 33 additions and 53 deletions.
80 changes: 30 additions & 50 deletions Core/HW/SasAudio.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -337,7 +337,6 @@ SasInstance::SasInstance()
sendBuffer(0),
sendBufferDownsampled(0),
sendBufferProcessed(0),
resampleBuffer(0),
grainSize(0) {
#ifdef AUDIO_TO_FILE
audioDump = fopen("D:\\audio.raw", "wb");
Expand Down Expand Up @@ -376,10 +375,8 @@ void SasInstance::ClearGrainSize() {
delete[] sendBuffer;
delete[] sendBufferDownsampled;
delete[] sendBufferProcessed;
delete[] resampleBuffer;
mixBuffer = nullptr;
sendBuffer = nullptr;
resampleBuffer = nullptr;
sendBufferDownsampled = nullptr;
sendBufferProcessed = nullptr;
}
Expand All @@ -392,7 +389,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
delete[] sendBuffer;
delete[] sendBufferDownsampled;
delete[] sendBufferProcessed;
delete[] resampleBuffer;

mixBuffer = new s32[grainSize * 2];
sendBuffer = new s32[grainSize * 2];
Expand All @@ -402,10 +398,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
memset(sendBuffer, 0, sizeof(int) * grainSize * 2);
memset(sendBufferDownsampled, 0, sizeof(s16) * grainSize);
memset(sendBufferProcessed, 0, sizeof(s16) * grainSize * 2);

// 2 samples padding at the start, that's where we copy the two last samples from the channel
// so that we can do bicubic resampling if necessary. Plus 1 for smoothness hackery.
resampleBuffer = new s16[grainSize * 4 + 3];
}

int SasInstance::EstimateMixUs() {
Expand Down Expand Up @@ -459,9 +451,7 @@ void SasVoice::ReadSamples(s16 *output, int numSamples) {
atrac3.getNextSamples(output, numSamples);
break;
default:
{
memset(output, 0, numSamples * sizeof(s16));
}
memset(output, 0, numSamples * sizeof(s16));
break;
}
}
Expand Down Expand Up @@ -493,47 +483,35 @@ void SasInstance::MixVoice(SasVoice &voice) {
break;
// else fallthrough! Don't change the check above.
default:
// Load resample history (so we can use a wide filter)
resampleBuffer[0] = voice.resampleHist[0];
resampleBuffer[1] = voice.resampleHist[1];

// Figure out number of samples to read.
// Actually this is not entirely correct - we need to get one extra sample, and store it
// for the next time around. A little complicated...
// But for now, see Smoothness HACKERY below :P
u32 numSamples = ((u32)voice.sampleFrac + (u32)grainSize * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
if ((int)numSamples > grainSize * 4) {
ERROR_LOG(SASMIX, "numSamples too large, clamping: %i vs %i", numSamples, grainSize * 4);
numSamples = grainSize * 4;
}

// This feels a bit hacky. The first 32 samples after a keyon are 0s.
const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
int delay = 0;
if (voice.envelope.NeedsKeyOn()) {
int delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
// VAG seems to have an extra sample delay (not shared by PCM.)
if (voice.type == VOICETYPE_VAG)
++delay;
voice.ReadSamples(resampleBuffer + 2 + delay, numSamples - delay);
} else {
voice.ReadSamples(resampleBuffer + 2, numSamples);
}

// Smoothness HACKERY
resampleBuffer[2 + numSamples] = resampleBuffer[2 + numSamples - 1];

// Save resample history
voice.resampleHist[0] = resampleBuffer[2 + numSamples - 2];
voice.resampleHist[1] = resampleBuffer[2 + numSamples - 1];

// Resample to the correct pitch, writing exactly "grainSize" samples.
// This is a HORRIBLE resampler by the way.
// TODO: Special case no-resample case (and 2x and 0.5x) for speed, it's not uncommon
int16_t temp[PSP_SAS_MAX_GRAIN + 2];

// Two passes: First read, then resample.
u32 sampleFrac = voice.sampleFrac;
for (int i = 0; i < grainSize; i++) {
// For now: nearest neighbour, not even using the resample history at all.
int sample = resampleBuffer[sampleFrac / PSP_SAS_PITCH_BASE + 2];
temp[0] = voice.resampleHist[0];
temp[1] = voice.resampleHist[1];

int samplesToRead = (sampleFrac + voice.pitch * (grainSize - delay)) >> PSP_SAS_PITCH_BASE_SHIFT;
voice.ReadSamples(&temp[2], samplesToRead);
int tempPos = 2 + samplesToRead;

for (int i = delay; i < grainSize; i++) {
const int16_t *s = temp + (sampleFrac >> PSP_SAS_PITCH_BASE_SHIFT);

// Linear interpolation. Good enough. Need to make resampleHist bigger if we want more.
int f = sampleFrac & PSP_SAS_PITCH_MASK;
int sample = (s[0] * (PSP_SAS_PITCH_MASK - f) + s[1] * f) >> PSP_SAS_PITCH_BASE_SHIFT;
sampleFrac += voice.pitch;

// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
Expand All @@ -549,21 +527,20 @@ void SasInstance::MixVoice(SasVoice &voice) {
// We mix into this 32-bit temp buffer and clip in a second loop
// Ideally, the shift right should be there too but for now I'm concerned about
// not overflowing.
mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> 12;
mixBuffer[i * 2] += (sample * voice.volumeLeft) >> 12;
mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> 12;
sendBuffer[i * 2] += sample * voice.effectLeft >> 12;
sendBuffer[i * 2 + 1] += sample * voice.effectRight >> 12;
}

voice.sampleFrac = sampleFrac;
// Let's hope grainSize is a power of 2.
//voice.sampleFrac &= grainSize * PSP_SAS_PITCH_BASE - 1;
voice.sampleFrac -= numSamples * PSP_SAS_PITCH_BASE;
voice.resampleHist[0] = temp[tempPos - 2];
voice.resampleHist[1] = temp[tempPos - 1];

voice.sampleFrac = sampleFrac - (tempPos - 2) * PSP_SAS_PITCH_BASE;;

if (voice.HaveSamplesEnded())
voice.envelope.End();
if (voice.envelope.HasEnded())
{
if (voice.envelope.HasEnded()) {
// NOTICE_LOG(SCESAS, "Hit end of envelope");
voice.playing = false;
voice.on = false;
Expand Down Expand Up @@ -711,8 +688,11 @@ void SasInstance::DoState(PointerWrap &p) {
if (sendBuffer != NULL && grainSize > 0) {
p.DoArray(sendBuffer, grainSize * 2);
}
if (resampleBuffer != NULL && grainSize > 0) {
p.DoArray(resampleBuffer, grainSize * 4 + 3);
if (sendBuffer != NULL && grainSize > 0) {
// Backwards compat
int16_t *resampleBuf = new int16_t[grainSize * 4 + 3]();
p.DoArray(resampleBuf, grainSize * 4 + 3);
delete[] resampleBuf;
}

int n = PSP_SAS_VOICES_MAX;
Expand Down
6 changes: 3 additions & 3 deletions Core/HW/SasAudio.h
View file
Original file line number Diff line number Diff line change
Expand Up @@ -33,10 +33,12 @@ enum {

PSP_SAS_PITCH_MIN = 0x0000,
PSP_SAS_PITCH_BASE = 0x1000,
PSP_SAS_PITCH_MASK = 0xFFF,
PSP_SAS_PITCH_BASE_SHIFT = 12,
PSP_SAS_PITCH_MAX = 0x4000,

PSP_SAS_VOL_MAX = 0x1000,
PSP_SAS_MAX_GRAIN = 1024, // VERY conservative! 256 is quite common but don't think I've ever seen bigger.

PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE = 0,
PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE = 1,
Expand Down Expand Up @@ -250,7 +252,7 @@ struct SasVoice {
int pcmLoopPos;
int sampleRate;

int sampleFrac;
uint32_t sampleFrac;
int pitch;
bool loop;

Expand Down Expand Up @@ -289,8 +291,6 @@ class SasInstance {
s16 *sendBufferDownsampled;
s16 *sendBufferProcessed;

s16 *resampleBuffer;

FILE *audioDump;

void Mix(u32 outAddr, u32 inAddr = 0, int leftVol = 0, int rightVol = 0);
Expand Down

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