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SasAudio: Straightforward implementation of linear interpolation.

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Not sure about the performance impact, but calling ReadSamples with 1 as
an argument can't be terribly fast. So this might need some redesign
before merge, but should be a good starting point.
This commit is contained in:
Henrik Rydgard 2016-09-03 11:51:00 +02:00
parent e9bea75a07
commit 61f80edda3
2 changed files with 20 additions and 52 deletions
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70
Core/HW/SasAudio.cpp
View file

@ -337,7 +337,6 @@ SasInstance::SasInstance()
sendBuffer(0), sendBuffer(0),
sendBufferDownsampled(0), sendBufferDownsampled(0),
sendBufferProcessed(0), sendBufferProcessed(0),
resampleBuffer(0),
grainSize(0) { grainSize(0) {
#ifdef AUDIO_TO_FILE #ifdef AUDIO_TO_FILE
audioDump = fopen("D:\\audio.raw", "wb"); audioDump = fopen("D:\\audio.raw", "wb");
@ -376,10 +375,8 @@ void SasInstance::ClearGrainSize() {
delete[] sendBuffer; delete[] sendBuffer;
delete[] sendBufferDownsampled; delete[] sendBufferDownsampled;
delete[] sendBufferProcessed; delete[] sendBufferProcessed;
delete[] resampleBuffer;
mixBuffer = nullptr; mixBuffer = nullptr;
sendBuffer = nullptr; sendBuffer = nullptr;
resampleBuffer = nullptr;
sendBufferDownsampled = nullptr; sendBufferDownsampled = nullptr;
sendBufferProcessed = nullptr; sendBufferProcessed = nullptr;
} }
@ -392,7 +389,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
delete[] sendBuffer; delete[] sendBuffer;
delete[] sendBufferDownsampled; delete[] sendBufferDownsampled;
delete[] sendBufferProcessed; delete[] sendBufferProcessed;
delete[] resampleBuffer;
mixBuffer = new s32[grainSize * 2]; mixBuffer = new s32[grainSize * 2];
sendBuffer = new s32[grainSize * 2]; sendBuffer = new s32[grainSize * 2];
@ -402,10 +398,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
memset(sendBuffer, 0, sizeof(int) * grainSize * 2); memset(sendBuffer, 0, sizeof(int) * grainSize * 2);
memset(sendBufferDownsampled, 0, sizeof(s16) * grainSize); memset(sendBufferDownsampled, 0, sizeof(s16) * grainSize);
memset(sendBufferProcessed, 0, sizeof(s16) * grainSize * 2); 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() { int SasInstance::EstimateMixUs() {
@ -459,9 +451,7 @@ void SasVoice::ReadSamples(s16 *output, int numSamples) {
atrac3.getNextSamples(output, numSamples); atrac3.getNextSamples(output, numSamples);
break; break;
default: default:
{ memset(output, 0, numSamples * sizeof(s16));
memset(output, 0, numSamples * sizeof(s16));
}
break; break;
} }
} }
@ -493,47 +483,28 @@ void SasInstance::MixVoice(SasVoice &voice) {
break; break;
// else fallthrough! Don't change the check above. // else fallthrough! Don't change the check above.
default: 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. // 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()) { 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.) // VAG seems to have an extra sample delay (not shared by PCM.)
if (voice.type == VOICETYPE_VAG) if (voice.type == VOICETYPE_VAG)
++delay; ++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. // 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 // TODO: Special case no-resample case (and 2x and 0.5x) for speed, it's not uncommon
u32 sampleFrac = voice.sampleFrac; u32 sampleFrac = voice.sampleFrac;
for (int i = 0; i < grainSize; i++) { for (int i = delay; i < grainSize; i++) {
// For now: nearest neighbour, not even using the resample history at all. while (sampleFrac >= PSP_SAS_PITCH_BASE) {
int sample = resampleBuffer[sampleFrac / PSP_SAS_PITCH_BASE + 2]; voice.resampleHist[0] = voice.resampleHist[1];
voice.ReadSamples(&voice.resampleHist[1], 1);
sampleFrac -= PSP_SAS_PITCH_BASE;
}
// Linear interpolation. Good enough. Need to make resampleHist bigger if we want more.
int sample = (voice.resampleHist[0] * (PSP_SAS_PITCH_BASE - 1 - (int)sampleFrac) + voice.resampleHist[1] * (int)sampleFrac) >> PSP_SAS_PITCH_BASE_SHIFT;
sampleFrac += voice.pitch; sampleFrac += voice.pitch;
// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1. // The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
@ -549,21 +520,17 @@ void SasInstance::MixVoice(SasVoice &voice) {
// We mix into this 32-bit temp buffer and clip in a second loop // 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 // Ideally, the shift right should be there too but for now I'm concerned about
// not overflowing. // not overflowing.
mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> 12; mixBuffer[i * 2] += (sample * voice.volumeLeft) >> 12;
mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> 12; mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> 12;
sendBuffer[i * 2] += sample * voice.effectLeft >> 12; sendBuffer[i * 2] += sample * voice.effectLeft >> 12;
sendBuffer[i * 2 + 1] += sample * voice.effectRight >> 12; sendBuffer[i * 2 + 1] += sample * voice.effectRight >> 12;
} }
voice.sampleFrac = sampleFrac; 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;
if (voice.HaveSamplesEnded()) if (voice.HaveSamplesEnded())
voice.envelope.End(); voice.envelope.End();
if (voice.envelope.HasEnded()) if (voice.envelope.HasEnded()) {
{
// NOTICE_LOG(SCESAS, "Hit end of envelope"); // NOTICE_LOG(SCESAS, "Hit end of envelope");
voice.playing = false; voice.playing = false;
voice.on = false; voice.on = false;
@ -711,8 +678,11 @@ void SasInstance::DoState(PointerWrap &p) {
if (sendBuffer != NULL && grainSize > 0) { if (sendBuffer != NULL && grainSize > 0) {
p.DoArray(sendBuffer, grainSize * 2); p.DoArray(sendBuffer, grainSize * 2);
} }
if (resampleBuffer != NULL && grainSize > 0) { if (sendBuffer != NULL && grainSize > 0) {
p.DoArray(resampleBuffer, grainSize * 4 + 3); // 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; int n = PSP_SAS_VOICES_MAX;

2
Core/HW/SasAudio.h
View file

@ -289,8 +289,6 @@ public:
s16 *sendBufferDownsampled; s16 *sendBufferDownsampled;
s16 *sendBufferProcessed; s16 *sendBufferProcessed;
s16 *resampleBuffer;
FILE *audioDump; FILE *audioDump;
void Mix(u32 outAddr, u32 inAddr = 0, int leftVol = 0, int rightVol = 0); void Mix(u32 outAddr, u32 inAddr = 0, int leftVol = 0, int rightVol = 0);