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SasAudio: Straightforward implementation of linear interpolation.
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
parent
e9bea75a07
commit
61f80edda3
2 changed files with 20 additions and 52 deletions
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@ -337,7 +337,6 @@ SasInstance::SasInstance()
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sendBuffer(0),
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sendBufferDownsampled(0),
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sendBufferProcessed(0),
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resampleBuffer(0),
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grainSize(0) {
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#ifdef AUDIO_TO_FILE
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audioDump = fopen("D:\\audio.raw", "wb");
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@ -376,10 +375,8 @@ void SasInstance::ClearGrainSize() {
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delete[] sendBuffer;
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delete[] sendBufferDownsampled;
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delete[] sendBufferProcessed;
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delete[] resampleBuffer;
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mixBuffer = nullptr;
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sendBuffer = nullptr;
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resampleBuffer = nullptr;
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sendBufferDownsampled = nullptr;
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sendBufferProcessed = nullptr;
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}
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@ -392,7 +389,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
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delete[] sendBuffer;
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delete[] sendBufferDownsampled;
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delete[] sendBufferProcessed;
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delete[] resampleBuffer;
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mixBuffer = new s32[grainSize * 2];
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sendBuffer = new s32[grainSize * 2];
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@ -402,10 +398,6 @@ void SasInstance::SetGrainSize(int newGrainSize) {
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memset(sendBuffer, 0, sizeof(int) * grainSize * 2);
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memset(sendBufferDownsampled, 0, sizeof(s16) * grainSize);
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memset(sendBufferProcessed, 0, sizeof(s16) * grainSize * 2);
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// 2 samples padding at the start, that's where we copy the two last samples from the channel
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// so that we can do bicubic resampling if necessary. Plus 1 for smoothness hackery.
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resampleBuffer = new s16[grainSize * 4 + 3];
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}
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int SasInstance::EstimateMixUs() {
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@ -459,9 +451,7 @@ void SasVoice::ReadSamples(s16 *output, int numSamples) {
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atrac3.getNextSamples(output, numSamples);
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break;
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default:
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{
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memset(output, 0, numSamples * sizeof(s16));
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}
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memset(output, 0, numSamples * sizeof(s16));
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break;
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}
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}
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@ -493,47 +483,28 @@ void SasInstance::MixVoice(SasVoice &voice) {
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break;
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// else fallthrough! Don't change the check above.
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default:
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// Load resample history (so we can use a wide filter)
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resampleBuffer[0] = voice.resampleHist[0];
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resampleBuffer[1] = voice.resampleHist[1];
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// Figure out number of samples to read.
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// Actually this is not entirely correct - we need to get one extra sample, and store it
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// for the next time around. A little complicated...
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// But for now, see Smoothness HACKERY below :P
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u32 numSamples = ((u32)voice.sampleFrac + (u32)grainSize * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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if ((int)numSamples > grainSize * 4) {
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ERROR_LOG(SASMIX, "numSamples too large, clamping: %i vs %i", numSamples, grainSize * 4);
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numSamples = grainSize * 4;
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}
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// This feels a bit hacky. The first 32 samples after a keyon are 0s.
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const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
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int delay = 0;
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if (voice.envelope.NeedsKeyOn()) {
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int delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
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delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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// VAG seems to have an extra sample delay (not shared by PCM.)
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if (voice.type == VOICETYPE_VAG)
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++delay;
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voice.ReadSamples(resampleBuffer + 2 + delay, numSamples - delay);
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} else {
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voice.ReadSamples(resampleBuffer + 2, numSamples);
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}
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// Smoothness HACKERY
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resampleBuffer[2 + numSamples] = resampleBuffer[2 + numSamples - 1];
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// Save resample history
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voice.resampleHist[0] = resampleBuffer[2 + numSamples - 2];
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voice.resampleHist[1] = resampleBuffer[2 + numSamples - 1];
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// Resample to the correct pitch, writing exactly "grainSize" samples.
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// This is a HORRIBLE resampler by the way.
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// TODO: Special case no-resample case (and 2x and 0.5x) for speed, it's not uncommon
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u32 sampleFrac = voice.sampleFrac;
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for (int i = 0; i < grainSize; i++) {
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// For now: nearest neighbour, not even using the resample history at all.
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int sample = resampleBuffer[sampleFrac / PSP_SAS_PITCH_BASE + 2];
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for (int i = delay; i < grainSize; i++) {
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while (sampleFrac >= PSP_SAS_PITCH_BASE) {
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voice.resampleHist[0] = voice.resampleHist[1];
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voice.ReadSamples(&voice.resampleHist[1], 1);
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sampleFrac -= PSP_SAS_PITCH_BASE;
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}
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// Linear interpolation. Good enough. Need to make resampleHist bigger if we want more.
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int sample = (voice.resampleHist[0] * (PSP_SAS_PITCH_BASE - 1 - (int)sampleFrac) + voice.resampleHist[1] * (int)sampleFrac) >> PSP_SAS_PITCH_BASE_SHIFT;
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sampleFrac += voice.pitch;
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// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
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@ -549,21 +520,17 @@ void SasInstance::MixVoice(SasVoice &voice) {
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// We mix into this 32-bit temp buffer and clip in a second loop
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// Ideally, the shift right should be there too but for now I'm concerned about
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// not overflowing.
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mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> 12;
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mixBuffer[i * 2] += (sample * voice.volumeLeft) >> 12;
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mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> 12;
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sendBuffer[i * 2] += sample * voice.effectLeft >> 12;
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sendBuffer[i * 2 + 1] += sample * voice.effectRight >> 12;
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}
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voice.sampleFrac = sampleFrac;
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// Let's hope grainSize is a power of 2.
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//voice.sampleFrac &= grainSize * PSP_SAS_PITCH_BASE - 1;
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voice.sampleFrac -= numSamples * PSP_SAS_PITCH_BASE;
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if (voice.HaveSamplesEnded())
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voice.envelope.End();
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if (voice.envelope.HasEnded())
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{
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if (voice.envelope.HasEnded()) {
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// NOTICE_LOG(SCESAS, "Hit end of envelope");
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voice.playing = false;
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voice.on = false;
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@ -711,8 +678,11 @@ void SasInstance::DoState(PointerWrap &p) {
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if (sendBuffer != NULL && grainSize > 0) {
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p.DoArray(sendBuffer, grainSize * 2);
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}
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if (resampleBuffer != NULL && grainSize > 0) {
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p.DoArray(resampleBuffer, grainSize * 4 + 3);
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if (sendBuffer != NULL && grainSize > 0) {
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// Backwards compat
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int16_t *resampleBuf = new int16_t[grainSize * 4 + 3]();
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p.DoArray(resampleBuf, grainSize * 4 + 3);
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delete[] resampleBuf;
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}
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int n = PSP_SAS_VOICES_MAX;
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@ -289,8 +289,6 @@ public:
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s16 *sendBufferDownsampled;
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s16 *sendBufferProcessed;
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s16 *resampleBuffer;
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FILE *audioDump;
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void Mix(u32 outAddr, u32 inAddr = 0, int leftVol = 0, int rightVol = 0);
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