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Merge pull request #5509 from unknownbrackets/sas-minor
Improve pitch handling in sas
This commit is contained in:
Henrik Rydgård
commit
106656f51e
@ -45,6 +45,7 @@ enum {
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ERROR_SAS_BAD_ADDRESS = 0x80420005,
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ERROR_SAS_INVALID_VOICE = 0x80420010,
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ERROR_SAS_INVALID_NOISE_FREQ = 0x80420011,
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ERROR_SAS_INVALID_PITCH = 0x80420012,
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ERROR_SAS_INVALID_ADSR_CURVE_MODE = 0x80420013,
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ERROR_SAS_INVALID_PARAMETER = 0x80420014,
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ERROR_SAS_INVALID_LOOP_POS = 0x80420015,
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@ -263,23 +264,17 @@ u32 sceSasSetVolume(u32 core, int voiceNum, int leftVol, int rightVol, int effec
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}
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u32 sceSasSetPitch(u32 core, int voiceNum, int pitch) {
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DEBUG_LOG(SCESAS, "sceSasSetPitch(%08x, %i, %i)", core, voiceNum, pitch);
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if (voiceNum >= PSP_SAS_VOICES_MAX || voiceNum < 0) {
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WARN_LOG(SCESAS, "%s: invalid voicenum %d", __FUNCTION__, voiceNum);
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return ERROR_SAS_INVALID_VOICE;
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}
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SasVoice &v = sas->voices[voiceNum];
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// Clamp pitch
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if (pitch < PSP_SAS_PITCH_MIN) {
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WARN_LOG(SCESAS, "sceSasSetPitch: bad pitch %i, clamping to %i", pitch, PSP_SAS_PITCH_MIN);
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pitch = PSP_SAS_PITCH_MIN;
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} else if (pitch > PSP_SAS_PITCH_MAX) {
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WARN_LOG(SCESAS, "sceSasSetPitch: bad pitch %i, clamping to %i", pitch, PSP_SAS_PITCH_MAX);
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pitch = PSP_SAS_PITCH_MAX;
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if (pitch < PSP_SAS_PITCH_MIN || pitch > PSP_SAS_PITCH_MAX) {
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WARN_LOG(SCESAS, "sceSasSetPitch(%08x, %i, %i): bad pitch", core, voiceNum, pitch);
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return ERROR_SAS_INVALID_PITCH;
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}
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DEBUG_LOG(SCESAS, "sceSasSetPitch(%08x, %i, %i)", core, voiceNum, pitch);
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SasVoice &v = sas->voices[voiceNum];
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v.pitch = pitch;
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v.ChangedParams(false);
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return 0;
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@ -368,16 +368,7 @@ void SasVoice::ReadSamples(s16 *output, int numSamples) {
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// Read N samples into the resample buffer. Could do either PCM or VAG here.
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switch (type) {
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case VOICETYPE_VAG:
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{
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vag.GetSamples(output, numSamples);
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if (vag.End()) {
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// NOTICE_LOG(SCESAS, "Hit end of VAG audio");
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playing = false;
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on = false; // ??
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// TODO: Should this remain on somehow or just hit rock bottom immediately?
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envelope.End();
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}
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}
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vag.GetSamples(output, numSamples);
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break;
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case VOICETYPE_PCM:
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{
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@ -428,8 +419,7 @@ void SasVoice::ReadSamples(s16 *output, int numSamples) {
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bool SasVoice::HaveSamplesEnded() {
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switch (type) {
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case VOICETYPE_VAG:
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// TODO: Is it here, or before the samples are processed?
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return false;
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return vag.End();
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case VOICETYPE_PCM:
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return pcmIndex >= pcmSize;
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@ -462,15 +452,17 @@ void SasInstance::MixVoice(SasVoice &voice) {
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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 = (voice.sampleFrac + grainSize * voice.pitch) / PSP_SAS_PITCH_BASE;
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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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if (voice.envelope.NeedsKeyOn()) {
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voice.ReadSamples(resampleBuffer + 2 + 32, numSamples - 32);
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int delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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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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@ -482,42 +474,14 @@ void SasInstance::MixVoice(SasVoice &voice) {
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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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// We need to shift by 12 anyway, so combine that with the volume shift.
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int volumeShift = (12 + MAX_CONFIG_VOLUME - g_Config.iSFXVolume);
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if (volumeShift < 0) volumeShift = 0;
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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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sampleFrac += voice.pitch;
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// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
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// Reduce it to 14 bits, by shifting off 15. Round up by adding (1 << 14) first.
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int envelopeValue = voice.envelope.GetHeight();
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envelopeValue = (envelopeValue + (1 << 14)) >> 15;
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// We just scale by the envelope before we scale by volumes.
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// Again, we round up by adding (1 << 14) first (*after* multiplying.)
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sample = ((sample * envelopeValue) + (1 << 14)) >> 15;
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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 ) >> volumeShift; // Max = 16 and Min = 12(default)
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mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> volumeShift; // Max = 16 and Min = 12(default)
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sendBuffer[i * 2] += sample * voice.volumeLeftSend >> 12;
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sendBuffer[i * 2 + 1] += sample * voice.volumeRightSend >> 12;
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voice.envelope.Step();
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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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// Let's try to optimize the easy case where we don't need to resample at all.
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if (voice.sampleFrac == 0 && (voice.pitch == PSP_SAS_PITCH_BASE || ignorePitch))
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MixSamplesOptimal(voice);
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// Half pitch is also quite common.
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else if (voice.pitch == PSP_SAS_PITCH_BASE / 2)
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MixSamplesHalfPitch(voice);
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else
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MixSamples(voice);
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if (voice.HaveSamplesEnded())
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voice.envelope.End();
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@ -530,6 +494,83 @@ void SasInstance::MixVoice(SasVoice &voice) {
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}
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}
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void SasInstance::MixSamples(SasVoice &voice) {
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// Resample to the correct pitch, writing exactly "grainSize" samples.
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// This is a poor resampler by the way.
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u32 sampleFrac = voice.sampleFrac;
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// We need to shift by 12 anyway, so combine that with the volume shift.
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u8 volumeShift = 12;
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if (g_Config.iSFXVolume >= 0 && g_Config.iSFXVolume < MAX_CONFIG_VOLUME)
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volumeShift += MAX_CONFIG_VOLUME - g_Config.iSFXVolume;
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const int offset = sampleFrac == 0 ? 2 : 1;
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for (int i = 0; i < grainSize; i++) {
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const int readIndex = sampleFrac >> PSP_SAS_PITCH_BASE_SHIFT;
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const int readFrac = sampleFrac & (PSP_SAS_PITCH_BASE - 1);
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int sample1 = resampleBuffer[readIndex + offset];
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int sample2 = resampleBuffer[readIndex + 1 + offset];
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int sample = (sample1 * (PSP_SAS_PITCH_BASE - readFrac) + sample2 * readFrac) / PSP_SAS_PITCH_BASE;
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sampleFrac += voice.pitch;
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MixSample(voice, i, sample, volumeShift);
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}
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voice.sampleFrac = sampleFrac & (PSP_SAS_PITCH_BASE - 1);
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}
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void SasInstance::MixSamplesHalfPitch(SasVoice &voice) {
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// We need to shift by 12 anyway, so combine that with the volume shift.
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u8 volumeShift = 12;
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if (g_Config.iSFXVolume >= 0 && g_Config.iSFXVolume < MAX_CONFIG_VOLUME)
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volumeShift += MAX_CONFIG_VOLUME - g_Config.iSFXVolume;
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int readIndex2 = voice.sampleFrac == 0 ? 0 : -1;
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for (int i = 0; i < grainSize; i++) {
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int sample1 = resampleBuffer[(readIndex2 >> 1) + 2];
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int sample2 = resampleBuffer[(readIndex2 >> 1) + 1 + 2];
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int sample = readIndex2 & 1 ? ((sample1 + sample2) >> 1) : sample1;
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++readIndex2;
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MixSample(voice, i, sample, volumeShift);
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}
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voice.sampleFrac = readIndex2 & 1 ? PSP_SAS_PITCH_BASE / 2 : 0;
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}
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void SasInstance::MixSamplesOptimal(SasVoice &voice) {
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// We need to shift by 12 anyway, so combine that with the volume shift.
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u8 volumeShift = 12;
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if (g_Config.iSFXVolume >= 0 && g_Config.iSFXVolume < MAX_CONFIG_VOLUME)
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volumeShift += MAX_CONFIG_VOLUME - g_Config.iSFXVolume;
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int readIndex = 2;
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for (int i = 0; i < grainSize; i++) {
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int sample = resampleBuffer[readIndex++];
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MixSample(voice, i, sample, volumeShift);
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}
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}
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inline void SasInstance::MixSample(SasVoice &voice, int i, int sample, u8 volumeShift) {
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// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
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// Reduce it to 14 bits, by shifting off 15. Round up by adding (1 << 14) first.
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int envelopeValue = voice.envelope.GetHeight();
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envelopeValue = (envelopeValue + (1 << 14)) >> 15;
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// We just scale by the envelope before we scale by volumes.
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// Again, we round up by adding (1 << 14) first (*after* multiplying.)
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sample = ((sample * envelopeValue) + (1 << 14)) >> 15;
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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 ) >> volumeShift; // Max = 16 and Min = 12(default)
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mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> volumeShift; // Max = 16 and Min = 12(default)
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sendBuffer[i * 2] += sample * voice.volumeLeftSend >> 12;
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sendBuffer[i * 2 + 1] += sample * voice.volumeRightSend >> 12;
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voice.envelope.Step();
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}
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void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
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int voicesPlayingCount = 0;
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@ -648,6 +689,7 @@ void SasVoice::KeyOn() {
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playing = true;
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on = true;
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paused = false;
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sampleFrac = 0;
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}
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void SasVoice::KeyOff() {
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@ -29,8 +29,9 @@
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enum {
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PSP_SAS_VOICES_MAX = 32,
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PSP_SAS_PITCH_MIN = 1,
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PSP_SAS_PITCH_MIN = 0x0000,
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PSP_SAS_PITCH_BASE = 0x1000,
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PSP_SAS_PITCH_BASE_SHIFT = 12,
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PSP_SAS_PITCH_MAX = 0x4000,
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PSP_SAS_VOL_MAX = 0x1000,
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@ -291,5 +292,10 @@ public:
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WaveformEffect waveformEffect;
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private:
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void MixSamples(SasVoice &voice);
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void MixSamplesHalfPitch(SasVoice &voice);
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void MixSamplesOptimal(SasVoice &voice);
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void MixSample(SasVoice &voice, int i, int sample, u8 volumeShift);
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int grainSize;
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};
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