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https://github.com/libretro/scummvm.git
synced 2025-04-03 07:11:49 +00:00
SCUMM: Aggregate speaker states of CPU cycles in between samples
- formerly only two sample levels were possible: 32767 or -32767. Now the speaker states (0/1) between two samples are aggregated by the new SampleConverter class to allow more accurate sample values between -32767 and 32767. - the player's state is moved into a separate state struct
This commit is contained in:
Tobias Gunkel
parent
e56d5df295
commit
8b7ad559c1
@ -25,9 +25,6 @@
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#include "scumm/player_appleII.h"
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#include "scumm/scumm.h"
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// CPU_CLOCK according to AppleWin
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static const double CPU_CLOCK = 1020484.5; // ~ 1.02 MHz
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namespace Scumm {
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Player_AppleII::Player_AppleII(ScummEngine *scumm, Audio::Mixer *mixer) {
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@ -35,9 +32,10 @@ Player_AppleII::Player_AppleII(ScummEngine *scumm, Audio::Mixer *mixer) {
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_soundNr = 0;
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_mixer = mixer;
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_sampleRate = _mixer->getOutputRate();
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_vm = scumm;
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setSampleRate(_mixer->getOutputRate());
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_mixer->playStream(Audio::Mixer::kPlainSoundType, &_soundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true);
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}
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@ -73,60 +71,62 @@ void Player_AppleII::startSound(int nr) {
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Common::StackLock lock(_mutex);
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_soundNr = nr;
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_buffer.clear();
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_sampleConverter.reset();
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byte *data = _vm->getResourceAddress(rtSound, nr);
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assert(data);
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byte *ptr1 = data + 4;
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int type = ptr1[0];
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if (type == 0)
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_state.type = ptr1[0];
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if (_state.type == 0)
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return;
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int loop = ptr1[1];
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assert(loop > 0);
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ptr1 += 2;
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_state.loop = ptr1[1];
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assert(_state.loop > 0);
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_state.params = &ptr1[2];
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debug(4, "startSound %d: type %d, loop %d",
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nr, type, loop);
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nr, _state.type, _state.loop);
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do {
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switch (type) {
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switch (_state.type) {
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case 1: // freq up/down
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soundFunc1(ptr1);
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soundFunc1();
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break;
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case 2: // symmetric wave (~)
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soundFunc2(ptr1);
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soundFunc2();
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break;
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case 3: // asymmetric wave (__-)
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soundFunc3(ptr1);
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soundFunc3();
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break;
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case 4: // polyphone (2 voices)
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soundFunc4(ptr1);
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soundFunc4();
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break;
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case 5: // periodic noise
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soundFunc5(ptr1);
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soundFunc5();
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break;
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}
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--loop;
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} while (loop > 0);
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--_state.loop;
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} while (_state.loop > 0);
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}
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void Player_AppleII::stopAllSounds() {
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Common::StackLock lock(_mutex);
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_buffer.clear();
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_sampleConverter.reset();
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}
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void Player_AppleII::stopSound(int nr) {
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Common::StackLock lock(_mutex);
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if (_soundNr == nr) {
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_buffer.clear();
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_sampleConverter.reset();
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}
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}
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int Player_AppleII::getSoundStatus(int nr) const {
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Common::StackLock lock(_mutex);
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return (_buffer.availableSize() > 0 ? 1 : 0);
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return (_sampleConverter.availableSize() > 0 ? 1 : 0);
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}
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int Player_AppleII::getMusicTimer() {
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@ -136,7 +136,7 @@ int Player_AppleII::getMusicTimer() {
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int Player_AppleII::readBuffer(int16 *buffer, const int numSamples) {
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Common::StackLock lock(_mutex);
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return _buffer.read((byte*)buffer, numSamples * 2) / 2;
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return _sampleConverter.readSamples(buffer, numSamples);
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}
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/************************************
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@ -149,16 +149,7 @@ void Player_AppleII::speakerToggle() {
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}
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void Player_AppleII::generateSamples(int cycles) {
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// sampleDiff is used to compensate fractional samples
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static double sampleDiff = 0;
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double fSamples = (double)cycles / CPU_CLOCK * _sampleRate + sampleDiff;
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int samples = (int)(fSamples + 0.5);
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sampleDiff = fSamples - samples;
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float vol = (float)_maxvol / 255;
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int16 value = vol * (_speakerState ? 32767 : -32767);
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for (int i = 0; i < samples; ++i)
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_buffer.write(&value, sizeof(value));
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_sampleConverter.addCycles(_speakerState, cycles);
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}
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void Player_AppleII::wait(int interval, int count /*y*/) {
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@ -177,12 +168,12 @@ void Player_AppleII::_soundFunc1(int interval /*a*/, int count /*y*/) { // D076
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}
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}
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void Player_AppleII::soundFunc1(const byte *params) { // D085
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int delta = params[0];
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int count = params[1];
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byte interval = params[2]; // must be byte ("interval < delta" possible)
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int limit = params[3];
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bool decInterval = (params[4] >= 0x40);
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void Player_AppleII::soundFunc1() { // D085
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const int delta = _state.params[0];
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const int count = _state.params[1];
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byte interval = _state.params[2]; // must be byte ("interval < delta" possible)
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const int limit = _state.params[3];
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const bool decInterval = (_state.params[4] >= 0x40);
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if (decInterval) {
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do {
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@ -215,12 +206,12 @@ void Player_AppleII::_soundFunc2(int interval /*a*/, int count) { // D0EF
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}
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}
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void Player_AppleII::soundFunc2(const byte *params) { // D0D6
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void Player_AppleII::soundFunc2() { // D0D6
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for (int pos = 1; pos < 256; ++pos) {
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byte interval = params[pos];
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byte interval = _state.params[pos];
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if (interval == 0xFF)
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return;
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_soundFunc2(interval, params[0] /*, LD12F=interval*/);
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_soundFunc2(interval, _state.params[0] /*, LD12F=interval*/);
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}
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}
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@ -238,12 +229,12 @@ void Player_AppleII::_soundFunc3(int interval /*a*/, int count /*LD12D*/) { // D
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}
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}
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void Player_AppleII::soundFunc3(const byte *params) { // D132
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void Player_AppleII::soundFunc3() { // D132
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for (int pos = 1; pos < 256; ++pos) {
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byte interval = params[pos];
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byte interval = _state.params[pos];
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if (interval == 0xFF)
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return;
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_soundFunc3(interval, params[0]);
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_soundFunc3(interval, _state.params[0]);
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}
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}
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@ -290,7 +281,7 @@ void Player_AppleII::_soundFunc4(byte param0, byte param1, byte param2) { // D1A
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if (speakerShiftReg & 0x1)
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speakerToggle();
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speakerShiftReg >>= 1;
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generateSamples(40);
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generateSamples(42); /* actually 42.5 */
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++count;
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if (count == 0) {
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@ -299,7 +290,8 @@ void Player_AppleII::_soundFunc4(byte param0, byte param1, byte param2) { // D1A
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}
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}
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void Player_AppleII::soundFunc4(const byte *params) { // D170
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void Player_AppleII::soundFunc4() { // D170
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const byte *params = _state.params;
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while (params[0] != 0x01) {
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_soundFunc4(params[0], params[1], params[2]);
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params += 3;
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@ -347,12 +339,12 @@ byte /*a*/ Player_AppleII::noise() { // D261
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return result;
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}
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void Player_AppleII::soundFunc5(const byte *params) { // D222
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void Player_AppleII::soundFunc5() { // D222
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const byte noiseMask[] = {
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0x3F, 0x3F, 0x7F, 0x7F, 0x7F, 0x7F, 0xFF, 0xFF, 0xFF, 0x0F, 0x0F
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};
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int param0 = params[0];
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int param0 = _state.params[0];
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assert(param0 > 0);
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for (int i = 0; i < 10; ++i) {
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int count = param0;
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@ -112,12 +112,97 @@ private:
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uint32 _pos;
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};
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// CPU_CLOCK according to AppleWin
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static const double CPU_CLOCK = 1020484.5; // ~ 1.02 MHz
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class SampleConverter {
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private:
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void newSample(int sample) {
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int16 value = _volume * sample / 255;
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_buffer.write(&value, sizeof(value));
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}
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public:
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SampleConverter() :
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_cyclesPerSample(0),
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_missingCycles(0),
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_sampleCyclesSum(0),
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_volume(255)
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{}
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void reset() {
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_missingCycles = 0;
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_sampleCyclesSum = 0;
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_buffer.clear();
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}
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uint32 availableSize() const {
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return _buffer.availableSize();
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}
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void setMusicVolume(int vol) {
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assert(vol >= 0 && vol <= 255);
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_volume = vol;
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}
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void setSampleRate(int rate) {
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_cyclesPerSample = CPU_CLOCK / (float)rate;
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reset();
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}
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void addCycles(byte level, int cycles) {
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// step 1: if cycles are left from the last call, process them first
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if (_missingCycles > 0) {
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int n = (_missingCycles < cycles) ? _missingCycles : cycles;
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if (level)
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_sampleCyclesSum += n;
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cycles -= n;
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_missingCycles -= n;
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if (_missingCycles == 0) {
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newSample(2*32767 * _sampleCyclesSum / _cyclesPerSample - 32767);
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} else {
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return;
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}
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}
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_sampleCyclesSum = 0;
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// step 2: process blocks of cycles fitting into a whole sample
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while (cycles >= _cyclesPerSample) {
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newSample(level ? 32767 : -32767);
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cycles -= _cyclesPerSample;
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}
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// step 3: remember cycles left for next call
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if (cycles > 0) {
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_missingCycles = _cyclesPerSample - cycles;
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if (level)
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_sampleCyclesSum = cycles;
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}
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}
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uint32 readSamples(void *buffer, int numSamples) {
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return _buffer.read((byte*)buffer, numSamples * 2) / 2;
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}
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private:
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float _cyclesPerSample;
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int _missingCycles;
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int _sampleCyclesSum;
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int _volume; /* 0 - 255 */
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DynamicMemoryStream _buffer;
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};
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class Player_AppleII : public Audio::AudioStream, public MusicEngine {
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public:
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Player_AppleII(ScummEngine *scumm, Audio::Mixer *mixer);
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virtual ~Player_AppleII();
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virtual void setMusicVolume(int vol) { _maxvol = vol; }
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virtual void setMusicVolume(int vol) { _sampleConverter.setMusicVolume(vol); }
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void setSampleRate(int rate) {
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_sampleRate = rate;
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_sampleConverter.setSampleRate(rate);
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}
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void startMusic(int songResIndex);
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virtual void startSound(int sound);
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virtual void stopSound(int sound);
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@ -132,17 +217,22 @@ public:
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int getRate() const { return _sampleRate; }
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private:
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struct state_t {
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int type;
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int loop;
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const byte *params;
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} _state;
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ScummEngine *_vm;
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Audio::Mixer *_mixer;
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Audio::SoundHandle _soundHandle;
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int _maxvol;
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int _sampleRate;
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Common::Mutex _mutex;
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private:
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byte _speakerState;
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DynamicMemoryStream _buffer;
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int _soundNr;
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SampleConverter _sampleConverter;
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private:
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void speakerToggle();
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@ -150,15 +240,15 @@ private:
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void wait(int interval, int count);
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byte noise();
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void soundFunc1(const byte *params);
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void soundFunc1();
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void _soundFunc1(int interval, int count);
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void soundFunc2(const byte *params);
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void soundFunc2();
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void _soundFunc2(int interval, int count);
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void soundFunc3(const byte *params);
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void soundFunc3();
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void _soundFunc3(int interval, int count);
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void soundFunc4(const byte *params);
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void soundFunc4();
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void _soundFunc4(byte param0, byte param1, byte param2);
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void soundFunc5(const byte *params);
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void soundFunc5();
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void _soundFunc5(int interval, int count);
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};
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