diff options
Diffstat (limited to 'engines/mads/nebular/sound_nebular.cpp')
-rw-r--r-- | engines/mads/nebular/sound_nebular.cpp | 150 |
1 files changed, 69 insertions, 81 deletions
diff --git a/engines/mads/nebular/sound_nebular.cpp b/engines/mads/nebular/sound_nebular.cpp index 0a054440b2..711f82a05b 100644 --- a/engines/mads/nebular/sound_nebular.cpp +++ b/engines/mads/nebular/sound_nebular.cpp @@ -8,12 +8,12 @@ * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. - + * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. - + * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. @@ -21,6 +21,7 @@ */ #include "audio/audiostream.h" +#include "audio/fmopl.h" #include "audio/decoders/raw.h" #include "common/algorithm.h" #include "common/debug.h" @@ -36,6 +37,7 @@ namespace Nebular { bool AdlibChannel::_channelsEnabled; AdlibChannel::AdlibChannel() { + _owner = nullptr; _activeCount = 0; _field1 = 0; _field2 = 0; @@ -43,6 +45,7 @@ AdlibChannel::AdlibChannel() { _field4 = 0; _sampleIndex = 0; _volume = 0; + _volumeOffset = 0; _field7 = 0; _field8 = 0; _field9 = 0; @@ -55,11 +58,11 @@ AdlibChannel::AdlibChannel() { _pSrc = nullptr; _ptr3 = nullptr; _ptr4 = nullptr; + _ptrEnd = nullptr; _field17 = 0; _field19 = 0; _soundData = nullptr; _field1D = 0; - _field1E = 0; _field1F = 0; _field20 = 0; @@ -95,6 +98,7 @@ void AdlibChannel::setPtr2(byte *pData) { void AdlibChannel::load(byte *pData) { _ptr1 = _pSrc = _ptr3 = pData; _ptr4 = _soundData = pData; + _volumeOffset = 0; _fieldA = 0xFF; _activeCount = 1; _fieldD = 64; @@ -102,17 +106,20 @@ void AdlibChannel::load(byte *pData) { _field1F = 0; _field2 = _field3 = 0; _volume = _field7 = 0; - _field1D = _field1E = 0; + _field1D = 0; _fieldE = 0; _field9 = 0; _fieldB = 0; _field17 = 0; _field19 = 0; + + CachedDataEntry &cacheEntry = _owner->getCachedData(pData); + _ptrEnd = cacheEntry._dataEnd; } void AdlibChannel::check(byte *nullPtr) { if (_activeCount && _fieldE) { - if (!_field1E) { + if (!_volumeOffset) { _pSrc = nullPtr; _fieldE = 0; } else { @@ -150,7 +157,7 @@ AdlibSample::AdlibSample(Common::SeekableReadStream &s) { /*-----------------------------------------------------------------------*/ -ASound::ASound(Audio::Mixer *mixer, FM_OPL *opl, const Common::String &filename, int dataOffset) { +ASound::ASound(Audio::Mixer *mixer, OPL::OPL *opl, const Common::String &filename, int dataOffset) { // Open up the appropriate sound file if (!_soundFile.open(filename)) error("Could not open file - %s", filename.c_str()); @@ -161,6 +168,7 @@ ASound::ASound(Audio::Mixer *mixer, FM_OPL *opl, const Common::String &filename, _samplePtr = nullptr; _frameCounter = 0; _isDisabled = false; + _masterVolume = 255; _v1 = 0; _v2 = 0; _activeChannelNumber = 0; @@ -181,17 +189,15 @@ ASound::ASound(Audio::Mixer *mixer, FM_OPL *opl, const Common::String &filename, _randomSeed = 1234; _amDep = _vibDep = _splitPoint = true; - _samplesTillCallback = 0; - _samplesTillCallbackRemainder = 0; - _samplesPerCallback = getRate() / CALLBACKS_PER_SECOND; - _samplesPerCallbackRemainder = getRate() % CALLBACKS_PER_SECOND; - for (int i = 0; i < 11; ++i) { _channelData[i]._field0 = 0; _channelData[i]._freqMask = 0; _channelData[i]._freqBase = 0; _channelData[i]._field6 = 0; } + + for (int i = 0; i < ADLIB_CHANNEL_COUNT; ++i) + _channels[i]._owner = this; AdlibChannel::_channelsEnabled = false; @@ -200,23 +206,19 @@ ASound::ASound(Audio::Mixer *mixer, FM_OPL *opl, const Common::String &filename, _mixer = mixer; _opl = opl; - _opl->init(getRate()); - _mixer->playStream(Audio::Mixer::kPlainSoundType, &_soundHandle, this, -1, - Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true); - // Initialize the Adlib adlibInit(); // Reset the adlib command0(); + + _opl->start(new Common::Functor0Mem<void, ASound>(this, &ASound::onTimer), CALLBACKS_PER_SECOND); } ASound::~ASound() { Common::List<CachedDataEntry>::iterator i; for (i = _dataCache.begin(); i != _dataCache.end(); ++i) delete[] (*i)._data; - - _mixer->stopHandle(_soundHandle); } void ASound::validate() { @@ -283,6 +285,17 @@ void ASound::noise() { } } +CachedDataEntry &ASound::getCachedData(byte *pData) { + Common::List<CachedDataEntry>::iterator i; + for (i = _dataCache.begin(); i != _dataCache.end(); ++i) { + CachedDataEntry &e = *i; + if (e._data == pData) + return e; + } + + error("Could not find previously loaded data"); +} + void ASound::write(int reg, int val) { _queue.push(RegisterValue(reg, val)); } @@ -331,6 +344,7 @@ byte *ASound::loadData(int offset, int size) { CachedDataEntry rec; rec._offset = offset; rec._data = new byte[size]; + rec._dataEnd = rec._data + size - 1; _soundFile.seek(_dataOffset + offset); _soundFile.read(rec._data, size); _dataCache.push_back(rec); @@ -449,6 +463,10 @@ void ASound::pollActiveChannel() { warning("pollActiveChannel(): No data found for sound channel"); break; } + if (pSrc > chan->_ptrEnd) { + warning("Read beyond end of loaded sound data"); + } + if (!(*pSrc & 0x80) || (*pSrc <= 0xF0)) { if (updateFlag) updateActiveChannel(); @@ -516,7 +534,7 @@ void ASound::pollActiveChannel() { chan->_field1 = 0; chan->_field2 = chan->_field3 = 0; chan->_volume = chan->_field7 = 0; - chan->_field1D = chan->_field1E = 0; + chan->_field1D = chan->_volumeOffset = 0; chan->_field8 = 0; chan->_field9 = 0; chan->_fieldB = 0; @@ -570,7 +588,7 @@ void ASound::pollActiveChannel() { break; case 8: - chan->_field1D = *++pSrc; + chan->_field1D = (int8)*++pSrc; chan->_pSrc += 2; break; @@ -591,7 +609,7 @@ void ASound::pollActiveChannel() { if (chan->_fieldE) { chan->_pSrc += 2; } else { - chan->_field1E = *pSrc >> 1; + chan->_volumeOffset = *pSrc >> 1; updateFlag = true; chan->_pSrc += 2; } @@ -635,7 +653,7 @@ void ASound::pollActiveChannel() { if (!--chan->_field9) { chan->_field9 = chan->_fieldA; if (chan->_field2) { - int8 newVal = (int8)chan->_field2 + (int8)chan->_field1E; + int8 newVal = (int8)chan->_field2 + (int8)chan->_volumeOffset; if (newVal < 0) { chan->_field9 = 0; newVal = 0; @@ -644,7 +662,7 @@ void ASound::pollActiveChannel() { newVal = 63; } - chan->_field1E = newVal; + chan->_volumeOffset = newVal; updateFlag = true; } } @@ -709,8 +727,8 @@ void ASound::updateChannelState() { resultCheck(); } else { int reg = 0xA0 + _activeChannelNumber; - int vTimes = (_activeChannelPtr->_field4 + _activeChannelPtr->_field1F) / 12; - int vOffset = (_activeChannelPtr->_field4 + _activeChannelPtr->_field1F) % 12; + int vTimes = (byte)(_activeChannelPtr->_field4 + _activeChannelPtr->_field1F) / 12; + int vOffset = (byte)(_activeChannelPtr->_field4 + _activeChannelPtr->_field1F) % 12; int val = _vList1[vOffset] + _activeChannelPtr->_field1D; write2(8, reg, val & 0xFF); @@ -727,32 +745,18 @@ static const int outputIndexes[] = { static const int outputChannels[] = { 0, 1, 2, 3, 4, 5, 8, 9, 10, 11, 12, 13, 16, 17, 18, 19, 20, 21, 0 }; -static const int volumeList[] = { - 0x3F, 0x3F, 0x36, 0x31, 0x2D, 0x2A, 0x28, 0x26, 0x24, 0x22, 0x21, 0x20, 0x1F, 0x1E, 0x1D, 0x1C, - 0x1B, 0x1A, 0x19, 0x19, 0x18, 0x17, 0x17, 0x16, 0x16, 0x15, 0x15, 0x14, 0x14, 0x13, 0x12, 0x12, - 0x11, 0x11, 0x10, 0x10, 0x0F, 0x0F, 0x0E, 0x0E, 0x0D, 0x0D, 0x0C, 0x0C, 0x0B, 0x0B, 0x0A, 0x0A, - 0x0A, 0x09, 0x09, 0x09, 0x09, 0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x07, 0x07, 0x07, 0x07, 0x07, - 0x06, 0x06, 0x06, 0x06, 0x06, 0x06, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x04, - 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x03, 0x03, 0x03, 0x03, 0x03, 0x03, - 0x03, 0x03, 0x03, 0x03, 0x03, 0x03, 0x02, 0x02, 0x02, 0x02, 0x02, 0x02, 0x02, 0x02, 0x02, 0x01, - 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, 0x01, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, -}; void ASound::updateActiveChannel() { int reg = 0x40 + outputChannels[outputIndexes[_activeChannelNumber * 2 + 1]]; int portVal = _ports[reg] & 0xFFC0; - int newVolume = CLIP(_activeChannelPtr->_volume + _activeChannelPtr->_field1E, 0, 63); + int newVolume = CLIP(_activeChannelPtr->_volume + _activeChannelPtr->_volumeOffset, 0, 63); + newVolume = newVolume * _masterVolume / 255; // Note: Original had a whole block not seeming to be used, since the initialisation // sets a variable to 5660h, and doesn't change it, so the branch is never taken - int val = CLIP(newVolume - volumeList[_activeChannelPtr->_fieldD], 0, 63); - val = (63 - val) | portVal; + portVal |= 63 - newVolume; - int val2 = CLIP(newVolume - volumeList[-(_activeChannelPtr->_fieldD - 127)], 0, 63); - val2 = (63 - val2) | portVal; - write2(0, reg, val); - write2(2, reg, val2); + write2(8, reg, portVal); } void ASound::loadSample(int sampleIndex) { @@ -820,32 +824,16 @@ void ASound::updateFNumber() { write2(8, hiReg, val2); } -int ASound::readBuffer(int16 *buffer, const int numSamples) { +void ASound::onTimer() { Common::StackLock slock(_driverMutex); + poll(); + flush(); +} - int32 samplesLeft = numSamples; - memset(buffer, 0, sizeof(int16) * numSamples); - while (samplesLeft) { - if (!_samplesTillCallback) { - poll(); - flush(); - - _samplesTillCallback = _samplesPerCallback; - _samplesTillCallbackRemainder += _samplesPerCallbackRemainder; - if (_samplesTillCallbackRemainder >= CALLBACKS_PER_SECOND) { - _samplesTillCallback++; - _samplesTillCallbackRemainder -= CALLBACKS_PER_SECOND; - } - } - - int32 render = MIN<int>(samplesLeft, _samplesTillCallback); - samplesLeft -= render; - _samplesTillCallback -= render; - - _opl->readBuffer(buffer, render); - buffer += render; - } - return numSamples; +void ASound::setVolume(int volume) { + _masterVolume = volume; + if (!volume) + command0(); } int ASound::command0() { @@ -966,7 +954,7 @@ const ASound1::CommandPtr ASound1::_commandList[42] = { &ASound1::command40, &ASound1::command41 }; -ASound1::ASound1(Audio::Mixer *mixer, FM_OPL *opl) +ASound1::ASound1(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.001", 0x1520) { _cmd23Toggle = false; @@ -1005,22 +993,22 @@ int ASound1::command10() { int ASound1::command11() { command111213(); - _channels[0]._field1E = 0; - _channels[1]._field1E = 0; + _channels[0]._volumeOffset = 0; + _channels[1]._volumeOffset = 0; return 0; } int ASound1::command12() { command111213(); - _channels[0]._field1E = 40; - _channels[1]._field1E = 0; + _channels[0]._volumeOffset = 40; + _channels[1]._volumeOffset = 0; return 0; } int ASound1::command13() { command111213(); - _channels[0]._field1E = 40; - _channels[1]._field1E = 50; + _channels[0]._volumeOffset = 40; + _channels[1]._volumeOffset = 50; return 0; } @@ -1267,7 +1255,7 @@ const ASound2::CommandPtr ASound2::_commandList[44] = { &ASound2::command40, &ASound2::command41, &ASound2::command42, &ASound2::command43 }; -ASound2::ASound2(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.002", 0x15E0) { +ASound2::ASound2(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.002", 0x15E0) { _command12Param = 0xFD; // Load sound samples @@ -1638,7 +1626,7 @@ const ASound3::CommandPtr ASound3::_commandList[61] = { &ASound3::command60 }; -ASound3::ASound3(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.003", 0x15B0) { +ASound3::ASound3(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.003", 0x15B0) { _command39Flag = false; // Load sound samples @@ -2042,7 +2030,7 @@ const ASound4::CommandPtr ASound4::_commandList[61] = { &ASound4::command60 }; -ASound4::ASound4(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.004", 0x14F0) { +ASound4::ASound4(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.004", 0x14F0) { // Load sound samples _soundFile.seek(_dataOffset + 0x122); for (int i = 0; i < 210; ++i) @@ -2298,7 +2286,7 @@ const ASound5::CommandPtr ASound5::_commandList[42] = { &ASound5::command40, &ASound5::command41 }; -ASound5::ASound5(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.002", 0x15E0) { +ASound5::ASound5(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.002", 0x15E0) { // Load sound samples _soundFile.seek(_dataOffset + 0x144); for (int i = 0; i < 164; ++i) @@ -2539,7 +2527,7 @@ const ASound6::CommandPtr ASound6::_commandList[30] = { &ASound6::nullCommand, &ASound6::command29 }; -ASound6::ASound6(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.006", 0x1390) { +ASound6::ASound6(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.006", 0x1390) { // Load sound samples _soundFile.seek(_dataOffset + 0x122); for (int i = 0; i < 200; ++i) @@ -2695,7 +2683,7 @@ const ASound7::CommandPtr ASound7::_commandList[38] = { &ASound7::command36, &ASound7::command37 }; -ASound7::ASound7(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.007", 0x1460) { +ASound7::ASound7(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.007", 0x1460) { // Load sound samples _soundFile.seek(_dataOffset + 0x122); for (int i = 0; i < 214; ++i) @@ -2901,7 +2889,7 @@ const ASound8::CommandPtr ASound8::_commandList[38] = { &ASound8::command36, &ASound8::command37 }; -ASound8::ASound8(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.008", 0x1490) { +ASound8::ASound8(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.008", 0x1490) { // Load sound samples _soundFile.seek(_dataOffset + 0x122); for (int i = 0; i < 174; ++i) @@ -3157,7 +3145,7 @@ const ASound9::CommandPtr ASound9::_commandList[52] = { &ASound9::command48, &ASound9::command49, &ASound9::command50, &ASound9::command51 }; -ASound9::ASound9(Audio::Mixer *mixer, FM_OPL *opl) : ASound(mixer, opl, "asound.009", 0x16F0) { +ASound9::ASound9(Audio::Mixer *mixer, OPL::OPL *opl) : ASound(mixer, opl, "asound.009", 0x16F0) { _v1 = _v2 = 0; _soundPtr = nullptr; |