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Diffstat (limited to 'engines/mads/nebular/sound_nebular.cpp')
-rw-r--r--engines/mads/nebular/sound_nebular.cpp150
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;
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-15 15:05:50 +0000