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Diffstat (limited to 'engines/mads/nebular/sound_nebular.cpp')
-rw-r--r--engines/mads/nebular/sound_nebular.cpp117
1 files changed, 81 insertions, 36 deletions
diff --git a/engines/mads/nebular/sound_nebular.cpp b/engines/mads/nebular/sound_nebular.cpp
index c540eb4382..240c18f6dc 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.
@@ -24,6 +24,7 @@
#include "audio/decoders/raw.h"
#include "common/algorithm.h"
#include "common/debug.h"
+#include "common/md5.h"
#include "common/memstream.h"
#include "mads/sound.h"
#include "mads/nebular/sound_nebular.h"
@@ -35,6 +36,7 @@ namespace Nebular {
bool AdlibChannel::_channelsEnabled;
AdlibChannel::AdlibChannel() {
+ _owner = nullptr;
_activeCount = 0;
_field1 = 0;
_field2 = 0;
@@ -42,6 +44,7 @@ AdlibChannel::AdlibChannel() {
_field4 = 0;
_sampleIndex = 0;
_volume = 0;
+ _volumeOffset = 0;
_field7 = 0;
_field8 = 0;
_field9 = 0;
@@ -54,11 +57,11 @@ AdlibChannel::AdlibChannel() {
_pSrc = nullptr;
_ptr3 = nullptr;
_ptr4 = nullptr;
+ _ptrEnd = nullptr;
_field17 = 0;
_field19 = 0;
_soundData = nullptr;
_field1D = 0;
- _field1E = 0;
_field1F = 0;
_field20 = 0;
@@ -94,6 +97,7 @@ 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;
@@ -101,17 +105,20 @@ 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 {
@@ -160,6 +167,7 @@ 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;
@@ -191,6 +199,9 @@ ASound::ASound(Audio::Mixer *mixer, FM_OPL *opl, const Common::String &filename,
_channelData[i]._freqBase = 0;
_channelData[i]._field6 = 0;
}
+
+ for (int i = 0; i < ADLIB_CHANNEL_COUNT; ++i)
+ _channels[i]._owner = this;
AdlibChannel::_channelsEnabled = false;
@@ -218,6 +229,32 @@ ASound::~ASound() {
_mixer->stopHandle(_soundHandle);
}
+void ASound::validate() {
+ Common::File f;
+ static const char *const MD5[] = {
+ "205398468de2c8873b7d4d73d5be8ddc",
+ "f9b2d944a2fb782b1af5c0ad592306d3",
+ "7431f8dad77d6ddfc24e6f3c0c4ac7df",
+ "eb1f3f5a4673d3e73d8ac1818c957cf4",
+ "f936dd853073fa44f3daac512e91c476",
+ "3dc139d3e02437a6d9b732072407c366",
+ "af0edab2934947982e9a405476702e03",
+ "8cbc25570b50ba41c9b5361cad4fbedc",
+ "a31e4783e098f633cbb6689adb41dd4f"
+ };
+
+ for (int i = 1; i <= 9; ++i) {
+ Common::String filename = Common::String::format("ASOUND.00%d", i);
+ if (!f.open(filename))
+ error("Could not process - %s", filename.c_str());
+ Common::String md5str = Common::computeStreamMD5AsString(f, 8192);
+ f.close();
+
+ if (md5str != MD5[i - 1])
+ error("Invalid sound file - %s", filename.c_str());
+ }
+}
+
void ASound::adlibInit() {
write(4, 0x60);
write(4, 0x80);
@@ -256,6 +293,17 @@ 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));
}
@@ -304,6 +352,7 @@ 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);
@@ -422,6 +471,10 @@ 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();
@@ -489,7 +542,7 @@ 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;
@@ -543,7 +596,7 @@ void ASound::pollActiveChannel() {
break;
case 8:
- chan->_field1D = *++pSrc;
+ chan->_field1D = (int8)*++pSrc;
chan->_pSrc += 2;
break;
@@ -564,7 +617,7 @@ void ASound::pollActiveChannel() {
if (chan->_fieldE) {
chan->_pSrc += 2;
} else {
- chan->_field1E = *pSrc >> 1;
+ chan->_volumeOffset = *pSrc >> 1;
updateFlag = true;
chan->_pSrc += 2;
}
@@ -608,7 +661,7 @@ 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;
@@ -617,7 +670,7 @@ void ASound::pollActiveChannel() {
newVal = 63;
}
- chan->_field1E = newVal;
+ chan->_volumeOffset = newVal;
updateFlag = true;
}
}
@@ -682,8 +735,8 @@ 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);
@@ -700,32 +753,18 @@ 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) {
@@ -821,6 +860,12 @@ int ASound::readBuffer(int16 *buffer, const int numSamples) {
return numSamples;
}
+void ASound::setVolume(int volume) {
+ _masterVolume = volume;
+ if (!volume)
+ command0();
+}
+
int ASound::command0() {
bool isDisabled = _isDisabled;
_isDisabled = true;
@@ -978,22 +1023,22 @@ 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;
}