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-rw-r--r--sound/audiostream.h5
-rw-r--r--sound/decoders/adpcm.cpp13
-rw-r--r--sound/decoders/adpcm.h17
-rw-r--r--sound/decoders/flac.cpp5
-rw-r--r--sound/decoders/vorbis.cpp5
-rw-r--r--sound/decoders/wave.cpp20
-rw-r--r--sound/mididrv.cpp45
-rw-r--r--sound/mixer.cpp1
-rw-r--r--sound/mods/tfmx.cpp4
-rw-r--r--sound/rate_arm.cpp14
-rw-r--r--sound/softsynth/cms.cpp242
-rw-r--r--sound/softsynth/fmtowns_pc98/towns_audio.cpp1
-rw-r--r--sound/softsynth/fmtowns_pc98/towns_euphony.cpp4
-rw-r--r--sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp1
-rw-r--r--sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp1
-rw-r--r--sound/softsynth/mt32/mt32_file.cpp18
-rw-r--r--sound/softsynth/mt32/mt32_file.h2
-rw-r--r--sound/softsynth/mt32/tables.cpp2
18 files changed, 208 insertions, 192 deletions
diff --git a/sound/audiostream.h b/sound/audiostream.h
index 64ad128931..b2c012841d 100644
--- a/sound/audiostream.h
+++ b/sound/audiostream.h
@@ -276,11 +276,6 @@ private:
* The same caveats apply to SubSeekableAudioStream as do to SeekableAudioStream.
*
* Manipulating the parent stream directly /will/ mess up a substream.
- *
- * IMPORTANT:
- * Note for engine authors. This object is currently under inspection. In case
- * we need to revise the looping API we might drop this. So if you really need
- * something like this object, please drop a mail to LordHoto.
*/
class SubSeekableAudioStream : public SeekableAudioStream {
public:
diff --git a/sound/decoders/adpcm.cpp b/sound/decoders/adpcm.cpp
index c8a907d13e..8a27658e4b 100644
--- a/sound/decoders/adpcm.cpp
+++ b/sound/decoders/adpcm.cpp
@@ -290,8 +290,8 @@ int Apple_ADPCMStream::readBuffer(int16 *buffer, const int numSamples) {
class MSIma_ADPCMStream : public Ima_ADPCMStream {
public:
- MSIma_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign)
- : Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign) {
+ MSIma_ADPCMStream(Common::SeekableReadStream *stream, DisposeAfterUse::Flag disposeAfterUse, uint32 size, int rate, int channels, uint32 blockAlign, bool invertSamples = false)
+ : Ima_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign), _invertSamples(invertSamples) {
if (blockAlign == 0)
error("ADPCMStream(): blockAlign isn't specified for MS IMA ADPCM");
}
@@ -305,6 +305,9 @@ public:
int readBufferMSIMA1(int16 *buffer, const int numSamples);
int readBufferMSIMA2(int16 *buffer, const int numSamples);
+
+private:
+ bool _invertSamples; // Some implementations invert the way samples are decoded
};
int MSIma_ADPCMStream::readBufferMSIMA1(int16 *buffer, const int numSamples) {
@@ -324,8 +327,8 @@ int MSIma_ADPCMStream::readBufferMSIMA1(int16 *buffer, const int numSamples) {
for (; samples < numSamples && _blockPos[0] < _blockAlign && !_stream->eos() && _stream->pos() < _endpos; samples += 2) {
data = _stream->readByte();
_blockPos[0]++;
- buffer[samples] = decodeIMA(data & 0x0f);
- buffer[samples + 1] = decodeIMA((data >> 4) & 0x0f);
+ buffer[samples] = decodeIMA(_invertSamples ? (data >> 4) & 0x0f : data & 0x0f);
+ buffer[samples + 1] = decodeIMA(_invertSamples ? data & 0x0f : (data >> 4) & 0x0f);
}
}
return samples;
@@ -733,6 +736,8 @@ RewindableAudioStream *makeADPCMStream(Common::SeekableReadStream *stream, Dispo
return new Oki_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
case kADPCMMSIma:
return new MSIma_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
+ case kADPCMMSImaLastExpress:
+ return new MSIma_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign, true);
case kADPCMMS:
return new MS_ADPCMStream(stream, disposeAfterUse, size, rate, channels, blockAlign);
case kADPCMTinsel4:
diff --git a/sound/decoders/adpcm.h b/sound/decoders/adpcm.h
index 04dbb1a521..edcdc01ce9 100644
--- a/sound/decoders/adpcm.h
+++ b/sound/decoders/adpcm.h
@@ -50,14 +50,15 @@ class RewindableAudioStream;
// Usually, if the audio stream we're trying to play has the FourCC header
// string intact, it's easy to discern which encoding is used
enum typesADPCM {
- kADPCMOki, // Dialogic/Oki ADPCM (aka VOX)
- kADPCMMSIma, // Microsoft IMA ADPCM
- kADPCMMS, // Microsoft ADPCM
- kADPCMTinsel4, // 4-bit ADPCM used by the Tinsel engine
- kADPCMTinsel6, // 6-bit ADPCM used by the Tinsel engine
- kADPCMTinsel8, // 8-bit ADPCM used by the Tinsel engine
- kADPCMIma, // Standard IMA ADPCM
- kADPCMApple // Apple QuickTime IMA ADPCM
+ kADPCMOki, // Dialogic/Oki ADPCM (aka VOX)
+ kADPCMMSIma, // Microsoft IMA ADPCM
+ kADPCMMSImaLastExpress, // Microsoft IMA ADPCM (with inverted samples)
+ kADPCMMS, // Microsoft ADPCM
+ kADPCMTinsel4, // 4-bit ADPCM used by the Tinsel engine
+ kADPCMTinsel6, // 6-bit ADPCM used by the Tinsel engine
+ kADPCMTinsel8, // 8-bit ADPCM used by the Tinsel engine
+ kADPCMIma, // Standard IMA ADPCM
+ kADPCMApple // Apple QuickTime IMA ADPCM
};
/**
diff --git a/sound/decoders/flac.cpp b/sound/decoders/flac.cpp
index 560b83e1ee..080141f224 100644
--- a/sound/decoders/flac.cpp
+++ b/sound/decoders/flac.cpp
@@ -23,6 +23,9 @@
*
*/
+// Disable symbol overrides for FILE as that is used in FLAC headers
+#define FORBIDDEN_SYMBOL_EXCEPTION_FILE
+
#include "sound/decoders/flac.h"
#ifdef USE_FLAC
@@ -302,7 +305,7 @@ int FLACStream::readBuffer(int16 *buffer, const int numSamples) {
const uint numChannels = getChannels();
if (numChannels == 0) {
- warning("FLACStream: Stream not sucessfully initialised, cant playback");
+ warning("FLACStream: Stream not successfully initialised, cant playback");
return -1; // streaminfo wasnt read!
}
diff --git a/sound/decoders/vorbis.cpp b/sound/decoders/vorbis.cpp
index 5aeb40c139..425eb6b751 100644
--- a/sound/decoders/vorbis.cpp
+++ b/sound/decoders/vorbis.cpp
@@ -23,6 +23,11 @@
*
*/
+// Disable symbol overrides for FILE and fseek as those are used in the
+// Vorbis headers.
+#define FORBIDDEN_SYMBOL_EXCEPTION_FILE
+#define FORBIDDEN_SYMBOL_EXCEPTION_fseek
+
#include "sound/decoders/vorbis.h"
#ifdef USE_VORBIS
diff --git a/sound/decoders/wave.cpp b/sound/decoders/wave.cpp
index eeab026ae5..fcaace5301 100644
--- a/sound/decoders/wave.cpp
+++ b/sound/decoders/wave.cpp
@@ -90,15 +90,15 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
if (blockAlign_ != 0)
*blockAlign_ = blockAlign;
#if 0
- printf("WAVE information:\n");
- printf(" total size: %d\n", wavLength);
- printf(" fmt size: %d\n", fmtLength);
- printf(" type: %d\n", type);
- printf(" numChannels: %d\n", numChannels);
- printf(" samplesPerSec: %d\n", samplesPerSec);
- printf(" avgBytesPerSec: %d\n", avgBytesPerSec);
- printf(" blockAlign: %d\n", blockAlign);
- printf(" bitsPerSample: %d\n", bitsPerSample);
+ debug("WAVE information:");
+ debug(" total size: %d", wavLength);
+ debug(" fmt size: %d", fmtLength);
+ debug(" type: %d", type);
+ debug(" numChannels: %d", numChannels);
+ debug(" samplesPerSec: %d", samplesPerSec);
+ debug(" avgBytesPerSec: %d", avgBytesPerSec);
+ debug(" blockAlign: %d", blockAlign);
+ debug(" bitsPerSample: %d", bitsPerSample);
#endif
if (type != 1 && type != 2 && type != 17) {
@@ -152,7 +152,7 @@ bool loadWAVFromStream(Common::SeekableReadStream &stream, int &size, int &rate,
offset = stream.readUint32LE();
#if 0
- printf(" found a '%s' tag of size %d\n", buf, offset);
+ debug(" found a '%s' tag of size %d", buf, offset);
#endif
} while (memcmp(buf, "data", 4) != 0);
diff --git a/sound/mididrv.cpp b/sound/mididrv.cpp
index aa9f8797ba..20d5a4e233 100644
--- a/sound/mididrv.cpp
+++ b/sound/mididrv.cpp
@@ -209,35 +209,40 @@ MidiDriver::DeviceHandle MidiDriver::detectDevice(int flags) {
hdl = getDeviceHandle("auto");
const MusicType type = getMusicType(hdl);
- if (type != MT_AUTO && type != MT_INVALID) {
- if (flags & MDT_PREFER_MT32)
- // If we have a preferred MT32 device we disable the gm/mt32 mapping (more about this in mididrv.h)
- _forceTypeMT32 = true;
- return hdl;
- }
+ // If have a "Don't use GM/MT-32" setting we skip this part and jump
+ // to AdLib, PC Speaker etc. detection right away.
+ if (type != MT_NULL) {
+ if (type != MT_AUTO && type != MT_INVALID) {
+ if (flags & MDT_PREFER_MT32)
+ // If we have a preferred MT32 device we disable the gm/mt32 mapping (more about this in mididrv.h)
+ _forceTypeMT32 = true;
+
+ return hdl;
+ }
- // If we have no specific device selected (neither in the scummvm nor in the game domain)
- // and no preferred MT32 or GM device selected we arrive here.
- // If MT32 is preferred we try for the first available device with music type 'MT_MT32' (usually the mt32 emulator)
- if (flags & MDT_PREFER_MT32) {
+ // If we have no specific device selected (neither in the scummvm nor in the game domain)
+ // and no preferred MT32 or GM device selected we arrive here.
+ // If MT32 is preferred we try for the first available device with music type 'MT_MT32' (usually the mt32 emulator)
+ if (flags & MDT_PREFER_MT32) {
+ for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
+ MusicDevices i = (**m)->getDevices();
+ for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
+ if (d->getMusicType() == MT_MT32)
+ return d->getHandle();
+ }
+ }
+ }
+
+ // Now we default to the first available device with music type 'MT_GM'
for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
MusicDevices i = (**m)->getDevices();
for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
- if (d->getMusicType() == MT_MT32)
+ if (d->getMusicType() == MT_GM || d->getMusicType() == MT_GS)
return d->getHandle();
}
}
}
-
- // Now we default to the first available device with music type 'MT_GM'
- for (MusicPlugin::List::const_iterator m = p.begin(); m != p.end(); ++m) {
- MusicDevices i = (**m)->getDevices();
- for (MusicDevices::iterator d = i.begin(); d != i.end(); ++d) {
- if (d->getMusicType() == MT_GM || d->getMusicType() == MT_GS)
- return d->getHandle();
- }
- }
}
MusicType tp = MT_AUTO;
diff --git a/sound/mixer.cpp b/sound/mixer.cpp
index cb3cc2a0f3..c40aa95d73 100644
--- a/sound/mixer.cpp
+++ b/sound/mixer.cpp
@@ -435,6 +435,7 @@ int MixerImpl::getVolumeForSoundType(SoundType type) const {
return _volumeForSoundType[type];
}
+
#pragma mark -
#pragma mark --- Channel implementations ---
#pragma mark -
diff --git a/sound/mods/tfmx.cpp b/sound/mods/tfmx.cpp
index b65a998e82..6ed1abcfb5 100644
--- a/sound/mods/tfmx.cpp
+++ b/sound/mods/tfmx.cpp
@@ -281,7 +281,7 @@ void Tfmx::macroRun(ChannelContext &channel) {
continue;
case 0x04: // Wait. Parameters: Ticks to wait(W).
- // TODO: some unkown Parameter? (macroPtr[1] & 1)
+ // TODO: some unknown Parameter? (macroPtr[1] & 1)
channel.macroWait = READ_BE_UINT16(&macroPtr[2]);
break;
@@ -1154,7 +1154,7 @@ void displayMacroStep(const void * const vptr) {
if (macroData[0] < ARRAYSIZE(tableMacros))
debug("%s %02X%02X%02X", tableMacros[macroData[0]], macroData[1], macroData[2], macroData[3]);
else
- debug("Unkown Macro #%02X %02X%02X%02X", macroData[0], macroData[1], macroData[2], macroData[3]);
+ debug("Unknown Macro #%02X %02X%02X%02X", macroData[0], macroData[1], macroData[2], macroData[3]);
}
void displayPatternstep(const void * const vptr) {
diff --git a/sound/rate_arm.cpp b/sound/rate_arm.cpp
index 5bab6ca59d..63008fcb87 100644
--- a/sound/rate_arm.cpp
+++ b/sound/rate_arm.cpp
@@ -168,18 +168,16 @@ extern "C" int SimpleRate_readFudge(Audio::AudioStream &input,
int16 *a, int b)
{
#ifdef DEBUG_RATECONV
- fprintf(stderr, "Reading ptr=%x n%d\n", a, b);
- fflush(stderr);
+ debug("Reading ptr=%x n%d", a, b);
#endif
- return input.readBuffer(a, b);
+ return input.readBuffer(a, b);
}
template<bool stereo, bool reverseStereo>
int SimpleRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
#ifdef DEBUG_RATECONV
-fprintf(stderr, "Simple st=%d rev=%d\n", stereo, reverseStereo);
-fflush(stderr);
+ debug("Simple st=%d rev=%d", stereo, reverseStereo);
#endif
st_sample_t *ostart = obuf;
@@ -318,8 +316,7 @@ template<bool stereo, bool reverseStereo>
int LinearRateConverter<stereo, reverseStereo>::flow(AudioStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
#ifdef DEBUG_RATECONV
-fprintf(stderr, "Linear st=%d rev=%d\n", stereo, reverseStereo);
-fflush(stderr);
+ debug("Linear st=%d rev=%d", stereo, reverseStereo);
#endif
st_sample_t *ostart = obuf;
@@ -393,8 +390,7 @@ public:
assert(input.isStereo() == stereo);
#ifdef DEBUG_RATECONV
-fprintf(stderr, "Copy st=%d rev=%d\n", stereo, reverseStereo);
-fflush(stderr);
+ debug("Copy st=%d rev=%d", stereo, reverseStereo);
#endif
st_size_t len;
st_sample_t *ostart = obuf;
diff --git a/sound/softsynth/cms.cpp b/sound/softsynth/cms.cpp
index 88f04a1ab9..b307146f14 100644
--- a/sound/softsynth/cms.cpp
+++ b/sound/softsynth/cms.cpp
@@ -86,35 +86,39 @@ static const int amplitude_lookup[16] = {
void CMSEmulator::portWrite(int port, int val) {
switch (port) {
- case 0x220:
- portWriteIntern(0, 1, val);
- break;
-
- case 0x221:
- _saa1099[0].selected_reg = val & 0x1f;
- if (_saa1099[0].selected_reg == 0x18 || _saa1099[0].selected_reg == 0x19) {
- /* clock the envelope channels */
- if (_saa1099[0].env_clock[0]) envelope(0, 0);
- if (_saa1099[0].env_clock[1]) envelope(0, 1);
- }
- break;
-
- case 0x222:
- portWriteIntern(1, 1, val);
- break;
-
- case 0x223:
- _saa1099[1].selected_reg = val & 0x1f;
- if (_saa1099[1].selected_reg == 0x18 || _saa1099[1].selected_reg == 0x19) {
- /* clock the envelope channels */
- if (_saa1099[1].env_clock[0]) envelope(1, 0);
- if (_saa1099[1].env_clock[1]) envelope(1, 1);
- }
- break;
+ case 0x220:
+ portWriteIntern(0, 1, val);
+ break;
+
+ case 0x221:
+ _saa1099[0].selected_reg = val & 0x1f;
+ if (_saa1099[0].selected_reg == 0x18 || _saa1099[0].selected_reg == 0x19) {
+ /* clock the envelope channels */
+ if (_saa1099[0].env_clock[0])
+ envelope(0, 0);
+ if (_saa1099[0].env_clock[1])
+ envelope(0, 1);
+ }
+ break;
+
+ case 0x222:
+ portWriteIntern(1, 1, val);
+ break;
+
+ case 0x223:
+ _saa1099[1].selected_reg = val & 0x1f;
+ if (_saa1099[1].selected_reg == 0x18 || _saa1099[1].selected_reg == 0x19) {
+ /* clock the envelope channels */
+ if (_saa1099[1].env_clock[0])
+ envelope(1, 0);
+ if (_saa1099[1].env_clock[1])
+ envelope(1, 1);
+ }
+ break;
- default:
- warning("CMSEmulator got port: 0x%X", port);
- break;
+ default:
+ warning("CMSEmulator got port: 0x%X", port);
+ break;
}
}
@@ -177,10 +181,10 @@ void CMSEmulator::update(int chip, int16 *buffer, int length) {
for (ch = 0; ch < 2; ch++) {
switch (saa->noise_params[ch]) {
- case 0: saa->noise[ch].freq = 31250.0 * 2; break;
- case 1: saa->noise[ch].freq = 15625.0 * 2; break;
- case 2: saa->noise[ch].freq = 7812.5 * 2; break;
- case 3: saa->noise[ch].freq = saa->channels[ch * 3].freq; break;
+ case 0: saa->noise[ch].freq = 31250.0 * 2; break;
+ case 1: saa->noise[ch].freq = 15625.0 * 2; break;
+ case 2: saa->noise[ch].freq = 7812.5 * 2; break;
+ case 3: saa->noise[ch].freq = saa->channels[ch * 3].freq; break;
}
}
@@ -254,95 +258,95 @@ void CMSEmulator::portWriteIntern(int chip, int offset, int data) {
int ch;
switch (reg) {
- /* channel i amplitude */
- case 0x00:
- case 0x01:
- case 0x02:
- case 0x03:
- case 0x04:
- case 0x05:
- ch = reg & 7;
- saa->channels[ch].amplitude[LEFT] = amplitude_lookup[data & 0x0f];
- saa->channels[ch].amplitude[RIGHT] = amplitude_lookup[(data >> 4) & 0x0f];
- break;
-
- /* channel i frequency */
- case 0x08:
- case 0x09:
- case 0x0a:
- case 0x0b:
- case 0x0c:
- case 0x0d:
- ch = reg & 7;
- saa->channels[ch].frequency = data & 0xff;
- break;
-
- /* channel i octave */
- case 0x10:
- case 0x11:
- case 0x12:
- ch = (reg - 0x10) << 1;
- saa->channels[ch + 0].octave = data & 0x07;
- saa->channels[ch + 1].octave = (data >> 4) & 0x07;
- break;
-
- /* channel i frequency enable */
- case 0x14:
- saa->channels[0].freq_enable = data & 0x01;
- saa->channels[1].freq_enable = data & 0x02;
- saa->channels[2].freq_enable = data & 0x04;
- saa->channels[3].freq_enable = data & 0x08;
- saa->channels[4].freq_enable = data & 0x10;
- saa->channels[5].freq_enable = data & 0x20;
- break;
-
- /* channel i noise enable */
- case 0x15:
- saa->channels[0].noise_enable = data & 0x01;
- saa->channels[1].noise_enable = data & 0x02;
- saa->channels[2].noise_enable = data & 0x04;
- saa->channels[3].noise_enable = data & 0x08;
- saa->channels[4].noise_enable = data & 0x10;
- saa->channels[5].noise_enable = data & 0x20;
- break;
-
- /* noise generators parameters */
- case 0x16:
- saa->noise_params[0] = data & 0x03;
- saa->noise_params[1] = (data >> 4) & 0x03;
- break;
-
- /* envelope generators parameters */
- case 0x18:
- case 0x19:
- ch = reg - 0x18;
- saa->env_reverse_right[ch] = data & 0x01;
- saa->env_mode[ch] = (data >> 1) & 0x07;
- saa->env_bits[ch] = data & 0x10;
- saa->env_clock[ch] = data & 0x20;
- saa->env_enable[ch] = data & 0x80;
- /* reset the envelope */
- saa->env_step[ch] = 0;
- break;
-
- /* channels enable & reset generators */
- case 0x1c:
- saa->all_ch_enable = data & 0x01;
- saa->sync_state = data & 0x02;
- if (data & 0x02) {
- int i;
- /* Synch & Reset generators */
- for (i = 0; i < 6; i++) {
- saa->channels[i].level = 0;
- saa->channels[i].counter = 0.0;
- }
+ /* channel i amplitude */
+ case 0x00:
+ case 0x01:
+ case 0x02:
+ case 0x03:
+ case 0x04:
+ case 0x05:
+ ch = reg & 7;
+ saa->channels[ch].amplitude[LEFT] = amplitude_lookup[data & 0x0f];
+ saa->channels[ch].amplitude[RIGHT] = amplitude_lookup[(data >> 4) & 0x0f];
+ break;
+
+ /* channel i frequency */
+ case 0x08:
+ case 0x09:
+ case 0x0a:
+ case 0x0b:
+ case 0x0c:
+ case 0x0d:
+ ch = reg & 7;
+ saa->channels[ch].frequency = data & 0xff;
+ break;
+
+ /* channel i octave */
+ case 0x10:
+ case 0x11:
+ case 0x12:
+ ch = (reg - 0x10) << 1;
+ saa->channels[ch + 0].octave = data & 0x07;
+ saa->channels[ch + 1].octave = (data >> 4) & 0x07;
+ break;
+
+ /* channel i frequency enable */
+ case 0x14:
+ saa->channels[0].freq_enable = data & 0x01;
+ saa->channels[1].freq_enable = data & 0x02;
+ saa->channels[2].freq_enable = data & 0x04;
+ saa->channels[3].freq_enable = data & 0x08;
+ saa->channels[4].freq_enable = data & 0x10;
+ saa->channels[5].freq_enable = data & 0x20;
+ break;
+
+ /* channel i noise enable */
+ case 0x15:
+ saa->channels[0].noise_enable = data & 0x01;
+ saa->channels[1].noise_enable = data & 0x02;
+ saa->channels[2].noise_enable = data & 0x04;
+ saa->channels[3].noise_enable = data & 0x08;
+ saa->channels[4].noise_enable = data & 0x10;
+ saa->channels[5].noise_enable = data & 0x20;
+ break;
+
+ /* noise generators parameters */
+ case 0x16:
+ saa->noise_params[0] = data & 0x03;
+ saa->noise_params[1] = (data >> 4) & 0x03;
+ break;
+
+ /* envelope generators parameters */
+ case 0x18:
+ case 0x19:
+ ch = reg - 0x18;
+ saa->env_reverse_right[ch] = data & 0x01;
+ saa->env_mode[ch] = (data >> 1) & 0x07;
+ saa->env_bits[ch] = data & 0x10;
+ saa->env_clock[ch] = data & 0x20;
+ saa->env_enable[ch] = data & 0x80;
+ /* reset the envelope */
+ saa->env_step[ch] = 0;
+ break;
+
+ /* channels enable & reset generators */
+ case 0x1c:
+ saa->all_ch_enable = data & 0x01;
+ saa->sync_state = data & 0x02;
+ if (data & 0x02) {
+ int i;
+ /* Synch & Reset generators */
+ for (i = 0; i < 6; i++) {
+ saa->channels[i].level = 0;
+ saa->channels[i].counter = 0.0;
}
- break;
+ }
+ break;
- default:
- // The CMS allows all registers to be written, so we just output some debug
- // message here
- debug(5, "CMS Unkown write to reg %x with %x",reg, data);
+ default:
+ // The CMS allows all registers to be written, so we just output some debug
+ // message here
+ debug(5, "CMS Unknown write to reg %x with %x",reg, data);
}
}
diff --git a/sound/softsynth/fmtowns_pc98/towns_audio.cpp b/sound/softsynth/fmtowns_pc98/towns_audio.cpp
index e6da237881..a9e67c5266 100644
--- a/sound/softsynth/fmtowns_pc98/towns_audio.cpp
+++ b/sound/softsynth/fmtowns_pc98/towns_audio.cpp
@@ -225,7 +225,6 @@ TownsAudioInterface::TownsAudioInterface(Audio::Mixer *mixer, TownsAudioInterfac
}
TownsAudioInterface::~TownsAudioInterface() {
- reset();
_ready = false;
deinit();
diff --git a/sound/softsynth/fmtowns_pc98/towns_euphony.cpp b/sound/softsynth/fmtowns_pc98/towns_euphony.cpp
index 7b52b4594f..e6f94b29b5 100644
--- a/sound/softsynth/fmtowns_pc98/towns_euphony.cpp
+++ b/sound/softsynth/fmtowns_pc98/towns_euphony.cpp
@@ -36,17 +36,15 @@ TownsEuphonyDriver::TownsEuphonyDriver(Audio::Mixer *mixer) : _activeChannels(0)
}
TownsEuphonyDriver::~TownsEuphonyDriver() {
+ delete _intf;
delete[] _activeChannels;
delete[] _sustainChannels;
delete[] _assignedChannels;
-
delete[] _tEnable;
delete[] _tMode;
delete[] _tOrdr;
delete[] _tLevel;
delete[] _tTranspose;
-
- delete _intf;
}
bool TownsEuphonyDriver::init() {
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp b/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
index 8047616dbf..303f08e6b1 100644
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
+++ b/sound/softsynth/fmtowns_pc98/towns_pc98_driver.cpp
@@ -1053,7 +1053,6 @@ TownsPC98_AudioDriver::TownsPC98_AudioDriver(Audio::Mixer *mixer, EmuType type)
}
TownsPC98_AudioDriver::~TownsPC98_AudioDriver() {
- reset();
_ready = false;
deinit();
diff --git a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp b/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp
index e779812c42..62f7d39771 100644
--- a/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp
+++ b/sound/softsynth/fmtowns_pc98/towns_pc98_fmsynth.cpp
@@ -1180,7 +1180,6 @@ int TownsPC98_FmSynth::readBuffer(int16 *buffer, const int numSamples) {
void TownsPC98_FmSynth::deinit() {
_ready = false;
_mixer->stopHandle(_soundHandle);
- Common::StackLock lock(_mutex);
_timers[0].cb = _timers[1].cb = &TownsPC98_FmSynth::idleTimerCallback;
}
diff --git a/sound/softsynth/mt32/mt32_file.cpp b/sound/softsynth/mt32/mt32_file.cpp
index f4eba73d33..ce5c2874c4 100644
--- a/sound/softsynth/mt32/mt32_file.cpp
+++ b/sound/softsynth/mt32/mt32_file.cpp
@@ -19,6 +19,12 @@
* IN THE SOFTWARE.
*/
+
+// FIXME: Disable symbol overrides so that we can use system headers.
+// But we *really* should get rid of this usage of FILE, fopen etc.
+#define FORBIDDEN_SYMBOL_ALLOW_ALL
+
+
#include <stdio.h>
#include "mt32emu.h"
@@ -37,15 +43,15 @@ namespace MT32Emu {
}
void ANSIFile::close() {
- fclose(fp);
+ fclose((FILE *)fp);
}
size_t ANSIFile::read(void *in, size_t size) {
- return fread(in, 1, size, fp);
+ return fread(in, 1, size, (FILE *)fp);
}
bool ANSIFile::readBit8u(Bit8u *in) {
- int c = fgetc(fp);
+ int c = fgetc((FILE *)fp);
if (c == EOF)
return false;
*in = (Bit8u)c;
@@ -69,11 +75,11 @@ namespace MT32Emu {
}
size_t ANSIFile::write(const void *out, size_t size) {
- return fwrite(out, 1, size, fp);
+ return fwrite(out, 1, size, (FILE *)fp);
}
bool ANSIFile::writeBit8u(Bit8u out) {
- return fputc(out, fp) != EOF;
+ return fputc(out, (FILE *)fp) != EOF;
}
bool File::writeBit16u(Bit16u out) {
@@ -103,6 +109,6 @@ namespace MT32Emu {
}
bool ANSIFile::isEOF() {
- return feof(fp) != 0;
+ return feof((FILE *)fp) != 0;
}
}
diff --git a/sound/softsynth/mt32/mt32_file.h b/sound/softsynth/mt32/mt32_file.h
index 27c8ccbe46..b311ad9626 100644
--- a/sound/softsynth/mt32/mt32_file.h
+++ b/sound/softsynth/mt32/mt32_file.h
@@ -49,7 +49,7 @@ public:
class ANSIFile: public File {
private:
- FILE *fp;
+ void *fp;
public:
bool open(const char *filename, OpenMode mode);
void close();
diff --git a/sound/softsynth/mt32/tables.cpp b/sound/softsynth/mt32/tables.cpp
index b0414154dc..722a5e3c41 100644
--- a/sound/softsynth/mt32/tables.cpp
+++ b/sound/softsynth/mt32/tables.cpp
@@ -587,7 +587,7 @@ File *Tables::initNote(Synth *synth, NoteLookup *noteLookup, float note, float r
initSaw(noteLookup, noteLookup->div2);
//synth->printDebug("Note %f; freq=%f, div=%f", note, freq, rate / freq);
- file = initWave(synth, noteLookup, (const float)WGAMP, div2, file);
+ file = initWave(synth, noteLookup, WGAMP, div2, file);
// Create the pitch tables
if (noteLookup->wavTable == NULL)
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-28 16:01:19 +0000