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-rw-r--r--sound/adpcm.cpp10
-rw-r--r--sound/fmopl.h4
-rw-r--r--sound/softsynth/opl/dosbox.cpp22
-rw-r--r--sound/softsynth/opl/opl_impl.h14
-rw-r--r--sound/softsynth/opl/opl_inc.h8
-rw-r--r--sound/timestamp.h10
-rw-r--r--sound/vag.cpp18
-rw-r--r--sound/vag.h6
8 files changed, 46 insertions, 46 deletions
diff --git a/sound/adpcm.cpp b/sound/adpcm.cpp
index a5f2f7c4eb..37b14140b7 100644
--- a/sound/adpcm.cpp
+++ b/sound/adpcm.cpp
@@ -112,12 +112,12 @@ public:
ADPCMInputStream::ADPCMInputStream(Common::SeekableReadStream *stream, bool disposeAfterUse, uint32 size, typesADPCM type, int rate, int channels, uint32 blockAlign, uint numLoops)
: _stream(stream), _disposeAfterUse(disposeAfterUse), _channels(channels), _type(type), _blockAlign(blockAlign), _rate(rate), _numLoops(numLoops) {
-
+
if (type == kADPCMMSIma && blockAlign == 0)
error("ADPCMInputStream(): blockAlign isn't specified for MS IMA ADPCM");
if (type == kADPCMMS && blockAlign == 0)
error("ADPCMInputStream(): blockAlign isn't specified for MS ADPCM");
-
+
if (type == kADPCMTinsel4 && blockAlign == 0)
error("ADPCMInputStream(): blockAlign isn't specified for Tinsel 4-bit ADPCM");
if (type == kADPCMTinsel6 && blockAlign == 0)
@@ -131,7 +131,7 @@ ADPCMInputStream::ADPCMInputStream(Common::SeekableReadStream *stream, bool disp
error("ADPCMInputStream(): Tinsel 6-bit ADPCM only supports mono");
if (type == kADPCMTinsel8 && channels != 1)
error("ADPCMInputStream(): Tinsel 8-bit ADPCM only supports mono");
-
+
_startpos = stream->pos();
_endpos = _startpos + size;
_curLoop = 0;
@@ -182,7 +182,7 @@ int ADPCMInputStream::readBuffer(int16 *buffer, const int numSamples) {
error("Unsupported ADPCM encoding");
break;
}
-
+
// Loop if necessary
if (samplesDecoded < numSamples || _stream->pos() == _endpos) {
_curLoop++;
@@ -192,7 +192,7 @@ int ADPCMInputStream::readBuffer(int16 *buffer, const int numSamples) {
return samplesDecoded + readBuffer(buffer + samplesDecoded, numSamples - samplesDecoded);
}
}
-
+
return samplesDecoded;
}
diff --git a/sound/fmopl.h b/sound/fmopl.h
index 81d9610a44..d0b15368e0 100644
--- a/sound/fmopl.h
+++ b/sound/fmopl.h
@@ -76,7 +76,7 @@ public:
/**
* Creates the specific driver with a specific type setup.
- */
+ */
static OPL *create(DriverId driver, OplType type);
/**
@@ -86,7 +86,7 @@ public:
static OPL *create(OplType type = kOpl2) { return create(detect(type), type); }
private:
- static const EmulatorDescription _drivers[];
+ static const EmulatorDescription _drivers[];
};
class OPL {
diff --git a/sound/softsynth/opl/dosbox.cpp b/sound/softsynth/opl/dosbox.cpp
index cbb3090608..25fda7f2e6 100644
--- a/sound/softsynth/opl/dosbox.cpp
+++ b/sound/softsynth/opl/dosbox.cpp
@@ -40,7 +40,7 @@
#include <string.h>
namespace OPL {
-namespace DOSBox {
+namespace DOSBox {
Timer::Timer() {
masked = false;
@@ -51,7 +51,7 @@ Timer::Timer() {
}
void Timer::update(double time) {
- if (!enabled || !delay)
+ if (!enabled || !delay)
return;
double deltaStart = time - startTime;
// Only set the overflow flag when not masked
@@ -66,7 +66,7 @@ void Timer::reset(double time) {
double delta = (time - startTime);
double rem = fmod(delta, delay);
double next = delay - rem;
- startTime = time + next;
+ startTime = time + next;
}
void Timer::stop() {
@@ -116,7 +116,7 @@ bool Chip::write(uint32 reg, uint8 val) {
timer[1].stop();
timer[1].masked = (val & 0x20) > 0;
-
+
if (timer[1].masked)
timer[1].overflow = false;
}
@@ -148,7 +148,7 @@ namespace OPL2 {
#include "opl_impl.h"
struct Handler : public DOSBox::Handler {
- void writeReg(uint32 reg, uint8 val) {
+ void writeReg(uint32 reg, uint8 val) {
adlib_write(reg, val);
}
@@ -171,7 +171,7 @@ namespace OPL3 {
#include "opl_impl.h"
struct Handler : public DOSBox::Handler {
- void writeReg(uint32 reg, uint8 val) {
+ void writeReg(uint32 reg, uint8 val) {
adlib_write(reg, val);
}
@@ -217,13 +217,13 @@ bool OPL::init(int rate) {
case Config::kOpl3:
_handler = new OPL3::Handler();
break;
-
+
default:
return false;
}
_handler->init(rate);
-
+
if (_type == Config::kDualOpl2) {
// Setup opl3 mode in the hander
_handler->writeReg(0x105, 1);
@@ -234,7 +234,7 @@ bool OPL::init(int rate) {
}
void OPL::reset() {
- init(_rate);
+ init(_rate);
}
void OPL::write(int port, int val) {
@@ -325,7 +325,7 @@ void OPL::writeReg(int r, int v) {
void OPL::dualWrite(uint8 index, uint8 reg, uint8 val) {
// Make sure you don't use opl3 features
- // Don't allow write to disable opl3
+ // Don't allow write to disable opl3
if (reg == 5)
return;
@@ -334,7 +334,7 @@ void OPL::dualWrite(uint8 index, uint8 reg, uint8 val) {
val &= 3;
// Write to the timer?
- if (_chip[index].write(reg, val))
+ if (_chip[index].write(reg, val))
return;
// Enabling panning
diff --git a/sound/softsynth/opl/opl_impl.h b/sound/softsynth/opl/opl_impl.h
index c5ef87ce3c..5c68baa485 100644
--- a/sound/softsynth/opl/opl_impl.h
+++ b/sound/softsynth/opl/opl_impl.h
@@ -6,12 +6,12 @@
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
- *
+ *
* This library 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
* Lesser General Public License for more details.
- *
+ *
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
@@ -131,7 +131,7 @@ static fltype decrelconst[4] = {1/39.28064,1/31.41608,1/26.17344,1/22.44608};
void operator_advance(op_type* op_pt, Bit32s vib) {
op_pt->wfpos = op_pt->tcount; // waveform position
-
+
// advance waveform time
op_pt->tcount += op_pt->tinc;
op_pt->tcount += (Bit32s)(op_pt->tinc)*vib/FIXEDPT;
@@ -311,7 +311,7 @@ void change_attackrate(Bitu regbase, op_type* op_pt) {
op_pt->env_step_a = (1<<(steps<=12?12-steps:0))-1;
Bits step_num = (step_skip<=48)?(4-(step_skip&3)):0;
- static Bit8u step_skip_mask[5] = {0xff, 0xfe, 0xee, 0xba, 0xaa};
+ static Bit8u step_skip_mask[5] = {0xff, 0xfe, 0xee, 0xba, 0xaa};
op_pt->env_step_skip_a = step_skip_mask[step_num];
#if defined(OPLTYPE_IS_OPL3)
@@ -1001,7 +1001,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
operator_advance(&cptr[9],vibval1[i]);
opfuncs[cptr[9].op_state](&cptr[9]);
operator_output(&cptr[9],0,tremval1[i]);
-
+
Bit32s chanval = cptr[9].cval*2;
CHANVAL_OUT
}
@@ -1033,7 +1033,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
operator_advance(&cptr[9],vibval2[i]);
opfuncs[cptr[9].op_state](&cptr[9]);
operator_output(&cptr[9],cptr[0].cval*FIXEDPT,tremval2[i]);
-
+
Bit32s chanval = cptr[9].cval*2;
CHANVAL_OUT
}
@@ -1347,7 +1347,7 @@ void adlib_getsample(Bit16s* sndptr, Bits numsamples) {
} else {
// FM-FM-style synthesis (op1[fb] * op2 * op3 * op4)
- if ((cptr[0].op_state != OF_TYPE_OFF) || (cptr[9].op_state != OF_TYPE_OFF) ||
+ if ((cptr[0].op_state != OF_TYPE_OFF) || (cptr[9].op_state != OF_TYPE_OFF) ||
(cptr[3].op_state != OF_TYPE_OFF) || (cptr[3+9].op_state != OF_TYPE_OFF)) {
if ((cptr[0].vibrato) && (cptr[0].op_state != OF_TYPE_OFF)) {
vibval1 = vibval_var1;
diff --git a/sound/softsynth/opl/opl_inc.h b/sound/softsynth/opl/opl_inc.h
index 0798bd9f6a..b7b6ac1d41 100644
--- a/sound/softsynth/opl/opl_inc.h
+++ b/sound/softsynth/opl/opl_inc.h
@@ -6,12 +6,12 @@
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
- *
+ *
* This library 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
* Lesser General Public License for more details.
- *
+ *
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
@@ -132,7 +132,7 @@ typedef struct operator_struct {
Bit32u act_state; // activity state (regular, percussion)
bool sus_keep; // keep sustain level when decay finished
bool vibrato,tremolo; // vibrato/tremolo enable bits
-
+
// variables used to provide non-continuous envelopes
Bit32u generator_pos; // for non-standard sample rates we need to determine how many samples have passed
Bits cur_env_step; // current (standardized) sample position
@@ -151,7 +151,7 @@ Bitu chip_num;
op_type op[MAXOPERATORS];
Bits int_samplerate;
-
+
Bit8u status;
Bit32u index;
#if defined(OPLTYPE_IS_OPL3)
diff --git a/sound/timestamp.h b/sound/timestamp.h
index af0cf96ff9..deb86bc2d7 100644
--- a/sound/timestamp.h
+++ b/sound/timestamp.h
@@ -45,23 +45,23 @@ protected:
public:
Timestamp();
-
+
/**
* Set up a timestamp with a given time and framerate.
* @param msecs staring time in milliseconds
* @param frameRate number of frames per second
*/
Timestamp(uint32 msecs, int frameRate);
-
+
/** Adds a number of frames to a timestamp. */
Timestamp addFrames(int frames) const;
-
+
/** Computes the difference (# of frames) between this timestamp and b. */
int frameDiff(const Timestamp &b) const;
-
+
/** Computes the difference (# of milliseconds) between this timestamp and b. */
int msecsDiff(const Timestamp &b) const;
-
+
/** Determines the time in milliseconds described by this timestamp. */
uint32 msecs() const;
};
diff --git a/sound/vag.cpp b/sound/vag.cpp
index 46e6a64233..a08c1c0a72 100644
--- a/sound/vag.cpp
+++ b/sound/vag.cpp
@@ -22,7 +22,7 @@
* $Id$
*
*/
-
+
#include "sound/vag.h"
namespace Audio {
@@ -48,10 +48,10 @@ double f[5][2] = { { 0.0, 0.0 },
int VagStream::readBuffer(int16 *buffer, const int numSamples) {
int32 samplesDecoded = 0;
-
+
if (_samplesRemaining) {
byte i = 0;
-
+
for (i = 28 - _samplesRemaining; i < 28 && samplesDecoded < numSamples; i++) {
_samples[i] = _samples[i] + _s1 * f[_predictor][0] + _s2 * f[_predictor][1];
_s2 = _s1;
@@ -70,17 +70,17 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
_samplesRemaining = 0;
}
-
+
while (samplesDecoded < numSamples) {
byte i = 0;
-
+
_predictor = _stream->readByte();
byte shift = _predictor & 0xf;
_predictor >>= 4;
-
+
if (_stream->readByte() == 7)
return samplesDecoded;
-
+
for (i = 0; i < 28; i += 2) {
byte d = _stream->readByte();
int16 s = (d & 0xf) << 12;
@@ -92,7 +92,7 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
s |= 0xffff0000;
_samples[i + 1] = (double)(s >> shift);
}
-
+
for (i = 0; i < 28 && samplesDecoded < numSamples; i++) {
_samples[i] = _samples[i] + _s1 * f[_predictor][0] + _s2 * f[_predictor][1];
_s2 = _s1;
@@ -101,7 +101,7 @@ int VagStream::readBuffer(int16 *buffer, const int numSamples) {
buffer[samplesDecoded] = d;
samplesDecoded++;
}
-
+
if (i != 27)
_samplesRemaining = 28 - i;
}
diff --git a/sound/vag.h b/sound/vag.h
index 228f68279e..f6f79d2b81 100644
--- a/sound/vag.h
+++ b/sound/vag.h
@@ -33,19 +33,19 @@
#include "sound/audiostream.h"
#include "common/stream.h"
-
+
namespace Audio {
class VagStream : public Audio::AudioStream {
public:
VagStream(Common::SeekableReadStream *stream, bool loop = false, int rate = 11025);
~VagStream();
-
+
bool isStereo() const { return false; }
bool endOfData() const { return _stream->pos() == _stream->size(); }
int getRate() const { return _rate; }
int readBuffer(int16 *buffer, const int numSamples);
- void rewind();
+ void rewind();
private:
Common::SeekableReadStream *_stream;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-16 20:00:59 +0000