commit b9d63e21780f167ce06a3ed8d92f2534d0545cec
parent 9e6e95166ac606b05631c0ebacd81e93038f073e
Author: Olav Sørensen <olav.sorensen@live.no>
Date: Wed, 8 Apr 2020 15:25:19 +0200
Pushed v1.19 code
- The "Trim" function had been 100% broken (and crashes the program) for a
very long time. Sorry for this! This happened after instruments were
changed to be allocated as needed.
- macOS: 48kHz is now the default audio output rate (instead of 44.1kHz)
- For x86_64/amd64 users: The audio channel mixer now has higher fractional
precision on mixing deltas, which is beneficial for 96kHz audio frequency
mode.
Diffstat:
22 files changed, 183 insertions(+), 399 deletions(-)
diff --git a/HOW-TO-COMPILE.txt b/HOW-TO-COMPILE.txt
@@ -24,10 +24,6 @@ Compiled Windows/macOS binaries are always available at 16-bits.org/ft2.php
Note: If you don't have libstdc++ and/or can't compile rtmidi, try running
make-linux-nomidi.sh instead.
-
- Note: If you want faster audio mixing (for SLOW devices), pass -DLERPMIX
- to the GCC command line (edit make-linux.sh/make-linux-nomidi.sh). This
- will lower the resampling interpolation quality, though...
Known issues: Audio recording (sampling) can update VERY slowly or not work at
all... I have no idea why, it works really well on Windows/maCOS.
diff --git a/make-linux-nomidi.sh b/make-linux-nomidi.sh
@@ -3,10 +3,6 @@
rm release/other/ft2-clone &> /dev/null
echo Compiling \(with no MIDI functionality\), please wait patiently...
-# If you're compiling for *SLOW* devices, try adding -DLERPMIX right after gcc
-# This will activate 2-tap linear interpolation mixing (blurrier sound) instead
-# of 3-tap quadratic interpolation mixing (sharper sound)
-
gcc -DNDEBUG src/gfxdata/*.c src/*.c -lSDL2 -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone
rm src/gfxdata/*.o src/*.o &> /dev/null
diff --git a/make-linux.sh b/make-linux.sh
@@ -3,10 +3,6 @@
rm release/other/ft2-clone &> /dev/null
echo Compiling, please wait patiently...
-# If you're compiling for *SLOW* devices, try adding -DLERPMIX right after gcc
-# This will activate 2-tap linear interpolation mixing (blurrier sound) instead
-# of 3-tap quadratic interpolation mixing (sharper sound)
-
gcc -DNDEBUG -DHAS_MIDI -D__LINUX_ALSA__ src/rtmidi/*.cpp src/gfxdata/*.c src/*.c -lSDL2 -lpthread -lasound -lstdc++ -lm -Wshadow -Winit-self -Wall -Wno-missing-field-initializers -Wno-unused-result -Wno-strict-aliasing -Wextra -Wunused -Wunreachable-code -Wswitch-default -march=native -mtune=native -O3 -o release/other/ft2-clone
rm src/rtmidi/*.o src/gfxdata/*.o src/*.o &> /dev/null
diff --git a/src/ft2_audio.c b/src/ft2_audio.c
@@ -20,23 +20,30 @@
static int8_t pmpCountDiv, pmpChannels = 2;
static uint16_t smpBuffSize;
static int32_t masterVol, oldAudioFreq, speedVal, pmpLeft, randSeed = INITIAL_DITHER_SEED;
-static int32_t prngStateL, prngStateR, oldPeriod;
-static uint32_t tickTimeLen, tickTimeLenFrac, oldSFrq, oldSFrqRev;
+static int32_t prngStateL, prngStateR;
+static uint32_t tickTimeLen, tickTimeLenFrac;
static float fAudioAmpMul;
static voice_t voice[MAX_VOICES * 2];
static void (*sendAudSamplesFunc)(uint8_t *, uint32_t, uint8_t); // "send mixed samples" routines
+#if !defined __amd64__ && !defined _WIN64
+static int32_t oldPeriod;
+static uint32_t oldSFrq, oldSFrqRev;
+#endif
+
pattSyncData_t *pattSyncEntry;
chSyncData_t *chSyncEntry;
volatile bool pattQueueReading, pattQueueClearing, chQueueReading, chQueueClearing;
+#if !defined __amd64__ && !defined _WIN64
void resetCachedMixerVars(void)
{
oldPeriod = -1;
oldSFrq = 0;
oldSFrqRev = 0xFFFFFFFF;
}
+#endif
void stopVoice(uint8_t i)
{
@@ -367,10 +374,14 @@ void mix_UpdateChannelVolPanFrq(void)
if (status & (IS_Vol + IS_Pan))
voiceUpdateVolumes(i, status);
- // frequency change
+ // frequency change (received even if the period didn't change!)
if (status & IS_Period)
{
- if (ch->finalPeriod != oldPeriod) // this value will very often be the same as before
+#if defined __amd64__ || defined _WIN64
+ v->SFrq = getFrequenceValue(ch->finalPeriod);
+#else
+ // use cached values to prevent a 32-bit divsion all the time
+ if (ch->finalPeriod != oldPeriod)
{
oldPeriod = ch->finalPeriod;
@@ -383,6 +394,7 @@ void mix_UpdateChannelVolPanFrq(void)
v->SFrq = oldSFrq;
v->SFrqRev = oldSFrqRev;
+#endif
}
// sample trigger (note)
@@ -1098,11 +1110,7 @@ bool setupAudio(bool showErrorMsg)
if (config.audioFreq < MIN_AUDIO_FREQ || config.audioFreq > MAX_AUDIO_FREQ)
{
// set default rate
-#ifdef __APPLE__
- config.audioFreq = 44100;
-#else
config.audioFreq = 48000;
-#endif
}
// get audio buffer size from config special flags
diff --git a/src/ft2_audio.h b/src/ft2_audio.h
@@ -8,9 +8,27 @@
enum
{
FREQ_TABLE_LINEAR = 0,
- FREQ_TABLE_AMIGA = 1,
+ FREQ_TABLE_AMIGA = 1,
};
+/* Warning: MIXER_FRAC_BITS must NOT be higher than 22!
+** This can create an overflow in certain calculations.
+**
+** Use 16 on non-x86_64 platforms so that we can avoid a
+** 64-bit division in the outside mixer loop. x86_64 users
+** are lucky and will get higher fractional delta precision.
+** This is beneficial in 96kHz mode, where deltas are lower
+** in value.
+*/
+#if defined __amd64__ || defined _WIN64
+#define MIXER_FRAC_BITS 22
+#else
+#define MIXER_FRAC_BITS 16
+#endif
+
+#define MIXER_FRAC_SCALE (1L << MIXER_FRAC_BITS)
+#define MIXER_FRAC_MASK (MIXER_FRAC_SCALE-1)
+
// for audio/video sync queue. (2^n-1 - don't change this! Queue buffer is already ~2.7MB in size)
#define SYNC_QUEUE_LEN 4095
@@ -19,8 +37,6 @@ enum
#define MIN_AUDIO_FREQ 44100
#define MAX_AUDIO_FREQ 96000
-#define CUBIC_TABLE_LEN (8192+1)
-
struct audio_t
{
char *currInputDevice, *currOutputDevice, *lastWorkingAudioDeviceName;
@@ -47,7 +63,12 @@ typedef struct
uint16_t SVol;
int32_t SLVol1, SRVol1, SLVol2, SRVol2, SLVolIP, SRVolIP;
int32_t SPos, SLen, SRepS, SRepL;
- uint32_t SVolIPLen, SPosDec, SFrq, SFrqRev;
+ uint32_t SVolIPLen, SPosDec, SFrq;
+
+#if !defined __amd64__ && !defined _WIN64
+ uint32_t SFrqRev;
+#endif
+
void (*mixRoutine)(void *, int32_t); // function pointer to mix routine
} voice_t;
@@ -80,7 +101,10 @@ extern chSyncData_t *chSyncEntry;
extern volatile bool pattQueueReading, pattQueueClearing, chQueueReading, chQueueClearing;
+#if !defined __amd64__ && !defined _WIN64
void resetCachedMixerVars(void);
+#endif
+
int32_t pattQueueReadSize(void);
int32_t pattQueueWriteSize(void);
bool pattQueuePush(pattSyncData_t t);
diff --git a/src/ft2_config.c b/src/ft2_config.c
@@ -71,7 +71,7 @@ static int32_t calcChecksum(uint8_t *p, uint16_t len) // for nibbles highscore d
return checksum;
}
-static void loadConfigFromBuffer(uint8_t defaults)
+static void loadConfigFromBuffer(void)
{
int32_t i, checksum;
@@ -79,13 +79,6 @@ static void loadConfigFromBuffer(uint8_t defaults)
memcpy(&config, configBuffer, CONFIG_FILE_SIZE);
-#ifdef __APPLE__
- if (defaults)
- config.audioFreq = 44100;
-#else
- (void)defaults; // prevent warning
-#endif
-
// if Nibbles highscore table checksum is incorrect, load default highscore table instead
checksum = calcChecksum((uint8_t *)&config.NI_HighScore, sizeof (config.NI_HighScore));
if (config.NI_HighScoreChecksum != checksum)
@@ -143,23 +136,10 @@ static void loadConfigFromBuffer(uint8_t defaults)
}
if (config.audioFreq != 44100 && config.audioFreq != 48000 && config.audioFreq != 96000)
- {
- // set default
-#ifdef __APPLE__
- config.audioFreq = 44100;
-#else
config.audioFreq = 48000;
-#endif
- }
if (config.audioInputFreq <= 1) // default value from FT2 (this was cdr_Sync) - set defaults
- {
-#ifdef __APPLE__
- config.audioInputFreq = INPUT_FREQ_44KHZ;
-#else
config.audioInputFreq = INPUT_FREQ_48KHZ;
-#endif
- }
if (config.specialFlags == 64) // default value from FT2 (this was ptnDefaultLen byte #1) - set defaults
config.specialFlags = BUFFSIZE_1024 | BITDEPTH_16;
@@ -204,7 +184,7 @@ static void configDrawAmp(void)
static void setDefaultConfigSettings(void)
{
memcpy(configBuffer, defConfigData, CONFIG_FILE_SIZE);
- loadConfigFromBuffer(true);
+ loadConfigFromBuffer();
}
void resetConfig(void)
@@ -344,7 +324,7 @@ bool loadConfig(bool showErrorFlag)
return false;
}
- loadConfigFromBuffer(false);
+ loadConfigFromBuffer();
return true;
}
@@ -736,7 +716,7 @@ void loadConfigOrSetDefaults(void)
return;
}
- loadConfigFromBuffer(false);
+ loadConfigFromBuffer();
}
static void drawQuantValue(void)
@@ -833,13 +813,8 @@ void setConfigIORadioButtonStates(void) // accessed by other .c files
uncheckRadioButtonGroup(RB_GROUP_CONFIG_AUDIO_FREQ);
switch (config.audioFreq)
{
-#ifdef __APPLE__
- default: case 44100: tmpID = RB_CONFIG_AUDIO_44KHZ; break;
- case 48000: tmpID = RB_CONFIG_AUDIO_48KHZ; break;
-#else
case 44100: tmpID = RB_CONFIG_AUDIO_44KHZ; break;
default: case 48000: tmpID = RB_CONFIG_AUDIO_48KHZ; break;
-#endif
case 96000: tmpID = RB_CONFIG_AUDIO_96KHZ; break;
}
radioButtons[tmpID].state = RADIOBUTTON_CHECKED;
@@ -848,13 +823,8 @@ void setConfigIORadioButtonStates(void) // accessed by other .c files
uncheckRadioButtonGroup(RB_GROUP_CONFIG_AUDIO_INPUT_FREQ);
switch (config.audioInputFreq)
{
-#ifdef __APPLE__
- default: case INPUT_FREQ_44KHZ: tmpID = RB_CONFIG_AUDIO_INPUT_44KHZ; break;
- case INPUT_FREQ_48KHZ: tmpID = RB_CONFIG_AUDIO_INPUT_48KHZ; break;
-#else
case INPUT_FREQ_44KHZ: tmpID = RB_CONFIG_AUDIO_INPUT_44KHZ; break;
default: case INPUT_FREQ_48KHZ: tmpID = RB_CONFIG_AUDIO_INPUT_48KHZ; break;
-#endif
case INPUT_FREQ_96KHZ: tmpID = RB_CONFIG_AUDIO_INPUT_96KHZ; break;
}
radioButtons[tmpID].state = RADIOBUTTON_CHECKED;
@@ -1172,13 +1142,8 @@ void showConfigScreen(void)
textOutShadow(406, 160, PAL_FORGRND, PAL_DSKTOP2, "1-bit dither");
textOutShadow(509, 3, PAL_FORGRND, PAL_DSKTOP2, "Mixing frequency:");
-#ifdef __APPLE__
- textOutShadow(525, 17, PAL_FORGRND, PAL_DSKTOP2, "44100Hz (default)");
- textOutShadow(525, 31, PAL_FORGRND, PAL_DSKTOP2, "48000Hz");
-#else
textOutShadow(525, 17, PAL_FORGRND, PAL_DSKTOP2, "44100Hz");
textOutShadow(525, 31, PAL_FORGRND, PAL_DSKTOP2, "48000Hz (default)");
-#endif
textOutShadow(525, 45, PAL_FORGRND, PAL_DSKTOP2, "96000Hz");
textOutShadow(509, 76, PAL_FORGRND, PAL_DSKTOP2, "Frequency table:");
diff --git a/src/ft2_header.h b/src/ft2_header.h
@@ -12,7 +12,7 @@
#endif
#include "ft2_replayer.h"
-#define PROG_VER_STR "1.18"
+#define PROG_VER_STR "1.19"
// do NOT change these! It will only mess things up...
diff --git a/src/ft2_intrp_table.h b/src/ft2_intrp_table.h
@@ -2,15 +2,14 @@
#include <stdint.h>
-// number of bits for fractional sample position in audio mixer
-#define FRAC_BITS 16
+#include "ft2_audio.h"
#define CUBIC_WIDTH 4
#define CUBIC_WIDTH_BITS 2
#define CUBIC_PHASES 4096
#define CUBIC_PHASES_BITS 12
-#define CUBIC_FSHIFT (FRAC_BITS-(CUBIC_PHASES_BITS+CUBIC_WIDTH_BITS))
+#define CUBIC_FSHIFT (MIXER_FRAC_BITS-(CUBIC_PHASES_BITS+CUBIC_WIDTH_BITS))
#define CUBIC_FMASK ((CUBIC_WIDTH*CUBIC_PHASES)-CUBIC_WIDTH)
#define CUBIC_QUANTSHIFT 15
diff --git a/src/ft2_main.c b/src/ft2_main.c
@@ -179,11 +179,8 @@ int main(int argc, char *argv[])
if (!setupAudio(CONFIG_HIDE_ERRORS))
{
// one LAST attempt (with default audio device and settings)
-#ifdef __APPLE__
- config.audioFreq = 44100;
-#else
config.audioFreq = 48000;
-#endif
+
// try 16-bit audio at 1024 samples (44.1kHz/48kHz)
config.specialFlags &= ~(BITDEPTH_24 + BUFFSIZE_512 + BUFFSIZE_2048);
config.specialFlags |= (BITDEPTH_16 + BUFFSIZE_1024);
diff --git a/src/ft2_mix.c b/src/ft2_mix.c
@@ -227,12 +227,9 @@ static void mix8bBidiLoop(voice_t *v, uint32_t numSamples)
static void mix8bNoLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -293,12 +290,9 @@ static void mix8bNoLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix8bLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -359,12 +353,9 @@ static void mix8bLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix8bBidiLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *CDA_LinAdrRev, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, delta, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -639,12 +630,9 @@ static void mix8bRampBidiLoop(voice_t *v, uint32_t numSamples)
static void mix8bRampNoLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
@@ -712,12 +700,9 @@ static void mix8bRampNoLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix8bRampLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
@@ -785,12 +770,9 @@ static void mix8bRampLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix8bRampBidiLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int8_t *CDA_LinearAdr, *CDA_LinAdrRev, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, delta, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
@@ -1056,12 +1038,9 @@ static void mix16bBidiLoop(voice_t *v, uint32_t numSamples)
static void mix16bNoLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -1122,12 +1101,9 @@ static void mix16bNoLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix16bLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -1188,12 +1164,9 @@ static void mix16bLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix16bBidiLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *CDA_LinAdrRev, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVol, CDA_RVol;
uint32_t pos, delta, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
GET_VOL
@@ -1468,12 +1441,9 @@ static void mix16bRampBidiLoop(voice_t *v, uint32_t numSamples)
static void mix16bRampNoLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
@@ -1541,12 +1511,9 @@ static void mix16bRampNoLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix16bRampLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
@@ -1614,12 +1581,9 @@ static void mix16bRampLoopIntrp(voice_t *v, uint32_t numSamples)
static void mix16bRampBidiLoopIntrp(voice_t *v, uint32_t numSamples)
{
const int16_t *CDA_LinearAdr, *CDA_LinAdrRev, *smpPtr;
- int32_t realPos, sample, sample2, *audioMixL, *audioMixR;
+ int32_t realPos, sample, sample2, sample3, sample4, *audioMixL, *audioMixR;
int32_t CDA_LVolIP, CDA_RVolIP, CDA_LVol, CDA_RVol;
uint32_t pos, delta, i, samplesToMix, CDA_BytesLeft;
-#ifndef LERPMIX
- int32_t sample3, sample4;
-#endif
if ((v->SLVol1 | v->SRVol1 | v->SLVol2 | v->SRVol2) == 0)
{
diff --git a/src/ft2_mix_macros.h b/src/ft2_mix_macros.h
@@ -1,6 +1,7 @@
#pragma once
#include "ft2_header.h"
+#include "ft2_audio.h"
/* ----------------------------------------------------------------------- */
/* GENERAL MIXER MACROS */
@@ -14,9 +15,30 @@
v->SLVol2 = CDA_LVol; \
v->SRVol2 = CDA_RVol; \
+#if defined __amd64__ || defined _WIN64
+
+#define GET_MIXER_VARS \
+ const uint32_t SFrq = v->SFrq; \
+ audioMixL = audio.mixBufferL; \
+ audioMixR = audio.mixBufferR; \
+ const bool mixInMono = (CDA_LVol == CDA_RVol); \
+ realPos = v->SPos; \
+ pos = v->SPosDec; \
+
+#define GET_MIXER_VARS_RAMP \
+ const uint32_t SFrq = v->SFrq; \
+ audioMixL = audio.mixBufferL; \
+ audioMixR = audio.mixBufferR; \
+ CDA_LVolIP = v->SLVolIP; \
+ CDA_RVolIP = v->SRVolIP; \
+ const bool mixInMono = (v->SLVol2 == v->SRVol2) && (CDA_LVolIP == CDA_RVolIP); \
+ realPos = v->SPos; \
+ pos = v->SPosDec; \
+
+#else
+
#define GET_MIXER_VARS \
const uint32_t SFrq = v->SFrq; \
- const uint32_t SFrqRev = v->SFrqRev; \
audioMixL = audio.mixBufferL; \
audioMixR = audio.mixBufferR; \
const bool mixInMono = (CDA_LVol == CDA_RVol); \
@@ -25,7 +47,6 @@
#define GET_MIXER_VARS_RAMP \
const uint32_t SFrq = v->SFrq; \
- const uint32_t SFrqRev = v->SFrqRev; \
audioMixL = audio.mixBufferL; \
audioMixR = audio.mixBufferR; \
CDA_LVolIP = v->SLVolIP; \
@@ -34,6 +55,8 @@
realPos = v->SPos; \
pos = v->SPosDec; \
+#endif
+
#define SET_BASE8 \
CDA_LinearAdr = v->SBase8; \
smpPtr = CDA_LinearAdr + realPos; \
@@ -52,14 +75,14 @@
#define INC_POS \
pos += SFrq; \
- smpPtr += pos >> 16; \
- pos &= 0xFFFF; \
+ smpPtr += pos >> MIXER_FRAC_BITS; \
+ pos &= MIXER_FRAC_MASK; \
#define INC_POS_BIDI \
pos += CDA_IPValL; \
- smpPtr += pos >> 16; \
+ smpPtr += pos >> MIXER_FRAC_BITS; \
smpPtr += CDA_IPValH; \
- pos &= 0xFFFF; \
+ pos &= MIXER_FRAC_MASK; \
#define SET_BACK_MIXER_POS \
v->SPosDec = pos; \
@@ -101,8 +124,6 @@
*audioMixL++ += sample; \
*audioMixR++ += sample; \
-#ifndef LERPMIX
-
// 4-tap cubic spline interpolation (default - slower than linear, but better quality)
// in: int32_t s0,s1,s2,s3 = -128..127 | f = 0..65535 (frac) | out: 16-bit s0 (will exceed 16-bits because of overshoot)
@@ -175,84 +196,41 @@
*audioMixL++ += sample; \
*audioMixR++ += sample; \
-#else
-
-// 2-tap linear interpolation (like FT2 - faster, but bad quality)
-
-// in: int32_t s1,s2 = -128..127 | f = 0..65535 (frac) | out: s1 = -32768..32767
-#define INTERPOLATE8(s1, s2, f) \
- s2 -= s1; \
- s2 = (s2 * (int32_t)f) >> (16 - 8); \
- s1 <<= 8; \
- s1 += s2; \
-
-// in: int32_t s1,s2 = -32768..32767 | f = 0..65535 (frac) | out: s1 = -32768..32767
-#define INTERPOLATE16(s1, s2, f) \
- s2 = (s2 - s1) >> 1; \
- s2 = (s2 * (int32_t)f) >> (16 - 1); \
- s1 += s2; \
-
-/* 8bitbubsy: It may look like we are potentially going out of bounds by looking up sample point
-** 1, but the sample data is actually padded on the right side, where a correct sample is stored
-** according to loop mode (or no loop). This is how original FT2 (MS-DOS) does it as well.
-*/
-
-#define RENDER_8BIT_SMP_INTRP \
- assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = smpPtr[0]; \
- sample2 = smpPtr[1]; \
- INTERPOLATE8(sample, sample2, pos) \
- sample <<= 12; \
- *audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
- *audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
-
-#define RENDER_8BIT_SMP_MONO_INTRP \
- assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = smpPtr[0]; \
- sample2 = smpPtr[1]; \
- INTERPOLATE8(sample, sample2, pos) \
- sample <<= 12; \
- sample = ((int64_t)sample * CDA_LVol) >> 32; \
- *audioMixL++ += sample; \
- *audioMixR++ += sample; \
-
-#define RENDER_16BIT_SMP_INTRP \
- assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = smpPtr[0]; \
- sample2 = smpPtr[1]; \
- INTERPOLATE16(sample, sample2, pos) \
- sample <<= 12; \
- *audioMixL++ += ((int64_t)sample * CDA_LVol) >> 32; \
- *audioMixR++ += ((int64_t)sample * CDA_RVol) >> 32; \
-
-#define RENDER_16BIT_SMP_MONO_INTRP \
- assert(smpPtr >= CDA_LinearAdr && smpPtr < CDA_LinearAdr+v->SLen); \
- sample = smpPtr[0]; \
- sample2 = smpPtr[1]; \
- INTERPOLATE16(sample, sample2, pos) \
- sample <<= 12; \
- sample = ((int64_t)sample * CDA_LVol) >> 32; \
- *audioMixL++ += sample; \
- *audioMixR++ += sample; \
-
-#endif
-
/* ----------------------------------------------------------------------- */
/* SAMPLES-TO-MIX LIMITING MACROS */
/* ----------------------------------------------------------------------- */
+#if defined __amd64__ || defined _WIN64
+
#define LIMIT_MIX_NUM \
+ samplesToMix = UINT32_MAX; \
i = (v->SLen - 1) - realPos; \
- if (i > 65535) \
- i = 65535; \
\
- i = (i << 16) | (pos ^ 0xFFFF); \
- samplesToMix = ((int64_t)i * SFrqRev) >> 32; \
+ if (SFrq > i>>MIXER_FRAC_BITS) \
+ { \
+ const uint64_t tmp64 = ((uint64_t)i << MIXER_FRAC_BITS) | (pos ^ MIXER_FRAC_MASK); \
+ samplesToMix = (uint32_t)(tmp64 / SFrq); \
+ samplesToMix++; \
+ } \
+ \
+ if (samplesToMix > CDA_BytesLeft) \
+ samplesToMix = CDA_BytesLeft; \
+
+#else
+
+#define LIMIT_MIX_NUM \
+ i = (v->SLen - 1) - realPos; \
+ if (i > (1UL << (32-MIXER_FRAC_BITS))) \
+ i = 1UL << (32-MIXER_FRAC_BITS); \
+ \
+ i = (i << MIXER_FRAC_BITS) | (pos ^ MIXER_FRAC_MASK); \
+ samplesToMix = ((int64_t)i * v->SFrqRev) >> 32; \
samplesToMix++; \
\
if (samplesToMix > CDA_BytesLeft) \
samplesToMix = CDA_BytesLeft; \
+#endif
#define LIMIT_MIX_NUM_RAMP \
if (v->SVolIPLen == 0) \
@@ -302,7 +280,7 @@
assert(realPos >= v->SRepS && realPos < v->SLen); \
realPos = ~realPos; \
smpPtr = CDA_LinAdrRev + realPos; \
- pos ^= 0xFFFF; \
+ pos ^= MIXER_FRAC_MASK; \
} \
else \
{ \
@@ -311,13 +289,13 @@
smpPtr = CDA_LinearAdr + realPos; \
} \
\
- const int32_t CDA_IPValH = (int32_t)delta >> 16; \
- const int32_t CDA_IPValL = delta & 0xFFFF; \
+ const int32_t CDA_IPValH = (int32_t)delta >> MIXER_FRAC_BITS; \
+ const int32_t CDA_IPValL = delta & MIXER_FRAC_MASK; \
#define END_BIDI \
if (v->backwards) \
{ \
- pos ^= 0xFFFF; \
+ pos ^= MIXER_FRAC_MASK; \
realPos = ~(int32_t)(smpPtr - CDA_LinAdrRev); \
} \
else \
@@ -330,44 +308,38 @@
/* VOLUME=0 OPTIMIZATION MACROS */
/* ----------------------------------------------------------------------- */
-#define VOL0_OPTIMIZATION_NO_LOOP \
- assert(numSamples <= 65536); \
- \
- pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
- realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
+#define VOL0_INC_POS \
+ const uint64_t newPos = (uint64_t)v->SFrq * numSamples; \
+ const uint32_t addPos = (uint32_t)(newPos >> MIXER_FRAC_BITS); \
+ uint32_t addFrac = (uint32_t)(newPos & MIXER_FRAC_MASK); \
\
+ addFrac += v->SPosDec; \
+ realPos = v->SPos + addPos + (addFrac >> MIXER_FRAC_BITS); \
+ pos = addFrac & MIXER_FRAC_MASK; \
+
+#define VOL0_OPTIMIZATION_NO_LOOP \
+ VOL0_INC_POS \
if (realPos >= v->SLen) \
{ \
v->mixRoutine = NULL; /* shut down voice */ \
return; \
} \
\
- pos &= 0xFFFF; \
SET_BACK_MIXER_POS
#define VOL0_OPTIMIZATION_LOOP \
- assert(numSamples <= 65536); \
- \
- pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
- realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
- \
+ VOL0_INC_POS \
while (realPos >= v->SLen) \
realPos -= v->SRepL; \
\
- pos &= 0xFFFF; \
SET_BACK_MIXER_POS
#define VOL0_OPTIMIZATION_BIDI_LOOP \
- assert(numSamples <= 65536); \
- \
- pos = v->SPosDec + ((v->SFrq & 0xFFFF) * numSamples); \
- realPos = v->SPos + ((v->SFrq >> 16) * numSamples) + (pos >> 16); \
- \
+ VOL0_INC_POS \
while (realPos >= v->SLen) \
{ \
realPos -= v->SRepL; \
v->backwards ^= 1; \
} \
\
- pos &= 0xFFFF; \
SET_BACK_MIXER_POS
diff --git a/src/ft2_radiobuttons.c b/src/ft2_radiobuttons.c
@@ -85,13 +85,8 @@ radioButton_t radioButtons[NUM_RADIOBUTTONS] =
// audio output frequency
//x, y, w, group, funcOnUp
-#ifdef __APPLE__
- { 509, 16, 121, RB_GROUP_CONFIG_AUDIO_FREQ, rbConfigAudio44kHz },
- { 509, 30, 66, RB_GROUP_CONFIG_AUDIO_FREQ, rbConfigAudio48kHz },
-#else
{ 509, 16, 66, RB_GROUP_CONFIG_AUDIO_FREQ, rbConfigAudio44kHz },
{ 509, 30, 121, RB_GROUP_CONFIG_AUDIO_FREQ, rbConfigAudio48kHz },
-#endif
{ 509, 44, 66, RB_GROUP_CONFIG_AUDIO_FREQ, rbConfigAudio96kHz },
// audio input frequency
diff --git a/src/ft2_replayer.c b/src/ft2_replayer.c
@@ -26,8 +26,9 @@
*/
static bool bxxOverflow;
-static int32_t oldPeriod, oldRate;
-static uint32_t frequenceDivFactor, frequenceMulFactor;
+static int32_t oldPeriod;
+static uint32_t period2DeltaTab[768][32], oldDelta;
+static double dAmigaPeriodDiv;
static tonTyp nilPatternLine;
// globally accessed
@@ -320,68 +321,78 @@ void keyOff(stmTyp *ch)
}
}
-void calcReplayRate(int32_t rate) // 100% FT2-accurate routine, do not touch!
+void calcReplayRate(int32_t rate)
{
if (rate == 0)
return;
- // the following calculations are 100% accurate to FT2, do not touch!
- frequenceDivFactor = (int32_t)round(65536.0 * 1712.0 / rate * 8363.0);
- frequenceMulFactor = (int32_t)round(256.0 * 65536.0 / rate * 8363.0);
- audio.quickVolSizeVal = rate / 200;
+ const double dRateFactor = (double)MIXER_FRAC_SCALE / rate;
- // the following are non-FT2 calculations
+ // generate period-to-delta table
+ const double dMul = dRateFactor * (8363.0 * 256.0);
+ for (int32_t i = 0; i < 768; i++)
+ {
+ const double dHz = exp2(i * (1.0 / 768.0)) * dMul;
+ for (int32_t j = 0; j < 32; j++)
+ {
+ const double dOut = dHz * exp2(-j);
+ period2DeltaTab[i][j] = (int32_t)(dOut + 0.5);
+ }
+ }
+
+ dAmigaPeriodDiv = dRateFactor * (8363.0 * 1712.0);
+
+ audio.quickVolSizeVal = rate / 200; // FT2 truncates here
audio.rampQuickVolMul = (int32_t)round((UINT32_MAX + 1.0) / audio.quickVolSizeVal);
audio.dSpeedValMul = editor.dPerfFreq / rate; // for audio/video sync
- const uint32_t deltaBase = frequenceDivFactor / (1712 * 16); // exact 16.16 delta base
- audio.dPianoDeltaMul = 1.0 / deltaBase; // for piano in Instr. Ed.
+ // exact integer fixed-point delta base for piano in Instr. Ed.
+ int32_t deltaBase = (int32_t)round(dAmigaPeriodDiv / (1712 * 16));
+ audio.dPianoDeltaMul = 1.0 / deltaBase;
}
-// 100% FT2-accurate routine, do not touch!
uint32_t getFrequenceValue(uint16_t period)
{
- uint8_t shift;
- uint16_t index;
- int32_t indexQuotient, indexRemainder;
- uint32_t rate;
+ uint32_t delta;
if (period == 0)
return 0;
if (period == oldPeriod)
- return oldRate; // added check: prevent this calculation if it would yield the same
+ return oldDelta; // we have already calculated this delta
if (linearFrqTab)
{
- index = (12 * 192 * 4) - period;
- indexQuotient = index / 768;
- indexRemainder = index % 768;
+ const uint16_t invPeriod = (12 * 192 * 4) - period; // this intentionally overflows uint16_t to be accurate to FT2
- rate = ((int64_t)logTab[indexRemainder] * frequenceMulFactor) >> LOG_TABLE_BITS;
+ const int32_t quotient = invPeriod / 768;
+ const int32_t remainder = invPeriod % 768;
+
+ const int32_t octave = (14 - quotient) & 0x1F; // this is accurate to FT2 (it can go crazy on very high periods)
- shift = (14 - indexQuotient) & 0x1F;
- if (shift != 0)
- rate >>= shift;
+ delta = period2DeltaTab[remainder][octave];
}
else
{
- rate = frequenceDivFactor / period;
+ const double dHz = dAmigaPeriodDiv / period;
+ delta = (int32_t)(dHz + 0.5); // rounded (don't cast to uint32_t as it will avoid SSE2 usage, and delta is <= 2^31 anyway)
}
oldPeriod = period;
- oldRate = rate;
+ oldDelta = delta;
- return rate;
+ return delta;
}
void resetCachedFrequencyVars(void)
{
oldPeriod = -1;
- oldRate = 0;
+ oldDelta = 0;
resetCachedScopeVars();
+#if !defined __amd64__ && !defined _WIN64
resetCachedMixerVars();
+#endif
}
static void startTone(uint8_t ton, uint8_t effTyp, uint8_t eff, stmTyp *ch)
diff --git a/src/ft2_replayer.h b/src/ft2_replayer.h
@@ -166,10 +166,7 @@ typedef struct sampleTyp_t
bool fixed;
int8_t fine, relTon, *pek, *origPek;
uint8_t vol, typ, pan;
- int16_t fixedSmp1;
-#ifndef LERPMIX
- int16_t fixedSmp2;
-#endif
+ int16_t fixedSmp1, fixedSmp2;
int32_t fixedPos, len, repS, repL;
} sampleTyp;
diff --git a/src/ft2_sample_ed.c b/src/ft2_sample_ed.c
@@ -81,9 +81,7 @@ void fixSample(sampleTyp *s)
// write new values
ptr16[-1] = 0;
ptr16[len+0] = 0;
-#ifndef LERPMIX
ptr16[len+1] = 0;
-#endif
}
else
{
@@ -93,9 +91,7 @@ void fixSample(sampleTyp *s)
// write new values
s->pek[-1] = 0;
s->pek[len+0] = 0;
-#ifndef LERPMIX
s->pek[len+1] = 0;
-#endif
}
return;
@@ -123,17 +119,14 @@ void fixSample(sampleTyp *s)
// store old fix position and old values
s->fixedPos = s->repS + s->repL;
s->fixedSmp1 = ptr16[loopEnd+0];
-#ifndef LERPMIX
s->fixedSmp2 = ptr16[loopEnd+1];
-#endif
+
// write new values
ptr16[loopEnd+0] = ptr16[loopStart+0];
-#ifndef LERPMIX
if (loopStart == 0 && loopEnd > 0)
ptr16[-1] = ptr16[loopEnd-1];
ptr16[loopEnd+1] = ptr16[loopStart+1];
-#endif
}
else
{
@@ -148,17 +141,14 @@ void fixSample(sampleTyp *s)
// store old fix position and old values
s->fixedPos = loopEnd;
s->fixedSmp1 = s->pek[loopEnd+0];
-#ifndef LERPMIX
s->fixedSmp2 = s->pek[loopEnd+1];
-#endif
+
// write new values
s->pek[loopEnd+0] = s->pek[loopStart+0];
-#ifndef LERPMIX
if (loopStart == 0 && loopEnd > 0)
s->pek[-1] = s->pek[loopEnd-1];
s->pek[loopEnd+1] = s->pek[loopStart+1];
-#endif
}
}
else
@@ -181,12 +171,10 @@ void fixSample(sampleTyp *s)
// store old fix position and old values
s->fixedPos = s->repS + s->repL;
s->fixedSmp1 = ptr16[loopEnd+0];
-#ifndef LERPMIX
s->fixedSmp2 = ptr16[loopEnd+1];
-#endif
+
// write new values
ptr16[loopEnd+0] = ptr16[loopEnd-1];
-#ifndef LERPMIX
if (loopStart == 0)
ptr16[-1] = ptr16[0];
@@ -194,7 +182,6 @@ void fixSample(sampleTyp *s)
ptr16[loopEnd+1] = ptr16[loopEnd-2];
else
ptr16[loopEnd+1] = ptr16[loopStart+0];
-#endif
}
else
{
@@ -211,12 +198,10 @@ void fixSample(sampleTyp *s)
// store old fix position and old values
s->fixedPos = loopEnd;
s->fixedSmp1 = s->pek[loopEnd+0];
-#ifndef LERPMIX
s->fixedSmp2 = s->pek[loopEnd+1];
-#endif
+
// write new values
s->pek[loopEnd+0] = s->pek[loopEnd-1];
-#ifndef LERPMIX
if (loopStart == 0)
s->pek[-1] = s->pek[0];
@@ -224,7 +209,6 @@ void fixSample(sampleTyp *s)
s->pek[loopEnd+1] = s->pek[loopEnd-2];
else
s->pek[loopEnd+1] = s->pek[loopStart+0];
-#endif
}
}
@@ -258,18 +242,14 @@ void restoreSample(sampleTyp *s)
fixedPos16 = s->fixedPos >> 1;
ptr16[fixedPos16+0] = s->fixedSmp1;
-#ifndef LERPMIX
ptr16[fixedPos16+1] = s->fixedSmp2;
-#endif
}
else
{
// 8-bit sample
s->pek[s->fixedPos+0] = (int8_t)s->fixedSmp1;
-#ifndef LERPMIX
s->pek[s->fixedPos+1] = (int8_t)s->fixedSmp2;
-#endif
}
}
@@ -646,10 +626,8 @@ static int8_t getScaledSample(sampleTyp *s, int32_t index)
{
if (index == s->fixedPos)
tmp32 = s->fixedSmp1;
-#ifndef LERPMIX
else if (index == s->fixedPos+2)
tmp32 = s->fixedSmp2;
-#endif
else
tmp32 = ptr16[index >> 1];
}
@@ -669,10 +647,8 @@ static int8_t getScaledSample(sampleTyp *s, int32_t index)
{
if (index == s->fixedPos)
tmp32 = s->fixedSmp1;
-#ifndef LERPMIX
else if (index == s->fixedPos+1)
tmp32 = s->fixedSmp2;
-#endif
else
tmp32 = ptr8[index];
}
diff --git a/src/ft2_sample_saver.c b/src/ft2_sample_saver.c
@@ -72,13 +72,11 @@ static bool fileRestoreSampleData(UNICHAR *filenameU, int32_t sampleDataOffset,
fwrite(&smp->fixedSmp1, sizeof (int16_t), 1, f);
}
-#ifndef LERPMIX
if (smp->fixedPos+2 < smp->len/2)
{
fseek(f, sampleDataOffset + ((smp->fixedPos + 2) * 2), SEEK_SET);
fwrite(&smp->fixedSmp2, sizeof (int16_t), 1, f);
}
-#endif
}
else
{
@@ -94,7 +92,6 @@ static bool fileRestoreSampleData(UNICHAR *filenameU, int32_t sampleDataOffset,
fwrite(&fixSpar8, sizeof (int8_t), 1, f);
}
-#ifndef LERPMIX
if (smp->fixedPos+1 < smp->len)
{
fseek(f, sampleDataOffset + (smp->fixedPos + 1), SEEK_SET);
@@ -105,7 +102,6 @@ static bool fileRestoreSampleData(UNICHAR *filenameU, int32_t sampleDataOffset,
fwrite(&fixSpar8, sizeof (int8_t), 1, f);
}
-#endif
}
fclose(f);
diff --git a/src/ft2_sampling.c b/src/ft2_sampling.c
@@ -370,14 +370,8 @@ void startSampling(void)
switch (config.audioInputFreq)
{
case INPUT_FREQ_96KHZ: samplingRate = 96000; break;
-
-#ifdef __APPLE__
- case INPUT_FREQ_48KHZ: samplingRate = 48000; break;
- default: samplingRate = 44100; break;
-#else
case INPUT_FREQ_44KHZ: samplingRate = 44100; break;
default: samplingRate = 48000; break;
-#endif
}
memset(&want, 0, sizeof (SDL_AudioSpec));
diff --git a/src/ft2_scopes.h b/src/ft2_scopes.h
@@ -3,11 +3,12 @@
#include <stdint.h>
#include <stdbool.h>
#include "ft2_header.h"
+#include "ft2_audio.h"
-// log2(65536 / SCOPE_HZ) where SCOPE_HZ is 2^n
-#define SCOPE_FRAC_BITS 22
+// 6 = log2(SCOPE_HZ) where SCOPE_HZ is 2^n
+#define SCOPE_FRAC_BITS (MIXER_FRAC_BITS+6)
-#define SCOPE_FRAC_SCALE (1UL << SCOPE_FRAC_BITS)
+#define SCOPE_FRAC_SCALE (1L << SCOPE_FRAC_BITS)
#define SCOPE_FRAC_MASK (SCOPE_FRAC_SCALE-1)
void resetCachedScopeVars(void);
diff --git a/src/ft2_tables.c b/src/ft2_tables.c
@@ -311,106 +311,6 @@ const int16_t amigaPeriods[1936] = // bit-exact to FT2 table
// the last 17 values are off (but identical to FT2) because of a bug in how FT2 calculates this table
};
-const uint32_t logTab[768] = // bit-exact to FT2 table
-{
- 16777216, 16792365, 16807527, 16822704, 16837894, 16853097, 16868315, 16883546,
- 16898791, 16914049, 16929322, 16944608, 16959908, 16975222, 16990549, 17005891,
- 17021246, 17036615, 17051999, 17067396, 17082806, 17098231, 17113670, 17129123,
- 17144589, 17160070, 17175564, 17191073, 17206595, 17222132, 17237683, 17253247,
- 17268826, 17284419, 17300026, 17315646, 17331282, 17346931, 17362594, 17378271,
- 17393963, 17409669, 17425389, 17441123, 17456871, 17472634, 17488410, 17504202,
- 17520007, 17535826, 17551660, 17567508, 17583371, 17599248, 17615139, 17631044,
- 17646964, 17662898, 17678847, 17694810, 17710787, 17726779, 17742785, 17758806,
- 17774841, 17790891, 17806955, 17823034, 17839127, 17855235, 17871357, 17887494,
- 17903645, 17919811, 17935992, 17952187, 17968397, 17984621, 18000860, 18017114,
- 18033382, 18049665, 18065963, 18082276, 18098603, 18114945, 18131302, 18147673,
- 18164060, 18180461, 18196877, 18213307, 18229753, 18246213, 18262689, 18279179,
- 18295684, 18312204, 18328739, 18345288, 18361853, 18378433, 18395028, 18411637,
- 18428262, 18444902, 18461556, 18478226, 18494911, 18511611, 18528325, 18545056,
- 18561801, 18578561, 18595336, 18612127, 18628932, 18645753, 18662589, 18679441,
- 18696307, 18713189, 18730086, 18746998, 18763925, 18780868, 18797826, 18814800,
- 18831788, 18848792, 18865812, 18882846, 18899897, 18916962, 18934043, 18951139,
- 18968251, 18985378, 19002521, 19019679, 19036853, 19054042, 19071247, 19088467,
- 19105703, 19122954, 19140221, 19157504, 19174802, 19192116, 19209445, 19226790,
- 19244151, 19261527, 19278919, 19296327, 19313750, 19331190, 19348645, 19366115,
- 19383602, 19401104, 19418622, 19436156, 19453706, 19471271, 19488853, 19506450,
- 19524063, 19541692, 19559337, 19576998, 19594675, 19612368, 19630077, 19647802,
- 19665543, 19683300, 19701072, 19718861, 19736666, 19754488, 19772325, 19790178,
- 19808047, 19825933, 19843835, 19861752, 19879686, 19897637, 19915603, 19933586,
- 19951585, 19969600, 19987631, 20005679, 20023743, 20041823, 20059920, 20078033,
- 20096162, 20114308, 20132470, 20150648, 20168843, 20187054, 20205282, 20223526,
- 20241787, 20260064, 20278358, 20296668, 20314995, 20333338, 20351698, 20370074,
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-};
-
/* ----------------------------------------------------------------------- */
/* AUDIO MIXER TABLES */
/* ----------------------------------------------------------------------- */
diff --git a/src/ft2_tables.h b/src/ft2_tables.h
@@ -6,8 +6,6 @@
#include "ft2_header.h" // MAX_VOICES
#include "ft2_config.h" // CONFIG_FILE_SIZE
-#define LOG_TABLE_BITS 24
-
#define KEY2VOL_ENTRIES (signed)(sizeof (key2VolTab) / sizeof (SDL_Keycode))
#define KEY2EFX_ENTRIES (signed)(sizeof (key2EfxTab) / sizeof (SDL_Keycode))
#define KEY2HEX_ENTRIES (signed)(sizeof (key2HexTab) / sizeof (SDL_Keycode))
@@ -18,7 +16,6 @@ extern const uint16_t amigaPeriod[12 * 8];
extern const uint16_t amigaFinePeriod[12 * 8];
extern const int16_t linearPeriods[1936];
extern const int16_t amigaPeriods[1936];
-extern const uint32_t logTab[768];
extern const uint32_t panningTab[257];
diff --git a/src/ft2_trim.c b/src/ft2_trim.c
@@ -31,7 +31,7 @@ void pbTrimCalc(void);
static void freeTmpInstruments(void)
{
- for (int16_t i = 0; i <= MAX_INST; i++)
+ for (int32_t i = 0; i <= MAX_INST; i++)
{
if (tmpInstr[i] != NULL)
{
@@ -47,7 +47,7 @@ static bool setTmpInstruments(void)
{
freeTmpInstruments();
- for (int16_t i = 0; i <= MAX_INST; i++)
+ for (int32_t i = 0; i <= MAX_INST; i++)
{
if (instr[i] != NULL)
{
@@ -58,7 +58,7 @@ static bool setTmpInstruments(void)
return false;
}
- tmpInstr[i] = instr[i];
+ *tmpInstr[i] = *instr[i];
}
}
diff --git a/src/ft2_video.c b/src/ft2_video.c
@@ -953,7 +953,7 @@ bool setupWindow(void)
return false;
}
-#ifdef __APPLE__ // for macOS we need to do this here for reasons I can't be bothered to explain
+#ifdef __APPLE__ // for macOS we need to do this here for reasons I have forgotten
SDL_PumpEvents();
SDL_ShowWindow(video.window);
#endif
