From 5390f50e516aaa8ec4370db30cc528a1757bd4b3 Mon Sep 17 00:00:00 2001 From: Cacodemon345 Date: Sun, 18 May 2025 13:47:06 +0600 Subject: [PATCH] Revert "Switch to SAASound for CMS" This reverts commit fd618440618e3d6e426d60e59921fd8495531b09. --- src/include/86box/snd_cms.h | 23 +- src/sound/CMakeLists.txt | 3 - src/sound/saasound/CMakeLists.txt | 16 - src/sound/saasound/SAAAmp.cpp | 203 --------- src/sound/saasound/SAAAmp.h | 44 -- src/sound/saasound/SAAConfig.h | 41 -- src/sound/saasound/SAADevice.cpp | 392 ----------------- src/sound/saasound/SAADevice.h | 69 --- src/sound/saasound/SAAEnv.cpp | 380 ---------------- src/sound/saasound/SAAEnv.h | 54 --- src/sound/saasound/SAAFreq.cpp | 287 ------------ src/sound/saasound/SAAFreq.dat | 141 ------ src/sound/saasound/SAAFreq.h | 72 --- src/sound/saasound/SAAImpl.cpp | 489 --------------------- src/sound/saasound/SAAImpl.h | 75 ---- src/sound/saasound/SAANoise.cpp | 180 -------- src/sound/saasound/SAANoise.h | 54 --- src/sound/saasound/SAASndC.cpp | 100 ----- src/sound/saasound/SAASndC.h | 102 ----- src/sound/saasound/SAASound.cpp | 13 - src/sound/saasound/SAASound.h | 130 ------ src/sound/saasound/defns.h | 59 --- src/sound/saasound/resource.h | 15 - src/sound/saasound/saasound_cmake_config.h | 14 - src/sound/saasound/types.h | 34 -- src/sound/snd_cms.c | 140 ++++-- 26 files changed, 115 insertions(+), 3015 deletions(-) delete mode 100644 src/sound/saasound/CMakeLists.txt delete mode 100755 src/sound/saasound/SAAAmp.cpp delete mode 100755 src/sound/saasound/SAAAmp.h delete mode 100644 src/sound/saasound/SAAConfig.h delete mode 100644 src/sound/saasound/SAADevice.cpp delete mode 100644 src/sound/saasound/SAADevice.h delete mode 100755 src/sound/saasound/SAAEnv.cpp delete mode 100755 src/sound/saasound/SAAEnv.h delete mode 100755 src/sound/saasound/SAAFreq.cpp delete mode 100755 src/sound/saasound/SAAFreq.dat delete mode 100755 src/sound/saasound/SAAFreq.h delete mode 100644 src/sound/saasound/SAAImpl.cpp delete mode 100755 src/sound/saasound/SAAImpl.h delete mode 100755 src/sound/saasound/SAANoise.cpp delete mode 100755 src/sound/saasound/SAANoise.h delete mode 100755 src/sound/saasound/SAASndC.cpp delete mode 100644 src/sound/saasound/SAASndC.h delete mode 100755 src/sound/saasound/SAASound.cpp delete mode 100644 src/sound/saasound/SAASound.h delete mode 100644 src/sound/saasound/defns.h delete mode 100755 src/sound/saasound/resource.h delete mode 100644 src/sound/saasound/saasound_cmake_config.h delete mode 100755 src/sound/saasound/types.h diff --git a/src/include/86box/snd_cms.h b/src/include/86box/snd_cms.h index 8201fe32c..8eec22935 100644 --- a/src/include/86box/snd_cms.h +++ b/src/include/86box/snd_cms.h @@ -7,20 +7,23 @@ #define MASTER_CLOCK 7159090 typedef struct cms_t { -#ifdef SAASOUND_H_INCLUDED - SAASND saasound; - SAASND saasound2; -#else - void* saasound; - void* saasound2; -#endif + int addrs[2]; + uint8_t regs[2][32]; + uint16_t latch[2][6]; + int freq[2][6]; + float count[2][6]; + int vol[2][6][2]; + int stat[2][6]; + uint16_t noise[2][2]; + uint16_t noisefreq[2][2]; + int noisecount[2][2]; + int noisetype[2][2]; uint8_t latched_data; - int16_t buffer[WTBUFLEN * 2]; - int16_t buffer2[WTBUFLEN * 2]; + int16_t buffer[SOUNDBUFLEN * 2]; - int pos, pos2; + int pos; } cms_t; extern void cms_update(cms_t *cms); diff --git a/src/sound/CMakeLists.txt b/src/sound/CMakeLists.txt index d575717a0..0a04b0ff1 100644 --- a/src/sound/CMakeLists.txt +++ b/src/sound/CMakeLists.txt @@ -180,9 +180,6 @@ endif() add_subdirectory(ymfm) target_link_libraries(86Box ymfm) -add_subdirectory(saasound) -target_link_libraries(86Box saasound) - if(GUSMAX) target_compile_definitions(snd PRIVATE USE_GUSMAX) endif() diff --git a/src/sound/saasound/CMakeLists.txt b/src/sound/saasound/CMakeLists.txt deleted file mode 100644 index 2db75493e..000000000 --- a/src/sound/saasound/CMakeLists.txt +++ /dev/null @@ -1,16 +0,0 @@ -add_library(saasound OBJECT - SAAAmp.cpp - SAAAmp.h - SAADevice.cpp - SAADevice.h - SAAEnv.cpp - SAAEnv.h - SAAFreq.cpp - SAAFreq.h - SAAImpl.cpp - SAAImpl.h - SAANoise.cpp - SAANoise.h - SAASndC.cpp - SAASndC.h - SAASound.cpp) \ No newline at end of file diff --git a/src/sound/saasound/SAAAmp.cpp b/src/sound/saasound/SAAAmp.cpp deleted file mode 100755 index 8f2473fb1..000000000 --- a/src/sound/saasound/SAAAmp.cpp +++ /dev/null @@ -1,203 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAAmp.cpp: implementation of the CSAAAmp class. -// This class handles Tone/Noise mixing, Envelope application and -// amplification. -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" -#include "types.h" -#include "SAANoise.h" -#include "SAAEnv.h" -#include "SAAFreq.h" -#include "SAAAmp.h" -#include "defns.h" - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAAAmp::CSAAAmp(CSAAFreq * const ToneGenerator, const CSAANoise * const NoiseGenerator, const CSAAEnv * const EnvGenerator) -: -m_pcConnectedToneGenerator(ToneGenerator), -m_pcConnectedNoiseGenerator(NoiseGenerator), -m_pcConnectedEnvGenerator(EnvGenerator), -m_bUseEnvelope(EnvGenerator != NULL) -{ - leftlevel = 0; - leftlevela0x0e = 0; - rightlevel = 0; - rightlevela0x0e = 0; - m_nMixMode = 0; - m_bMute=true; - m_bSync = false; - m_nOutputIntermediate=0; - last_level_byte=0; - SetAmpLevel(0x00); - -} - -CSAAAmp::~CSAAAmp() -{ - // Nothing to do -} - -void CSAAAmp::SetAmpLevel(BYTE level_byte) -{ - // if level unchanged since last call then do nothing - if (level_byte != last_level_byte) - { - last_level_byte = level_byte; - leftlevel = level_byte & 0x0f; - leftlevela0x0e = leftlevel & 0x0e; - - rightlevel = (level_byte >> 4) & 0x0f; - rightlevela0x0e = rightlevel & 0x0e; - } - -} - -void CSAAAmp::SetToneMixer(BYTE bEnabled) -{ - if (bEnabled == 0) - { - // clear mixer bit - m_nMixMode &= ~(0x01); - } - else - { - // set mixer bit - m_nMixMode |= 0x01; - } -} - -void CSAAAmp::SetNoiseMixer(BYTE bEnabled) -{ - if (bEnabled == 0) - { - m_nMixMode &= ~(0x02); - } - else - { - m_nMixMode |= 0x02; - } -} - -void CSAAAmp::Mute(bool bMute) -{ - // m_bMute refers to the GLOBAL mute setting (register 28 bit 0) - // NOT the per-channel mixer settings !! - m_bMute = bMute; -} - -void CSAAAmp::Sync(bool bSync) -{ - // m_bSync refers to the GLOBAL sync setting (register 28 bit 1) - m_bSync = bSync; -} - -void CSAAAmp::Tick(void) -{ - // updates m_nOutputIntermediate to 0, 1 or 2 - // - - // connected oscillator always ticks (this isn't really connected to the amp) - int level = m_pcConnectedToneGenerator->Tick(); - - switch (m_nMixMode) - { - case 0: - // no tone or noise for this channel - m_nOutputIntermediate = 0; - break; - case 1: - // tone only for this channel - m_nOutputIntermediate = level * 2; - // NOTE: ConnectedToneGenerator returns either 0 or 1 - break; - case 2: - // noise only for this channel - m_nOutputIntermediate = m_pcConnectedNoiseGenerator->Level() * 2; - // NOTE: ConnectedNoiseGenerator()->Level() returns either 0 or 1 - break; - case 3: - // tone+noise for this channel ... mixing algorithm : - // tone noise output - // 0 0 0 - // 1 0 2 - // 0 1 0 - // 1 1 1 - // = 2 * tone - 1 * (tone & noise) - // = tone * (2 - noise) - m_nOutputIntermediate = level * (2 - m_pcConnectedNoiseGenerator->Level()); - break; - } - // intermediate is between 0 and 2 -} - -inline int CSAAAmp::EffectiveAmplitude(int amp, int env) const -{ - // Return the effective amplitude of the low-pass-filtered result of the logical - // AND of the amplitude PDM and envelope PDM patterns. This is a more accurate - // evaluation of the SAA than simply returning amp * env , based on how the SAA - // implements pulse-density modulation. - static const int pdm[16][16] = { - {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}, - {0,0,0,0,2,2,2,2,2,2,2,2,4,4,4,4}, - {0,1,1,2,2,3,3,4,4,5,5,6,6,7,7,8}, - {0,1,1,2,4,5,5,6,6,7,7,8,10,11,11,12}, - {0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15}, - {0,1,2,3,6,7,8,9,10,11,12,13,16,17,18,19}, - {0,2,3,5,6,8,9,11,12,14,15,17,18,20,21,23}, - {0,2,3,5,8,10,11,13,14,16,17,19,22,24,25,27}, - {0,2,4,6,8,10,12,14,16,18,20,22,24,26,28,30}, - {0,2,4,6,10,12,14,16,18,20,22,24,28,30,32,34}, - {0,3,5,8,10,13,15,18,20,23,25,28,30,33,35,38}, - {0,3,5,8,12,15,17,20,22,25,27,30,34,37,39,42}, - {0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45}, - {0,3,6,9,14,17,20,23,26,29,32,35,40,43,46,49}, - {0,4,7,11,14,18,21,25,28,32,35,39,42,46,49,53}, - {0,4,7,11,16,20,23,27,30,34,37,41,46,50,53,57} - }; - - return(pdm[amp][env] * 4); -} - -void CSAAAmp::TickAndOutputStereo(unsigned int & left, unsigned int & right) -{ - // This returns a value between 0 and 480 inclusive. - // This represents the full dynamic range of one output mixer (tone, or noise+tone, at full volume, - // without envelopes enabled). Note that, with envelopes enabled, the actual dynamic range - // is reduced on-chip to just over 88% of this (424), so the "loudest" output requires disabling envs. - // NB for 6 channels at full volume, with simple additive mixing, you would see a combined - // output of 2880, and a multiplier of 11 (=31680) fits comfortably within 16-bit signed output range. - - if (m_bSync) - { - // TODO check this - left = right = 0; - return; - } - - // first, do the Tick: - Tick(); - - // now calculate the returned amplitude for this sample: - //////////////////////////////////////////////////////// - - if (m_bMute) - { - left = right = 0; - } - else if (m_bUseEnvelope && m_pcConnectedEnvGenerator->IsActive()) - { - left = EffectiveAmplitude(m_pcConnectedEnvGenerator->LeftLevel(), leftlevela0x0e) * (2 - m_nOutputIntermediate); - right = EffectiveAmplitude(m_pcConnectedEnvGenerator->RightLevel(), rightlevela0x0e) * (2 - m_nOutputIntermediate); - } - else - { - left = leftlevel * m_nOutputIntermediate * 16; - right = rightlevel * m_nOutputIntermediate * 16; - } -} diff --git a/src/sound/saasound/SAAAmp.h b/src/sound/saasound/SAAAmp.h deleted file mode 100755 index 4a6761f21..000000000 --- a/src/sound/saasound/SAAAmp.h +++ /dev/null @@ -1,44 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAAmp.h: interface for the CSAAAmp class. -// This class handles Tone/Noise mixing, Envelope application and -// amplification. -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAAAMP_H_INCLUDED -#define SAAAMP_H_INCLUDED - -class CSAAAmp -{ -private: - int leftlevel; - int leftlevela0x0e; - int rightlevel; - int rightlevela0x0e; - int m_nOutputIntermediate; - unsigned int m_nMixMode; - CSAAFreq * const m_pcConnectedToneGenerator; // not const because amp calls ->Tick() - const CSAANoise * const m_pcConnectedNoiseGenerator; - const CSAAEnv * const m_pcConnectedEnvGenerator; - const bool m_bUseEnvelope; - mutable bool m_bMute; - mutable bool m_bSync; - mutable BYTE last_level_byte; - int EffectiveAmplitude(int amp, int env) const; - -public: - CSAAAmp(CSAAFreq * const ToneGenerator, const CSAANoise * const NoiseGenerator, const CSAAEnv * const EnvGenerator); - ~CSAAAmp(); - - void SetAmpLevel(BYTE level_byte); // really just a BYTE - void SetToneMixer(BYTE bEnabled); - void SetNoiseMixer(BYTE bEnabled); - void Mute(bool bMute); - void Sync(bool bSync); - void Tick(void); - void TickAndOutputStereo(unsigned int & left, unsigned int & right); - -}; - -#endif // SAAAMP_H_INCLUDED diff --git a/src/sound/saasound/SAAConfig.h b/src/sound/saasound/SAAConfig.h deleted file mode 100644 index a655ec59f..000000000 --- a/src/sound/saasound/SAAConfig.h +++ /dev/null @@ -1,41 +0,0 @@ -// Part of SAASound copyright 2020 Dave Hooper -// -// SAAConfig.h: configuration file handler class -// -////////////////////////////////////////////////////////////////////// - -#include "defns.h" -#ifdef USE_CONFIG_FILE - -#ifndef SAA_CONFIG_H_INCLUDED -#define SAA_CONFIG_H_INCLUDED - -#define INI_READONLY -#define INI_ANSIONLY /*nb not really 'ANSI', this just forces all read/write to use 8-bit char*/ -#include "minIni/minIni.h" - -class SAAConfig -{ -private: - minIni m_minIni; - bool m_bHasReadConfig; - -public: - bool m_bGenerateRegisterLogs; - bool m_bGeneratePcmLogs; - bool m_bGeneratePcmSeparateChannels; - t_string m_strRegisterLogPath; - t_string m_strPcmOutputPath; - unsigned int m_nOversample; - bool m_bHighpass; - double m_nBoost; - - SAAConfig(); - void ReadConfig(); - - t_string getChannelPcmOutputPath(int); -}; - -#endif // SAA_CONFIG_H_INCLUDED - -#endif // USE_CONFIG_FILE \ No newline at end of file diff --git a/src/sound/saasound/SAADevice.cpp b/src/sound/saasound/SAADevice.cpp deleted file mode 100644 index 718b05a95..000000000 --- a/src/sound/saasound/SAADevice.cpp +++ /dev/null @@ -1,392 +0,0 @@ -// Part of SAASound copyright 2020 Dave Hooper -// -// SAADevice.cpp: connecting the subcomponents of the SAA1099 together. -// This class handles device inputs and outputs (clocking, data and -// address bus, and simulated output) -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" -#include "types.h" -#include "SAAEnv.h" -#include "SAANoise.h" -#include "SAAFreq.h" -#include "SAAAmp.h" -#include "SAASound.h" -#include "SAAImpl.h" -#include "defns.h" - - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAADevice::CSAADevice() - : - m_nCurrentSaaReg(0), - m_bOutputEnabled(false), - m_bSync(false), - m_bHighpass(true), - m_nOversample(0), - m_Noise0(0xffffffff), - m_Noise1(0xffffffff), - m_Env0(), - m_Env1(), - m_Osc0(&m_Noise0, NULL), - m_Osc1(NULL, &m_Env0), - m_Osc2(NULL, NULL), - m_Osc3(&m_Noise1, NULL), - m_Osc4(NULL, &m_Env1), - m_Osc5(NULL, NULL), - m_Amp0(&m_Osc0, &m_Noise0, NULL), - m_Amp1(&m_Osc1, &m_Noise0, NULL), - m_Amp2(&m_Osc2, &m_Noise0, &m_Env0), - m_Amp3(&m_Osc3, &m_Noise1, NULL), - m_Amp4(&m_Osc4, &m_Noise1, NULL), - m_Amp5(&m_Osc5, &m_Noise1, &m_Env1) -{ - // Create and link up the objects that make up the emulator - Noise[0] = &m_Noise0; - Noise[1] = &m_Noise1; - Env[0] = &m_Env0; - Env[1] = &m_Env1; - - // Create oscillators (tone generators) and link to noise generators and - // envelope controllers - Osc[0] = &m_Osc0; - Osc[1] = &m_Osc1; - Osc[2] = &m_Osc2; - Osc[3] = &m_Osc3; - Osc[4] = &m_Osc4; - Osc[5] = &m_Osc5; - - // Create amplification/mixing stages and link to appropriate oscillators, - // noise generators and envelope controllers - Amp[0] = &m_Amp0; - Amp[1] = &m_Amp1; - Amp[2] = &m_Amp2; - Amp[3] = &m_Amp3; - Amp[4] = &m_Amp4; - Amp[5] = &m_Amp5; - - _SetClockRate(EXTERNAL_CLK_HZ); - _SetOversample(DEFAULT_OVERSAMPLE); -} - -CSAADevice::~CSAADevice() -{ -} - -////////////////////////////////////////////////////////////////////// -// CSAASound members -////////////////////////////////////////////////////////////////////// - -void CSAADevice::_SetClockRate(unsigned int nClockRate) -{ - m_Osc0._SetClockRate(nClockRate); - m_Osc1._SetClockRate(nClockRate); - m_Osc2._SetClockRate(nClockRate); - m_Osc3._SetClockRate(nClockRate); - m_Osc4._SetClockRate(nClockRate); - m_Osc5._SetClockRate(nClockRate); - m_Noise0._SetClockRate(nClockRate); - m_Noise1._SetClockRate(nClockRate); -} - -void CSAADevice::_SetSampleRate(unsigned int nSampleRate) -{ - m_Osc0._SetSampleRate(nSampleRate); - m_Osc1._SetSampleRate(nSampleRate); - m_Osc2._SetSampleRate(nSampleRate); - m_Osc3._SetSampleRate(nSampleRate); - m_Osc4._SetSampleRate(nSampleRate); - m_Osc5._SetSampleRate(nSampleRate); - m_Noise0._SetSampleRate(nSampleRate); - m_Noise1._SetSampleRate(nSampleRate); -} - -void CSAADevice::_SetOversample(unsigned int nOversample) -{ - if (((int) nOversample) != m_nOversample) - { - m_nOversample = nOversample; - m_Osc0._SetOversample(nOversample); - m_Osc1._SetOversample(nOversample); - m_Osc2._SetOversample(nOversample); - m_Osc3._SetOversample(nOversample); - m_Osc4._SetOversample(nOversample); - m_Osc5._SetOversample(nOversample); - m_Noise0._SetOversample(nOversample); - m_Noise1._SetOversample(nOversample); - } -} - -void CSAADevice::_WriteData(BYTE nData) -{ -#if defined(DEBUG) || defined(DEBUGSAA) - m_Reg[m_nCurrentSaaReg] = nData; -#endif - - // route nData to the appropriate place - switch (m_nCurrentSaaReg) - { - // Amplitude data (==> Amp) - case 0: - m_Amp0.SetAmpLevel(nData); - break; - case 1: - m_Amp1.SetAmpLevel(nData); - break; - case 2: - m_Amp2.SetAmpLevel(nData); - break; - case 3: - m_Amp3.SetAmpLevel(nData); - break; - case 4: - m_Amp4.SetAmpLevel(nData); - break; - case 5: - m_Amp5.SetAmpLevel(nData); - break; - - // Freq data (==> Osc) - case 8: - m_Osc0.SetFreqOffset(nData); - break; - case 9: - m_Osc1.SetFreqOffset(nData); - break; - case 10: - m_Osc2.SetFreqOffset(nData); - break; - case 11: - m_Osc3.SetFreqOffset(nData); - break; - case 12: - m_Osc4.SetFreqOffset(nData); - break; - case 13: - m_Osc5.SetFreqOffset(nData); - break; - - // Freq octave data (==> Osc) for channels 0,1 - case 16: - m_Osc0.SetFreqOctave(nData & 0x07); - m_Osc1.SetFreqOctave((nData >> 4) & 0x07); - break; - - // Freq octave data (==> Osc) for channels 2,3 - case 17: - m_Osc2.SetFreqOctave(nData & 0x07); - m_Osc3.SetFreqOctave((nData >> 4) & 0x07); - break; - - // Freq octave data (==> Osc) for channels 4,5 - case 18: - m_Osc4.SetFreqOctave(nData & 0x07); - m_Osc5.SetFreqOctave((nData >> 4) & 0x07); - break; - - // Tone mixer control (==> Amp) - case 20: - m_Amp0.SetToneMixer(nData & 0x01); - m_Amp1.SetToneMixer(nData & 0x02); - m_Amp2.SetToneMixer(nData & 0x04); - m_Amp3.SetToneMixer(nData & 0x08); - m_Amp4.SetToneMixer(nData & 0x10); - m_Amp5.SetToneMixer(nData & 0x20); - break; - - // Noise mixer control (==> Amp) - case 21: - m_Amp0.SetNoiseMixer(nData & 0x01); - m_Amp1.SetNoiseMixer(nData & 0x02); - m_Amp2.SetNoiseMixer(nData & 0x04); - m_Amp3.SetNoiseMixer(nData & 0x08); - m_Amp4.SetNoiseMixer(nData & 0x10); - m_Amp5.SetNoiseMixer(nData & 0x20); - break; - - // Noise frequency/source control (==> Noise) - case 22: - m_Noise0.SetSource(nData & 0x03); - m_Noise1.SetSource((nData >> 4) & 0x03); - break; - - // Envelope control data (==> Env) for envelope controller #0 - case 24: - m_Env0.SetEnvControl(nData); - break; - - // Envelope control data (==> Env) for envelope controller #1 - case 25: - m_Env1.SetEnvControl(nData); - break; - - // Global enable and reset (sync) controls - case 28: - { - // Reset (sync) bit - bool bSync = bool(nData & 0x02); - if (bSync != m_bSync) - { - // Sync all devices - // This amounts to telling them all to reset to a - // known state, which is also a state that doesn't change - // (i.e. no audio output, although there are some exceptions) - // bSync=true => all devices are sync (aka reset); - // bSync=false => all devices are allowed to run and generate changing output - m_Osc0.Sync(bSync); - m_Osc1.Sync(bSync); - m_Osc2.Sync(bSync); - m_Osc3.Sync(bSync); - m_Osc4.Sync(bSync); - m_Osc5.Sync(bSync); - m_Noise0.Sync(bSync); - m_Noise1.Sync(bSync); - m_Amp0.Sync(bSync); - m_Amp1.Sync(bSync); - m_Amp2.Sync(bSync); - m_Amp3.Sync(bSync); - m_Amp4.Sync(bSync); - m_Amp5.Sync(bSync); - m_bSync = bSync; - } - - // Global mute bit - bool bOutputEnabled = bool(nData & 0x01); - if (bOutputEnabled != m_bOutputEnabled) - { - // unmute all amps - sound 'enabled' - m_Amp0.Mute(!bOutputEnabled); - m_Amp1.Mute(!bOutputEnabled); - m_Amp2.Mute(!bOutputEnabled); - m_Amp3.Mute(!bOutputEnabled); - m_Amp4.Mute(!bOutputEnabled); - m_Amp5.Mute(!bOutputEnabled); - m_bOutputEnabled = bOutputEnabled; - } - } - break; - - default: - // anything else means data is being written to a register - // that is not used within the SAA-1099 architecture - // hence, we ignore it. - {} - } -} - -void CSAADevice::_WriteAddress(BYTE nReg) -{ - m_nCurrentSaaReg = nReg & 31; - if (m_nCurrentSaaReg == 24) - { - m_Env0.ExternalClock(); - } - else if (m_nCurrentSaaReg == 25) - { - m_Env1.ExternalClock(); - } -} - -#if 1 -BYTE CSAADevice::_ReadAddress(void) -{ - // Not a real hardware function of the SAA-1099, which is write-only - // However, this is used by SAAImpl to generate debug logs (if enabled) - return(m_nCurrentSaaReg); -} -#endif -#if defined(DEBUG) -BYTE CSAADevice::_ReadData(void) -{ - // Not a real hardware function of the SAA-1099, which is write-only - // This is only compiled for Debug builds - return(m_Reg[m_nCurrentSaaReg]); -} -#endif - -void CSAADevice::_TickAndOutputStereo(unsigned int& left_mixed, unsigned int& right_mixed) -{ - unsigned int temp_left, temp_right; - unsigned int accum_left = 0, accum_right = 0; - for (int i = 1 << m_nOversample; i > 0; i--) - { - m_Noise0.Tick(); - m_Noise1.Tick(); - m_Amp0.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - m_Amp1.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - m_Amp2.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - m_Amp3.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - m_Amp4.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - m_Amp5.TickAndOutputStereo(temp_left, temp_right); - accum_left += temp_left; - accum_right += temp_right; - } - left_mixed = accum_left; - right_mixed = accum_right; -} - -void CSAADevice::_TickAndOutputSeparate(unsigned int& left_mixed, unsigned int& right_mixed, - unsigned int& left0, unsigned int& right0, - unsigned int& left1, unsigned int& right1, - unsigned int& left2, unsigned int& right2, - unsigned int& left3, unsigned int& right3, - unsigned int& left4, unsigned int& right4, - unsigned int& left5, unsigned int& right5 -) -{ - unsigned int temp_left, temp_right; - unsigned int accum_left = 0, accum_right = 0; - left0 = left1 = left2 = left3 = left4 = left5 = 0; - right0 = right1 = right2 = right3 = right4 = right5 = 0; - for (int i = 1 << m_nOversample; i > 0; i--) - { - m_Noise0.Tick(); - m_Noise1.Tick(); - m_Amp0.TickAndOutputStereo(temp_left, temp_right); - left0 += temp_left; - right0 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - m_Amp1.TickAndOutputStereo(temp_left, temp_right); - left1 += temp_left; - right1 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - m_Amp2.TickAndOutputStereo(temp_left, temp_right); - left2 += temp_left; - right2 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - m_Amp3.TickAndOutputStereo(temp_left, temp_right); - left3 += temp_left; - right3 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - m_Amp4.TickAndOutputStereo(temp_left, temp_right); - left4 += temp_left; - right4 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - m_Amp5.TickAndOutputStereo(temp_left, temp_right); - left5 += temp_left; - right5 += temp_right; - accum_left += temp_left; - accum_right += temp_right; - } - left_mixed = accum_left; - right_mixed = accum_right; -} \ No newline at end of file diff --git a/src/sound/saasound/SAADevice.h b/src/sound/saasound/SAADevice.h deleted file mode 100644 index 7b697821f..000000000 --- a/src/sound/saasound/SAADevice.h +++ /dev/null @@ -1,69 +0,0 @@ -// Part of SAASound copyright 2020 Dave Hooper -// -// SAADevice.h: connecting the subcomponents of the SAA1099 together. -// This class handles device inputs and outputs (clocking, data and -// address bus, and simulated output) -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAADEVICE_H_INCLUDED -#define SAADEVICE_H_INCLUDED - -#include "SAASound.h" -#include "SAANoise.h" -#include "SAAEnv.h" -#include "SAAFreq.h" -#include "SAAAmp.h" - -class CSAADevice -{ -private: - int m_nCurrentSaaReg; - bool m_bOutputEnabled; - bool m_bSync; - bool m_bHighpass; - int m_nOversample; - - CSAANoise m_Noise0, m_Noise1; - CSAAEnv m_Env0, m_Env1; - CSAAFreq m_Osc0, m_Osc1, m_Osc2, m_Osc3, m_Osc4, m_Osc5; - CSAAAmp m_Amp0, m_Amp1, m_Amp2, m_Amp3, m_Amp4, m_Amp5; - - CSAANoise* Noise[2]; - CSAAEnv* Env[2]; - CSAAFreq* Osc[6]; - CSAAAmp* Amp[6]; - -#if defined(DEBUG) || defined(DEBUGSAA) - BYTE m_Reg[32]; -#endif - -public: - CSAADevice(); - ~CSAADevice(); - - void _WriteAddress(BYTE nReg); - void _WriteData(BYTE nData); -#if 1 - BYTE _ReadAddress(void); -#endif -#if defined(DEBUG) - BYTE _ReadData(void); -#endif - - void _SetClockRate(unsigned int nClockRate); - void _SetSampleRate(unsigned int nSampleRate); - void _SetOversample(unsigned int nOversample); - void _TickAndOutputStereo(unsigned int& left_mixed, unsigned int& right_mixed); - void _TickAndOutputSeparate(unsigned int& left_mixed, unsigned int& right_mixed, - unsigned int& left0, unsigned int& right0, - unsigned int& left1, unsigned int& right1, - unsigned int& left2, unsigned int& right2, - unsigned int& left3, unsigned int& right3, - unsigned int& left4, unsigned int& right4, - unsigned int& left5, unsigned int& right5 - ); - -}; - -#endif // SAADEVICE_H_INCLUDED \ No newline at end of file diff --git a/src/sound/saasound/SAAEnv.cpp b/src/sound/saasound/SAAEnv.cpp deleted file mode 100755 index 049f51f96..000000000 --- a/src/sound/saasound/SAAEnv.cpp +++ /dev/null @@ -1,380 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAEnv.cpp: implementation of the CSAAEnv class. -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" -#include "types.h" -#include "SAAEnv.h" - - -////////////////////////////////////////////////////////////////////// -// Static member initialisation -////////////////////////////////////////////////////////////////////// - -const ENVDATA CSAAEnv::cs_EnvData[8] = -{ - {1,false, { {{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}, - {{0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}}}, - {1,true, { {{15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15},{15,15,15,15,15,15,15,15,15,15,15,15,15,15,15,15}}, - {{14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14},{14,14,14,14,14,14,14,14,14,14,14,14,14,14,14,14}}}}, - {1,false, { {{15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}, - {{14,14,12,12,10,10,8,8,6,6,4,4,2,2,0,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}}}, - {1,true, { {{15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}, - {{14,14,12,12,10,10,8,8,6,6,4,4,2,2,0,0}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}}}, - {2,false, { {{0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15}, {15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0}}, - {{0,0,2,2,4,4,6,6,8,8,10,10,12,12,14,14}, {14,14,12,12,10,10,8,8,6,6,4,4,2,2,0,0}}}}, - {2,true, { {{0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15}, {15,14,13,12,11,10,9,8,7,6,5,4,3,2,1,0}}, - {{0,0,2,2,4,4,6,6,8,8,10,10,12,12,14,14}, {14,14,12,12,10,10,8,8,6,6,4,4,2,2,0,0}}}}, - {1,false, { {{0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}, - {{0,0,2,2,4,4,6,6,8,8,10,10,12,12,14,14}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}}}, - {1,true, { {{0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}, - {{0,0,2,2,4,4,6,6,8,8,10,10,12,12,14,14}, {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}}}} -}; - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAAEnv::CSAAEnv() -: -m_bEnabled(false), -m_nPhase(0), -m_nPhasePosition(0), -m_bEnvelopeEnded(true), -m_nResolution(1), -m_bNewData(false), -m_nNextData(0) -{ - // initialise itself with the value 'zero' - SetEnvControl(0); -} - -CSAAEnv::~CSAAEnv() -{ - // Nothing to do -} - -void CSAAEnv::InternalClock(void) -{ - // will only do something if envelope clock mode is set to internal - // and the env control is enabled - if (m_bEnabled && (!m_bClockExternally)) Tick(); -} - -void CSAAEnv::ExternalClock(void) -{ - // will only do something if envelope clock mode is set to external - // and the env control is enabled - if (m_bClockExternally && m_bEnabled) Tick(); -} - -void CSAAEnv::SetEnvControl(int nData) -{ - // process immediate stuff first: - // start with the Enabled flag. if env is disabled, - // there's not much to do - bool bEnabled = ((nData & 0x80)==0x80); - if (!bEnabled && !m_bEnabled) - return; - m_bEnabled = bEnabled; - if (!m_bEnabled) - { - // env control was enabled, and now disabled - // Any subsequent env control changes are immediate. - m_bEnvelopeEnded = true; - return; - } - - // Resolution (3bit/4bit) is also immediately processed - int new_resolution = ((nData & 0x10) == 0x10) ? 2 : 1; - // NOTE: undocumented behaviour when changing resolution mid-waveform - // Empirically, the following matches observations: - // * When ticking the env generator with 4-bit resolution, the position += 1 - // * When ticking the env generator with 3-bit resolution, the position += 2 - // * When changing between 4-bit resolution and 3-bit resolution - // without ticking the env generator, the position is unchanged - // (although, effectively, the LSB is ignored. Purely as an implementation - // detail, I'm implementing this as clearing the LSB ie LSB=0; see next point) - // * When changing between 3-bit resolution and 4-bit resolution - // without ticking the env generator, the position LSB is set to 1 - // See test case: envext_34b - // - if (m_nResolution == 1 && new_resolution == 2) - { - // change from 4-bit to 3-bit - m_nPhasePosition &= 0xe; - } - else if (m_nResolution == 2 && new_resolution == 1) - { - // change from 3-bit to 4-bit - m_nPhasePosition |= 0x1; - } - m_nResolution = new_resolution; - - // now buffered stuff: but only if it's ok to, and only if the - // envgenerator is not disabled. otherwise it just stays buffered until - // the Tick() function sets m_bEnvelopeEnded to true and realises there is - // already some new data waiting - if (m_bEnvelopeEnded) - { - SetNewEnvData(nData); // also does the SetLevels() call for us. - m_bNewData=false; - } - else - { - // since the 'next resolution' changes arrive unbuffered, we - // may need to change the current level because of this: - SetLevels(); - - // store current new data, and set the newdata flag: - m_bNewData = true; - m_nNextData = nData; - } - -} - -int CSAAEnv::LeftLevel(void) const -{ - return m_nLeftLevel; -} - -int CSAAEnv::RightLevel(void) const -{ - return m_nRightLevel; -} - -inline void CSAAEnv::Tick(void) -{ - // if disabled, do nothing - if (!m_bEnabled) // m_bEnabled is set directly, not buffered, so this is ok - { - // for sanity, reset stuff: - m_bEnvelopeEnded = true; - m_nPhase = 0; - m_nPhasePosition = 0; - return; - } - - // else : m_bEnabled - - - if (m_bEnvelopeEnded) - { - // do nothing - // (specifically, don't change the values of m_bEnvelopeEnded, - // m_nPhase and m_nPhasePosition, as these will still be needed - // by SetLevels() should it be called again) - - return; - } - - - // else : !m_bEnvelopeEnded - // Continue playing the same envelope ... - // increments the phaseposition within an envelope. - // also handles looping and resolution appropriately. - // Changes the level of the envelope accordingly - // through calling SetLevels() . This must be called after making - // any changes that will affect the output levels of the env controller!! - // SetLevels also handles left-right channel inverting - - // increment phase position - m_nPhasePosition += m_nResolution; - - // if this means we've gone past 16 (the end of a phase) - // then change phase, and if necessary, loop - // Refer to datasheet for meanings of (3) and (4) in following text - // w.r.t SAA1099 envelopes - - // Note that we will always reach position (3) or (4), even if we keep toggling - // resolution from 4-bit to 3-bit and back, because the counter will always wrap to 0. - // In fact it's quite elegant: - // No matter how you increment and toggle and increment and toggle, the counter - // will at some point be either 0xe (either 4-bit mode or 3-bit mode) or 0xf (4-bit mode only). - // Depending on the mode, even if you change the mode, the next increment, - // or the one after it, will then take it to 0. - // 0xe + 2 (3bit mode) => 0x0 - // 0xe + 1 (4bit mode) => 0xf - // 0xf + 1 (4bit mode) => 0x0 - // 0xe -> (toggle 3bit mode to 4bit mode) => 0xf - // 0xe -> (toggle 4bit mode to 3bit mode) => 0xe - // 0xf -> (toggle 4bit mode to 3bit mode) => 0xe - // - // but there is a subtlety (of course), which is that any changes at point (3) - // can take place immediately you hit point (3), but changes at point (4) are actually - // only acted upon when the counter transitions from 0xe (or 0xf) to 0x0 (which also - // means that, for these looping envelopes, which are the ones that have a point(4), - // immediately after the counter wrapping to 0x0, a write to the env data register will - // NOT set the waveform and will NOT reset the phase/phaseposition (even though it - // will still let you toggle the 4bit/3bit mode, which will change the phaseposition LSB!) - // See test case: envext_34c - - bool bProcessNewDataIfAvailable = false; - if (m_nPhasePosition >= 16) - { - m_nPhase++; - - // if we should loop, then do so - and we've reached position (4) - // otherwise, if we shouldn't loop, - // then we've reached position (3) and so we say that - // we're ok for new data. - if (m_nPhase == m_nNumberOfPhases) - { - // at position (3) or (4) - if (!m_bLooping) - { - // position (3) only - // note that it seems that the sustain level is ALWAYS zero - // in the case of non-looping waveforms - m_bEnvelopeEnded = true; - bProcessNewDataIfAvailable = true; - } - else - { - // position (4) only - // note that any data already latched is ONLY acted upon - // at THIS point. If (after this Tick has completed) any new - // env data is written, it will NOT be acted upon, until - // we get back to position (4) again. - // this is why m_bEnvelopeEnded (which affects the behaviour - // of the SetEnvControl method) is FALSE here. - // See test case: envext_34c (as noted earlier) - m_bEnvelopeEnded = false; - // set phase pointer to start of envelope for loop - // and reset m_nPhasePosition - m_nPhase=0; - m_nPhasePosition -= 16; - bProcessNewDataIfAvailable = true; - } - } - else // (m_nPhase < m_nNumberOfPhases) - { - // not at position (3) or (4) ... - // (i.e., we're in the middle of an envelope with - // more than one phase. Specifically, we're in - // the middle of envelope 4 or 5 - the - // triangle envelopes - but that's not important) - - // any commands sent to this envelope controller - // will be buffered. Set the flag to indicate this. - m_bEnvelopeEnded = false; - m_nPhasePosition -= 16; - } - } - else // (m_nPhasePosition < 16) - { - // still within the same phase; - // but, importantly, we are no longer at the start of the phase ... - // so new data cannot be acted on immediately, and must - // be buffered - m_bEnvelopeEnded = false; - // Phase and PhasePosition have already been updated. - // SetLevels() will need to be called to actually calculate - // the output 'level' of this envelope controller - } - - - // if we have new (buffered) data, now is the time to act on it - if (m_bNewData && bProcessNewDataIfAvailable) - { - m_bNewData = false; - SetNewEnvData(m_nNextData); - } - else - { - // ok, we didn't have any new buffered date to act on, - // so we just call SetLevels() to calculate the output level - // for whatever the current envelope is - SetLevels(); - } - -} - -inline void CSAAEnv::SetLevels(void) -{ - // sets m_nLeftLevel - // Also sets m_nRightLevel in terms of m_nLeftLevel - // and m_bInvertRightChannel - - // m_nResolution: 1 means 4-bit resolution; 2 means 3-bit resolution. Resolution of envelope waveform. - - // Note that this is handled 'immediately', and doesn't wait for synchronisation of - // the envelope waveform (this is important, see test case EnvExt_imm) - // It is therefore possible to switch between 4-bit and 3-bit resolution in the middle of - // an envelope waveform. if you are at an 'odd' phase position, you would be able to hear - // the difference. if you are at an 'even' phase position, the volume level for 4-bit - // and 3-bit would be the same. - // NOTE: additional test cases are required. - - switch (m_nResolution) - { - case 1: // 4 bit res waveforms - default: - { - // special case: if envelope is not a looping one, and we're at the end - // then our level should be zero (all of the non-looping waveforms have - // a sustain level of zero): - if (m_bEnvelopeEnded && !m_bLooping) - m_nLeftLevel = 0; - else - m_nLeftLevel = m_pEnvData->nLevels[0][m_nPhase][m_nPhasePosition]; - - if (m_bInvertRightChannel) - m_nRightLevel = 15-m_nLeftLevel; - else - m_nRightLevel = m_nLeftLevel; - break; - } - case 2: // 3 bit res waveforms - { - // special case: if envelope is not a looping one, and we're at the end - // then our level should be zero (all of the non-looping waveforms have - // a sustain level of zero): - if (m_bEnvelopeEnded && !m_bLooping) - m_nLeftLevel = 0; - else - m_nLeftLevel = m_pEnvData->nLevels[1][m_nPhase][m_nPhasePosition]; - if (m_bInvertRightChannel) - m_nRightLevel = 14-m_nLeftLevel; - else - m_nRightLevel = m_nLeftLevel; - break; - } - } -} - - -inline void CSAAEnv::SetNewEnvData(int nData) -{ - // loads envgenerator's registers according to the bits set - // in nData - - m_nPhase = 0; - m_nPhasePosition = 0; - m_pEnvData = &(cs_EnvData[(nData >> 1) & 0x07]); - m_bInvertRightChannel = ((nData & 0x01) == 0x01); - m_bClockExternally = ((nData & 0x20) == 0x20); - m_nNumberOfPhases = m_pEnvData->nNumberOfPhases; - m_bLooping = m_pEnvData->bLooping; - m_nResolution = (((nData & 0x10)==0x10) ? 2 : 1); - m_bEnabled = ((nData & 0x80) == 0x80); - if (m_bEnabled) - { - m_bEnvelopeEnded = false; - // is this right? - // YES. See test case EnvExt_34c (setting data multiple times - // when at a point (3) resets the waveform so you're no longer - // at a point (3). - } - else - { - // DISABLED - so set stuff accordingly - m_bEnvelopeEnded = true; - m_nPhase = 0; - m_nPhasePosition = 0; - } - - SetLevels(); -} diff --git a/src/sound/saasound/SAAEnv.h b/src/sound/saasound/SAAEnv.h deleted file mode 100755 index 131659c06..000000000 --- a/src/sound/saasound/SAAEnv.h +++ /dev/null @@ -1,54 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAEnv.h: interface for the CSAAEnv class. -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAAENV_H_INCLUDED -#define SAAENV_H_INCLUDED - -class CSAAEnv -{ -private: - int m_nLeftLevel, m_nRightLevel; - ENVDATA const * m_pEnvData; - - bool m_bEnabled; - bool m_bInvertRightChannel; - BYTE m_nPhase; - BYTE m_nPhasePosition; - bool m_bEnvelopeEnded; - char m_nPhaseAdd[2]; - char m_nCurrentPhaseAdd; - bool m_bLooping; - char m_nNumberOfPhases; - char m_nResolution; - char m_nInitialLevel; - bool m_bNewData; - BYTE m_nNextData; - bool m_bClockExternally; - static const ENVDATA cs_EnvData[8]; - - void Tick(void); - void SetLevels(void); - void SetNewEnvData(int nData); - -public: - CSAAEnv(); - ~CSAAEnv(); - - void InternalClock(void); - void ExternalClock(void); - void SetEnvControl(int nData); // really just a BYTE - int LeftLevel(void) const; - int RightLevel(void) const; - bool IsActive(void) const; - -}; - -inline bool CSAAEnv::IsActive(void) const -{ - return m_bEnabled; -} - -#endif // SAAENV_H_INCLUDED diff --git a/src/sound/saasound/SAAFreq.cpp b/src/sound/saasound/SAAFreq.cpp deleted file mode 100755 index 61a04f6ad..000000000 --- a/src/sound/saasound/SAAFreq.cpp +++ /dev/null @@ -1,287 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAFreq.cpp: implementation of the CSAAFreq class. -// only 7-bit fractional accuracy on oscillator periods. I may consider fixing that. -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" -#include "types.h" -#include "SAANoise.h" -#include "SAAEnv.h" -#include "SAAFreq.h" -#include "defns.h" - -#ifdef SAAFREQ_FIXED_CLOCKRATE -// 'load in' the data for the static frequency lookup table -// precomputed for a fixed clockrate -// See: tools/freqdat.py -const unsigned long CSAAFreq::m_FreqTable[2048] = { -#include "SAAFreq.dat" -}; -#else -unsigned long CSAAFreq::m_FreqTable[2048]; -unsigned long CSAAFreq::m_nClockRate = 0; -#endif // SAAFREQ_FIXED_CLOCKRATE - -const int INITIAL_LEVEL = 1; - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAAFreq::CSAAFreq(CSAANoise * const NoiseGenerator, CSAAEnv * const EnvGenerator) -: -m_nCounter(0), -m_nAdd(0), -m_nCounter_low(0), -m_nOversample(0), -m_nCounterLimit_low(1), -m_nLevel(INITIAL_LEVEL), -m_nCurrentOffset(0), -m_nCurrentOctave(0), -m_nNextOffset(0), -m_nNextOctave(0), -m_bIgnoreOffsetData(false), -m_bNewData(false), -m_bSync(false), -m_nSampleRate(SAMPLE_RATE_HZ), -m_pcConnectedNoiseGenerator(NoiseGenerator), -m_pcConnectedEnvGenerator(EnvGenerator), -m_nConnectedMode((NoiseGenerator == NULL) ? ((EnvGenerator == NULL) ? 0 : 1) : 2) -{ - _SetClockRate(EXTERNAL_CLK_HZ); - SetAdd(); // current octave, current offset -} - -CSAAFreq::~CSAAFreq() -{ - // Nothing to do -} - -void CSAAFreq::SetFreqOffset(BYTE nOffset) -{ - // nOffset between 0 and 255 - - if (!m_bSync) - { - m_nNextOffset = nOffset; - m_bNewData=true; - if (m_nNextOctave==m_nCurrentOctave) - { - // According to Philips, if you send the SAA-1099 - // new Octave data and then new Offset data in that - // order, on the next half-cycle of the current frequency - // generator, ONLY the octave data is acted upon. - // The offset data will be acted upon next time. - - // ?? TEST CASE : if you set the octave and then the offset - // but the octave you set it to is the same one it already was. - // Will this ignore the offset data? - // Do you get the same behaviour if you set offset THEN octave - // even if you set octave to the same value it was before? - - m_bIgnoreOffsetData=true; - } - } - else - { - // updates straightaway if m_bSync - m_bNewData=false; - m_bIgnoreOffsetData = false; - m_nCurrentOffset = nOffset; - m_nNextOffset = nOffset; - m_nCurrentOctave = m_nNextOctave; - SetAdd(); - } - -} - -void CSAAFreq::SetFreqOctave(BYTE nOctave) -{ - // nOctave between 0 and 7 - - if (!m_bSync) - { - m_nNextOctave = nOctave; - m_bNewData=true; - m_bIgnoreOffsetData = false; - } - else - { - // updates straightaway if m_bSync - m_bNewData=false; - m_bIgnoreOffsetData = false; - m_nCurrentOctave = nOctave; - m_nNextOctave = nOctave; - m_nCurrentOffset = m_nNextOffset; - SetAdd(); - } -} - -void CSAAFreq::UpdateOctaveOffsetData(void) -{ - // loads the buffered new octave and new offset data into the current registers - // and sets up the new frequency for this frequency generator (i.e. sets up m_nAdd) - // - called during Sync, and called when waveform half-cycle completes - - // How the SAA-1099 really treats new data: - // if only new octave data is present, - // then set new period based on just the octave data - // Otherwise, if only new offset data is present, - // then set new period based on just the offset data - // Otherwise, if new octave data is present, and new offset data is present, - // and the offset data was set BEFORE the octave data, - // then set new period based on both the octave and offset data - // Else, if the offset data came AFTER the new octave data - // then set new period based on JUST THE OCTAVE DATA, and continue - // signalling the offset data as 'new', so it will be acted upon - // next half-cycle - // - // Weird, I know. But that's how it works. Philips even documented as much. - - if (!m_bNewData) - { - // optimise for the most common case! No new data! - return; - } - - m_nCurrentOctave=m_nNextOctave; - if (!m_bIgnoreOffsetData) - { - m_nCurrentOffset=m_nNextOffset; - m_bNewData=false; - } - m_bIgnoreOffsetData=false; - - SetAdd(); -} - -void CSAAFreq::_SetSampleRate(unsigned int nSampleRate) -{ - m_nSampleRate = nSampleRate; -} - -void CSAAFreq::_SetOversample(unsigned int oversample) -{ - // oversample is a power of 2 i.e. - // if oversample == 2 then 4x oversample - // if oversample == 6 then 64x oversample - if (oversample < m_nOversample) - { - m_nCounter_low <<= (m_nOversample - oversample); - } - else - { - m_nCounter_low >>= (oversample - m_nOversample); - } - - m_nCounterLimit_low = 1<= (m_nSampleRate<<12)) - { - m_nCounter -= (m_nSampleRate<<12); - m_nCounter_low++; - if (m_nCounter_low >= m_nCounterLimit_low) - { - // period elapsed for (at least) one half-cycle of - // current frequency - m_nCounter_low = 0; - // flip state - from 0 to 1 or vice versa - m_nLevel = 1 - m_nLevel; - - // trigger any connected devices - switch (m_nConnectedMode) - { - case 1: - // env trigger - m_pcConnectedEnvGenerator->InternalClock(); - break; - - case 2: - // noise trigger - m_pcConnectedNoiseGenerator->Trigger(); - break; - - default: - // do nothing - break; - } - - // get new frequency (set period length m_nAdd) if new data is waiting: - UpdateOctaveOffsetData(); - } - } - - return m_nLevel; -} - -void CSAAFreq::SetAdd(void) -{ - // nOctave between 0 and 7; nOffset between 0 and 255 - - // Used to be: - // m_nAdd = (15625 << nOctave) / (511 - nOffset); - // Now just table lookup: - m_nAdd = m_FreqTable[m_nCurrentOctave<<8 | m_nCurrentOffset]; -} - -void CSAAFreq::Sync(bool bSync) -{ - m_bSync = bSync; - - // update straightaway if m_bSync - if (m_bSync) - { - m_nCounter = 0; - m_nCounter_low = 0; - - // this seems to need to be required to make the Fred59 SPACE DEMO audio work correctly - m_nLevel = INITIAL_LEVEL; - - m_nCurrentOctave=m_nNextOctave; - m_nCurrentOffset=m_nNextOffset; - SetAdd(); - } -} diff --git a/src/sound/saasound/SAAFreq.dat b/src/sound/saasound/SAAFreq.dat deleted file mode 100755 index 04fb9081a..000000000 --- a/src/sound/saasound/SAAFreq.dat +++ /dev/null @@ -1,141 +0,0 @@ -/* -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// Precalculated oscillator frequency period steps -// Higher scaling for better accuracy. -// -// After construction, it's important to SetSampleRate before -// trying to use the generator. -// (Just because the CSAANoise object has a default samplerate -// doesn't mean you should rely on it) -// -////////////////////////////////////////////////////////////////////// -*/ - 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57087108 , 57286713 , 57487719 , 57690141 , 57893993 , 58099291 , 58306050 , 58514286 , 58724014 , 58935252 , 59148014 , 59362319 , 59578182 , 59795620 , 60014652 , 60235294 , - 60457565 , 60681481 , 60907063 , 61134328 , 61363296 , 61593985 , 61826415 , 62060606 , 62296578 , 62534351 , 62773946 , 63015385 , 63258687 , 63503876 , 63750973 , 64000000 diff --git a/src/sound/saasound/SAAFreq.h b/src/sound/saasound/SAAFreq.h deleted file mode 100755 index 478754621..000000000 --- a/src/sound/saasound/SAAFreq.h +++ /dev/null @@ -1,72 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAFreq.h: interface for the CSAAFreq class. -// Note about Samplerates: 0=44100, 1=22050; 2=11025 -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAAFREQ_H_INCLUDE -#define SAAFREQ_H_INCLUDE - -#include "defns.h" - -class CSAAFreq -{ -private: -#ifdef SAAFREQ_FIXED_CLOCKRATE - // 'load in' the data for the static frequency lookup table - // precomputed for a fixed clockrate - // See: tools/freqdat.py - const static unsigned long m_FreqTable[2048]; -#else - // we'll calculate the frequency lookup table at runtime. - static unsigned long m_FreqTable[2048]; - static unsigned long m_nClockRate; -#endif - - unsigned long m_nCounter; - unsigned long m_nAdd; - unsigned long m_nCounter_low; - unsigned int m_nOversample; - unsigned long m_nCounterLimit_low; - int m_nLevel; - - int m_nCurrentOffset; - int m_nCurrentOctave; - int m_nNextOffset; - int m_nNextOctave; - bool m_bIgnoreOffsetData; - bool m_bNewData; - bool m_bSync; - - unsigned long m_nSampleRate; - CSAANoise * const m_pcConnectedNoiseGenerator; - CSAAEnv * const m_pcConnectedEnvGenerator; - const int m_nConnectedMode; // 0 = nothing; 1 = envgenerator; 2 = noisegenerator - - void UpdateOctaveOffsetData(void); - void SetAdd(void); - -public: - CSAAFreq(CSAANoise * const pcNoiseGenerator, CSAAEnv * const pcEnvGenerator); - ~CSAAFreq(); - void SetFreqOffset(BYTE nOffset); - void SetFreqOctave(BYTE nOctave); - void _SetSampleRate(unsigned int nSampleRate); - void _SetOversample(unsigned int oversample); - void _SetClockRate(int nClockRate); - void Sync(bool bSync); - int Tick(void); - int Level(void) const; - -}; - -inline int CSAAFreq::Level(void) const -{ - if (m_bSync) - return 1; - - return m_nLevel; -} - -#endif // SAAFREQ_H_INCLUDE diff --git a/src/sound/saasound/SAAImpl.cpp b/src/sound/saasound/SAAImpl.cpp deleted file mode 100644 index f136eefc6..000000000 --- a/src/sound/saasound/SAAImpl.cpp +++ /dev/null @@ -1,489 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAAImpl.cpp: implementation of the CSAASound class. -// the bones of the 'virtual SAA-1099' emulation -// -// the actual sound generation is carried out in the other classes; -// this class provides the output stage and the external interface only -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" - -#include "types.h" -#include "SAAImpl.h" -#include "defns.h" - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAASoundInternal::CSAASoundInternal() - : -m_chip(), -m_uParam(0), -m_uParamRate(0), -m_nClockRate(EXTERNAL_CLK_HZ), -m_nSampleRate(SAMPLE_RATE_HZ), -m_nOversample(DEFAULT_OVERSAMPLE), -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) -m_bHighpass(false), -m_nDebugSample(0) -#else -m_bHighpass(false) -#endif -{ -#ifdef USE_CONFIG_FILE - m_Config.ReadConfig(); -#endif - -#if defined(DEBUGSAA) - m_dbgfile.open(_T(DEBUG_SAA_REGISTER_LOG), std::ios_base::out); - m_pcmfile.open(_T(DEBUG_SAA_PCM_LOG), std::ios_base::out | std::ios_base::binary); -#elif defined(USE_CONFIG_FILE) - if (m_Config.m_bGenerateRegisterLogs) - m_dbgfile.open(m_Config.m_strRegisterLogPath, std::ios_base::out); - if (m_Config.m_bGeneratePcmLogs) - m_pcmfile.open(m_Config.m_strPcmOutputPath, std::ios_base::out | std::ios_base::binary); - - if (m_Config.m_bGeneratePcmLogs && m_Config.m_bGeneratePcmSeparateChannels) - { - for (int i = 0; i < 6; i++) - { - m_channel_pcmfile[i].open(m_Config.getChannelPcmOutputPath(i), std::ios_base::out | std::ios_base::binary); - } - } - - -#endif - // set parameters - // TODO support defaults and overrides from config file - // m_chip.SetSoundParameters(SAAP_FILTER | SAAP_11025 | SAAP_8BIT | SAAP_MONO); - // reset the virtual SAA - // m_chip.Clear(); - - m_chip._SetClockRate(m_nClockRate); - m_chip._SetOversample(m_nOversample); -} - -CSAASoundInternal::~CSAASoundInternal() -{ - // -} - -////////////////////////////////////////////////////////////////////// -// CSAASound members -////////////////////////////////////////////////////////////////////// - -void CSAASoundInternal::SetClockRate(unsigned int nClockRate) -{ - m_nClockRate = nClockRate; - m_chip._SetClockRate(m_nClockRate); -} - -void CSAASoundInternal::Clear(void) -{ - // reinitialises virtual SAA: - // sets reg 28 to 0x02; - sync and disabled - // sets regs 00-31 (except 28) to 0x00; - // sets reg 28 to 0x00; - // sets current reg to 0 - WriteAddressData(28,2); - for (int i=31; i>=0; i--) - { - if (i!=28) WriteAddressData(i,0); - } - WriteAddressData(28,0); - WriteAddress(0); -} - -void CSAASoundInternal::WriteData(BYTE nData) -{ - // originated from an OUT 255,d call - m_chip._WriteData(nData); -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) -#ifdef USE_CONFIG_FILE - if (m_Config.m_bGenerateRegisterLogs) - { -#endif - m_dbgfile << m_nDebugSample << " " << (int)m_chip._ReadAddress() << ":" << (int)nData << std::endl; -#ifdef USE_CONFIG_FILE - } -#endif -#endif -} - -void CSAASoundInternal::WriteAddress(BYTE nReg) -{ - // originated from an OUT 511,r call - m_chip._WriteAddress(nReg); -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) -#ifdef USE_CONFIG_FILE - if (m_Config.m_bGenerateRegisterLogs) - { -#endif - m_dbgfile << m_nDebugSample << " " << (int)nReg << ":"; - if (nReg==24) - { - m_dbgfile << ""; - } - else if (nReg==25) - { - m_dbgfile << ""; - } - m_dbgfile << std::endl; -#ifdef USE_CONFIG_FILE - } -#endif -#endif -} - -void CSAASoundInternal::WriteAddressData(BYTE nReg, BYTE nData) -{ - // performs WriteAddress(nReg) followed by WriteData(nData) - m_chip._WriteAddress(nReg); - m_chip._WriteData(nData); -} - -#if 1 -BYTE CSAASoundInternal::ReadAddress(void) -{ - // Not a real hardware function of the SAA-1099, which is write-only - return(m_chip._ReadAddress()); -} -#else -BYTE CSAASoundInternal::ReadAddress(void) -{ - // Not a real hardware function of the SAA-1099, which is write-only - return(0); -} -#endif - -void CSAASoundInternal::SetSoundParameters(SAAPARAM uParam) -{ - // set samplerate properties from uParam (deprecated but still supported) - unsigned int nSampleRate = m_nSampleRate; - switch (uParam & SAAP_MASK_SAMPLERATE) - { - case SAAP_44100: - nSampleRate = 44100; - m_uParamRate = (m_uParamRate & ~SAAP_MASK_SAMPLERATE) | SAAP_44100; - break; - case SAAP_22050: - nSampleRate = 22050; - m_uParamRate = (m_uParamRate & ~SAAP_MASK_SAMPLERATE) | SAAP_22050; - break; - case SAAP_11025: - nSampleRate = 11025; - m_uParamRate = (m_uParamRate & ~SAAP_MASK_SAMPLERATE) | SAAP_11025; - break; - case 0:// change nothing! - default: - break; - } - - if (nSampleRate != m_nSampleRate) - { - m_nSampleRate = nSampleRate; - m_chip._SetSampleRate(m_nSampleRate); - } - - // set filter properties from uParam - m_uParam = (m_uParam & ~SAAP_MASK_FILTER) | (uParam & SAAP_MASK_FILTER); - - m_bHighpass=true; -} - -void CSAASoundInternal::SetSampleRate(unsigned int nSampleRate) -{ - if (nSampleRate != m_nSampleRate) - { - m_nSampleRate = nSampleRate; - m_chip._SetSampleRate(m_nSampleRate); - } -} - -void CSAASoundInternal::SetOversample(unsigned int nOversample) -{ - if (nOversample != m_nOversample) - { - m_nOversample = nOversample; - m_chip._SetOversample(m_nOversample); - } -} - -SAAPARAM CSAASoundInternal::GetCurrentSoundParameters(void) -{ - return m_uParam | m_uParamRate; -} - -unsigned short CSAASoundInternal::GetCurrentBytesPerSample(void) -{ - // 16 bit stereo => 4 bytes per sample - return 4; -} - -/*static*/ unsigned short CSAASound::GetBytesPerSample(SAAPARAM uParam) -{ - // 16 bit stereo => 4 bytes per sample - switch (uParam & (SAAP_MASK_CHANNELS | SAAP_MASK_BITDEPTH)) - { - case SAAP_STEREO | SAAP_16BIT: - return 4; - default: - return 0; - } -} - -unsigned long CSAASoundInternal::GetCurrentSampleRate(void) -{ - return CSAASound::GetSampleRate(m_uParamRate); -} - -/*static*/ unsigned long CSAASound::GetSampleRate(SAAPARAM uParam) // static member function -{ - switch (uParam & SAAP_MASK_SAMPLERATE) - { - case SAAP_11025: - return 11025; - case SAAP_22050: - return 22050; - case SAAP_44100: - return 44100; - default: - return 0; - } -} - -#if defined(USE_CONFIG_FILE) || (defined(DEFAULT_BOOST) && DEFAULT_BOOST>1) -#define DO_BOOST -#endif - -void scale_for_output(unsigned int left_input, unsigned int right_input, - double oversample_scalar, bool highpass, double boost, - double& filterout_z1_left, double& filterout_z1_right, - BYTE* &pBuffer) -{ - double float_left = (double)left_input; - double float_right = (double)right_input; - float_left /= oversample_scalar; - float_right /= oversample_scalar; - - // scale output into good range - float_left *= DEFAULT_UNBOOSTED_MULTIPLIER; - float_right *= DEFAULT_UNBOOSTED_MULTIPLIER; - - if (highpass) - { - /* cutoff = 5 Hz (say) - const double b1 = exp(-2.0 * M_PI * (Fc/Fs)) - const double a0 = 1.0 - b1; - */ - const double b1 = 0.99928787; - const double a0 = 1.0 - b1; - - filterout_z1_left = float_left * a0 + filterout_z1_left * b1; - filterout_z1_right = float_right * a0 + filterout_z1_right * b1; - float_left -= filterout_z1_left; - float_right -= filterout_z1_right; - } - - // multiply by boost, if defined -#if defined(DO_BOOST) - float_left *= boost; - float_right *= boost; -#endif - // convert to 16-bit signed range with hard clipping - signed short left_output = (signed short)(float_left > 32767 ? 32767 : float_left < -32768 ? -32768 : float_left); - signed short right_output = (signed short)(float_right > 32767 ? 32767 : float_right < -32768 ? -32768 : float_right); - - *pBuffer++ = left_output & 0x00ff; - *pBuffer++ = (left_output >> 8) & 0x00ff; - *pBuffer++ = right_output & 0x00ff; - *pBuffer++ = (right_output >> 8) & 0x00ff; -} - -void CSAASoundInternal::GenerateMany(BYTE* pBuffer, unsigned long nSamples) -{ - unsigned int left_mixed, right_mixed; - static double filterout_z1_left_mixed = 0, filterout_z1_right_mixed = 0; - -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) - BYTE* pBufferStart = pBuffer; - unsigned long nTotalSamples = nSamples; -#endif - -#if defined(DO_BOOST) -#if defined(USE_CONFIG_FILE) - double nBoost = m_Config.m_nBoost; -#else - double nBoost = DEFAULT_BOOST; -#endif -#else - double nBoost = 1.0; -#endif - - double oversample = double(1 << m_nOversample); - -#if defined(USE_CONFIG_FILE) - static double filterout_z1_left_0 = 0, filterout_z1_right_0 = 0; - static double filterout_z1_left_1 = 0, filterout_z1_right_1 = 0; - static double filterout_z1_left_2 = 0, filterout_z1_right_2 = 0; - static double filterout_z1_left_3 = 0, filterout_z1_right_3 = 0; - static double filterout_z1_left_4 = 0, filterout_z1_right_4 = 0; - static double filterout_z1_left_5 = 0, filterout_z1_right_5 = 0; - - if (m_Config.m_bGeneratePcmLogs && m_Config.m_bGeneratePcmSeparateChannels) - { - unsigned int left0, right0, left1, right1, left2, right2, left3, right3, left4, right4, left5, right5; - BYTE* pChannelBufferPtr[6] = { m_pChannelBuffer[0], m_pChannelBuffer[1], m_pChannelBuffer[2], m_pChannelBuffer[3], m_pChannelBuffer[4], m_pChannelBuffer[5] }; - - while (nSamples--) - { - m_chip._TickAndOutputSeparate(left_mixed, right_mixed, - left0, right0, - left1, right1, - left2, right2, - left3, right3, - left4, right4, - left5, right5); - scale_for_output(left_mixed, right_mixed, oversample, m_bHighpass, nBoost, filterout_z1_left_mixed, filterout_z1_right_mixed, pBuffer); - - // and the separate channels - scale_for_output(left0, right0, oversample, m_bHighpass, nBoost, filterout_z1_left_0, filterout_z1_right_0, pChannelBufferPtr[0]); - scale_for_output(left1, right1, oversample, m_bHighpass, nBoost, filterout_z1_left_1, filterout_z1_right_1, pChannelBufferPtr[1]); - scale_for_output(left2, right2, oversample, m_bHighpass, nBoost, filterout_z1_left_2, filterout_z1_right_2, pChannelBufferPtr[2]); - scale_for_output(left3, right3, oversample, m_bHighpass, nBoost, filterout_z1_left_3, filterout_z1_right_3, pChannelBufferPtr[3]); - scale_for_output(left4, right4, oversample, m_bHighpass, nBoost, filterout_z1_left_4, filterout_z1_right_4, pChannelBufferPtr[4]); - scale_for_output(left5, right5, oversample, m_bHighpass, nBoost, filterout_z1_left_5, filterout_z1_right_5, pChannelBufferPtr[5]); - - // flush channel output PCM buffers when full - if (pChannelBufferPtr[0] >= m_pChannelBuffer[0] + CHANNEL_BUFFER_SIZE) - { - for (int i = 0; i < 6; i++) - { - m_channel_pcmfile[i].write((const char*)m_pChannelBuffer[i], CHANNEL_BUFFER_SIZE); - pChannelBufferPtr[i] = m_pChannelBuffer[i]; - } - } - } - // flush remaining channel PCM output data - if (pChannelBufferPtr[0] >= m_pChannelBuffer[0]) - { - for (int i = 0; i < 6; i++) - { - m_channel_pcmfile[i].write((const char*)m_pChannelBuffer[i], pChannelBufferPtr[i]-m_pChannelBuffer[i]); - } - } - } - else - { -#endif - while (nSamples--) - { - m_chip._TickAndOutputStereo(left_mixed, right_mixed); - scale_for_output(left_mixed, right_mixed, oversample, m_bHighpass, nBoost, filterout_z1_left_mixed, filterout_z1_right_mixed, pBuffer); - } - -#if defined(USE_CONFIG_FILE) - } -#endif - -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) -#ifdef USE_CONFIG_FILE - if (m_Config.m_bGeneratePcmLogs) - { -#endif - m_pcmfile.write((const char *)pBufferStart, nTotalSamples * (unsigned long)GetCurrentBytesPerSample()); - m_nDebugSample += nTotalSamples; -#ifdef USE_CONFIG_FILE - } -#endif - -#endif -} - -/////////////////////////////////////////////////////// - -LPCSAASOUND SAAAPI CreateCSAASound(void) -{ - return (new CSAASoundInternal); -} - -void SAAAPI DestroyCSAASound(LPCSAASOUND object) -{ - delete (object); -} - - -/* thoughts on lowpass filtering as part of oversampling. -I tried this and really it didn't seem to make a lot of (audible) difference. - -// lowpass oversample filter adds complexity and not particularly audibly better than simple averaging. -// use_lowpass_oversample_filter_average_output adds an additional averaging step to the output of the oversample -// filter. this seems critical, because without this, the raw output of the lowpass filter is full of aliases -// If use_lowpass_oversample_filter is False, then the _average_output flag is ignored. -// Default, use_lowpass_oversample_filter is False, it sounds just fine really. - -//#define USE_LOWPASS_OVERSAMPLE_FILTER -#undef USE_LOWPASS_OVERSAMPLE_FILTER -//#define USE_LOWPASS_OVERSAMPLE_FILTER_AVERAGE_OUTPUT -#undef USE_LOWPASS_OVERSAMPLE_FILTER_AVERAGE_OUTPUT - -#ifdef USE_LOWPASS_OVERSAMPLE_FILTER -static double oversample_lp_filterout_z1_left_stages[10] = { 0,0,0,0,0,0,0,0,0,0 }; -static double oversample_lp_filterout_z1_right_stages[10] = { 0,0,0,0,0,0,0,0,0,0 }; -double averaged_filterout_left = 0.0, averaged_filterout_right = 0.0; -const int nStages = 10; -for (int i = 0; i < 1 << m_nOversample; i++) -{ - Noise[0]->Tick(); - Noise[1]->Tick(); - f_left = f_right = 0; - for (int c = 0; c < 6; c++) - { - Amp[c]->TickAndOutputStereo(temp_left, temp_right); - f_left += (double)temp_left; - f_right += (double)temp_right; - } - // apply lowpass here. - // HACK: ASSUME m_nOversample is 64 (I was experimenting only using the 64x oversample anyway) - // therefore Fs = 44100*64 - // let's set Fc = 10kHz - // so Fc/Fs = 0.00354308390022675736961451247166 - // const double b1 = exp(-2.0 * M_PI * (Fc/Fs)) - // const double a0 = 1.0 - b1; - // const double b1 = 0.9779841137335348363722276130195; - const double b1 = 0.977; - const double a0 = 1.0 - b1; - - oversample_lp_filterout_z1_left_stages[0] = f_left * a0 + oversample_lp_filterout_z1_left_stages[0] * b1; - for (int stage = 1; stage < nStages; stage++) - oversample_lp_filterout_z1_left_stages[stage] = oversample_lp_filterout_z1_left_stages[stage - 1] * a0 + oversample_lp_filterout_z1_left_stages[stage] * b1; - oversample_lp_filterout_z1_right_stages[0] = f_right * a0 + oversample_lp_filterout_z1_right_stages[0] * b1; - for (int stage = 1; stage < nStages; stage++) - oversample_lp_filterout_z1_right_stages[stage] = oversample_lp_filterout_z1_right_stages[stage - 1] * a0 + oversample_lp_filterout_z1_right_stages[stage] * b1; - -#ifdef USE_LOWPASS_OVERSAMPLE_FILTER_AVERAGE_OUTPUT - averaged_filterout_left += oversample_lp_filterout_4z1_left; - averaged_filterout_right += oversample_lp_filterout_4z1_right; -#endif -} - -// by the end of this loop we will have computed the oversample lowpass filter m_nOversample times -// and yielded exactly ONE sample output. -#ifdef USE_LOWPASS_OVERSAMPLE_FILTER_AVERAGE_OUTPUT -f_left = averaged_filterout_left / (1 << m_nOversample); -f_right = averaged_filterout_right / (1 << m_nOversample); -#else -f_left = oversample_lp_filterout_z1_left_stages[nStages - 1]; -f_right = oversample_lp_filterout_z1_right_stages[nStages - 1]; -#endif - -#else - // do the simple 1/N averaging which is easier and sounds good enough - -#endif - -*/ - diff --git a/src/sound/saasound/SAAImpl.h b/src/sound/saasound/SAAImpl.h deleted file mode 100755 index 61fa79c58..000000000 --- a/src/sound/saasound/SAAImpl.h +++ /dev/null @@ -1,75 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// This is the internal implementation (header file) of the SAASound object. -// This is done so that the external interface to the object always stays the same -// (SAASound.h) even though the internal object can change -// .. Meaning future releases don't require relinking everyone elses code against -// the updated saasound stuff -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAAIMPL_H_INCLUDED -#define SAAIMPL_H_INCLUDED - -#include "SAASound.h" -#include "SAADevice.h" -#ifdef USE_CONFIG_FILE -#include "SAAConfig.h" -#endif - -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) -#include -#include -#include - -#if defined(USE_CONFIG_FILE) -const int CHANNEL_BUFFER_SIZE=1024; -#endif -#endif - -class CSAASoundInternal : public CSAASound -{ -private: - CSAADevice m_chip; - int m_uParam, m_uParamRate; - unsigned int m_nClockRate; - unsigned int m_nSampleRate; - unsigned int m_nOversample; - bool m_bHighpass; -#ifdef USE_CONFIG_FILE - SAAConfig m_Config; -#endif -#if defined(DEBUGSAA) || defined(USE_CONFIG_FILE) - unsigned long m_nDebugSample; - std::ofstream m_dbgfile, m_pcmfile; -#if defined(USE_CONFIG_FILE) - std::ofstream m_channel_pcmfile[6]; - BYTE m_pChannelBuffer[6][CHANNEL_BUFFER_SIZE]; -#endif -#endif - -public: - CSAASoundInternal(); - ~CSAASoundInternal(); - - void SetClockRate(unsigned int nClockRate); - void SetSampleRate(unsigned int nClockRate); - void SetOversample(unsigned int nOversample); - void SetSoundParameters(SAAPARAM uParam); - void WriteAddress(BYTE nReg); - void WriteData(BYTE nData); - void WriteAddressData(BYTE nReg, BYTE nData); - BYTE ReadAddress(void); - void Clear(void); - - SAAPARAM GetCurrentSoundParameters(void); - unsigned long GetCurrentSampleRate(void); - static unsigned long GetSampleRate(SAAPARAM uParam); - unsigned short GetCurrentBytesPerSample(void); - static unsigned short GetBytesPerSample(SAAPARAM uParam); - - void GenerateMany(BYTE * pBuffer, unsigned long nSamples); - -}; - -#endif // SAAIMPL_H_INCLUDED diff --git a/src/sound/saasound/SAANoise.cpp b/src/sound/saasound/SAANoise.cpp deleted file mode 100755 index 1cf3458dd..000000000 --- a/src/sound/saasound/SAANoise.cpp +++ /dev/null @@ -1,180 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAANoise.cpp: implementation of the CSAANoise class. -// One noise generator -// -// After construction, it's important to SetSampleRate before -// trying to use the generator. -// (Just because the CSAANoise object has a default samplerate -// doesn't mean you should rely on it) -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" - -#include "types.h" -#include "SAANoise.h" -#include "defns.h" - - -////////////////////////////////////////////////////////////////////// -// Construction/Destruction -////////////////////////////////////////////////////////////////////// - -CSAANoise::CSAANoise() -: -m_nCounter(0), -m_nCounter_low(0), -m_nOversample(0), -m_nCounterLimit_low(1), -m_bSync(false), -m_nSampleRate(SAMPLE_RATE_HZ), -m_nSourceMode(0), -m_nRand(1) -{ - _SetClockRate(EXTERNAL_CLK_HZ); - m_nAdd = m_nAddBase; -} - -CSAANoise::CSAANoise(unsigned long seed) -: -m_nCounter(0), -m_nCounter_low(0), -m_nOversample(0), -m_nCounterLimit_low(1), -m_bSync(false), -m_nSampleRate(SAMPLE_RATE_HZ), -m_nSourceMode(0), -m_nRand(seed) -{ - _SetClockRate(EXTERNAL_CLK_HZ); - m_nAdd = m_nAddBase; -} - -CSAANoise::~CSAANoise() -{ - // Nothing to do -} - -void CSAANoise::_SetClockRate(int nClockRate) -{ - // at 8MHz the clock rate is 31.250kHZ - // This is simply the clock rate divided by 256 i.e. 2^8 - // We then shift this by 2^12 (like the Freq) for better - // period accuracy. So that's the same as shifting by (12-8) - m_nAddBase = nClockRate << (12 - 8); -} - -void CSAANoise::Seed(unsigned long seed) -{ - m_nRand = seed; -} - -void CSAANoise::SetSource(int nSource) -{ - m_nSourceMode = nSource; - m_nAdd = m_nAddBase >> m_nSourceMode; -} - -void CSAANoise::Trigger(void) -{ - // Trigger only does anything useful when we're - // clocking from the frequency generator - i.e - // if bUseFreqGen = true (i.e. SourceMode = 3) - - // So if we're clocking from the noise generator - // clock (ie, SourceMode = 0, 1 or 2) then do nothing - -// No point actually checking m_bSync here ... because if sync is true, -// then frequency generators won't actually be generating Trigger pulses -// so we wouldn't even get here! - // EXCEPT - cool edge case: if sync is set, then actually the Noise Generator - // is triggered on EVERY CLOCK PULSE (i.e. 8MHz noise). So indeed it is correct - // to not check for sync here. NEEDS TEST CASE. - - if (m_nSourceMode == 3) - { - ChangeLevel(); - } -} - -void CSAANoise::Tick(void) -{ - // Tick only does anything useful when we're - // clocking from the noise generator clock - // (ie, SourceMode = 0, 1 or 2) - - // So, if SourceMode = 3 (ie, we're clocking from a - // frequency generator ==> bUseFreqGen = true) - // then do nothing - if ( (!m_bSync) && (m_nSourceMode!=3) ) - { - m_nCounter += m_nAdd; - while (m_nCounter >= (m_nSampleRate<<12)) - { - m_nCounter -= (m_nSampleRate<<12); - m_nCounter_low++; - if (m_nCounter_low >= m_nCounterLimit_low) - { - m_nCounter_low = 0; - ChangeLevel(); - } - } - } -} - -void CSAANoise::Sync(bool bSync) -{ - if (bSync) - { - m_nCounter = 0; - m_nCounter_low = 0; - } - m_bSync = bSync; -} - - -void CSAANoise::_SetSampleRate(int nSampleRate) -{ - m_nSampleRate = nSampleRate; -} - - -void CSAANoise::_SetOversample(unsigned int oversample) -{ - // oversample is a power of 2 i.e. - // if oversample == 2 then 4x oversample - // if oversample == 6 then 64x oversample - if (oversample < m_nOversample) - { - m_nCounter_low <<= (m_nOversample - oversample); - } - else - { - m_nCounter_low >>= (oversample - m_nOversample); - } - - m_nCounterLimit_low = 1<> 1) ^ 0x20400; - } - else - { - m_nRand >>= 1; - } -} diff --git a/src/sound/saasound/SAANoise.h b/src/sound/saasound/SAANoise.h deleted file mode 100755 index 61a65dee8..000000000 --- a/src/sound/saasound/SAANoise.h +++ /dev/null @@ -1,54 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAANoise.h: interface for the CSAANoise class. -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAANOISE_H_INCLUDED -#define SAANOISE_H_INCLUDED - -class CSAANoise -{ -private: - unsigned long m_nCounter; - unsigned long m_nAdd; - unsigned long m_nCounter_low; - unsigned int m_nOversample; - unsigned long m_nCounterLimit_low; - bool m_bSync; // see description of "SYNC" bit of register 28 - unsigned long m_nSampleRate; // = 44100 when RateMode=0, for example - int m_nSourceMode; - unsigned long m_nAddBase; // nAdd for 31.25 kHz noise at 44.1 kHz samplerate - - // pseudo-random number generator - unsigned long m_nRand; - - void ChangeLevel(void); - - -public: - CSAANoise(); - CSAANoise(unsigned long seed); - ~CSAANoise(); - - void SetSource(int nSource); - void Trigger(void); - void _SetSampleRate(int nSampleRate); - void _SetOversample(unsigned int oversample); - void _SetClockRate(int nClockRate); - void Seed(unsigned long seed); - - void Tick(void); - int Level(void) const; - void Sync(bool bSync); - -}; - -inline int CSAANoise::Level(void) const -{ - // returns 0 or 1 - return (m_nRand & 0x00000001); -} - - -#endif // SAANOISE_H_INCLUDED diff --git a/src/sound/saasound/SAASndC.cpp b/src/sound/saasound/SAASndC.cpp deleted file mode 100755 index 9af0d76e7..000000000 --- a/src/sound/saasound/SAASndC.cpp +++ /dev/null @@ -1,100 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// Thanks to this file (and associated header file) you can now -// use CSAASound from within a standard 'C' program -// -////////////////////////////////////////////////////////////////////// - -#include "SAASound.h" -#include "types.h" -#include "SAAEnv.h" -#include "SAANoise.h" -#include "SAAFreq.h" -#include "SAAAmp.h" -#include "SAASound.h" -#include "SAAImpl.h" - -SAASND SAAAPI newSAASND(void) -{ - return (SAASND)(new CSAASoundInternal()); -} - -void SAAAPI deleteSAASND(SAASND object) -{ - delete (LPCSAASOUND)(object); -} - -void SAAAPI SAASNDSetClockRate(SAASND object, unsigned int nClockRate) -{ - ((LPCSAASOUND)(object))->SetClockRate(nClockRate); -} - -void SAAAPI SAASNDSetSoundParameters(SAASND object, SAAPARAM uParam) -{ - ((LPCSAASOUND)(object))->SetSoundParameters(uParam); -} - -void SAAAPI SAASNDWriteAddress(SAASND object, BYTE nReg) -{ - ((LPCSAASOUND)(object))->WriteAddress(nReg); -} - -void SAAAPI SAASNDWriteData(SAASND object, BYTE nData) -{ - ((LPCSAASOUND)(object))->WriteData(nData); -} - -void SAAAPI SAASNDWriteAddressData(SAASND object, BYTE nReg, BYTE nData) -{ - ((LPCSAASOUND)(object))->WriteAddressData(nReg, nData); -} - -void SAAAPI SAASNDClear(SAASND object) -{ - ((LPCSAASOUND)(object))->Clear(); -} - -SAAPARAM SAAAPI SAASNDGetCurrentSoundParameters(SAASND object) -{ - return ((LPCSAASOUND)(object))->GetCurrentSoundParameters(); -} - -unsigned short SAAAPI SAASNDGetCurrentBytesPerSample(SAASND object) -{ - return ((LPCSAASOUND)(object))->GetCurrentBytesPerSample(); -} - -unsigned short SAAAPI SAASNDGetBytesPerSample(SAAPARAM uParam) -{ - return CSAASound::GetBytesPerSample(uParam); -} - -unsigned long SAAAPI SAASNDGetCurrentSampleRate(SAASND object) -{ - return ((LPCSAASOUND)(object))->GetCurrentSampleRate(); -} - -unsigned long SAAAPI SAASNDGetSampleRate(SAAPARAM uParam) -{ - return CSAASound::GetSampleRate(uParam); -} - -void SAAAPI SAASNDGenerateMany(SAASND object, BYTE * pBuffer, unsigned long nSamples) -{ - ((LPCSAASOUND)(object))->GenerateMany(pBuffer, nSamples); -} - -void SAAAPI SAASNDSetSampleRate(SAASND object, unsigned int nSampleRate) -{ - return ((LPCSAASOUND)(object))->SetSampleRate(nSampleRate); -} - -void SAAAPI SAASNDSetOversample(SAASND object, unsigned int nOversample) -{ - return ((LPCSAASOUND)(object))->SetOversample(nOversample); -} - -BYTE SAAAPI SAASNDReadAddress(SAASND object) -{ - return ((LPCSAASOUND)(object))->ReadAddress(); -} diff --git a/src/sound/saasound/SAASndC.h b/src/sound/saasound/SAASndC.h deleted file mode 100644 index c6fd65765..000000000 --- a/src/sound/saasound/SAASndC.h +++ /dev/null @@ -1,102 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// ********** -// * PUBLIC * -// ********** -// -// SAASndC.h: "C-style" interface for the CSAASound class. -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAASNDC_H_INCLUDED -#define SAASNDC_H_INCLUDED - -#ifdef _MSC_VER -#if _MSC_VER >= 1000 -#pragma once -#endif // _MSC_VER >= 1000 -#endif - -#ifndef SAASOUND_H_INCLUDED - -// Parameters for use with SetSoundParameters, for example, -// SetSoundParameters(SAAP_NOFILTER | SAAP_44100 | SAAP_16BIT | SAAP_STEREO); -#define SAAP_FILTER_HIGHPASS_SIMPLE 0x00000400 -#define SAAP_FILTER_OVERSAMPLE64x 0x00000300 -#define SAAP_FILTER_OVERSAMPLE2x 0x00000200 -#define SAAP_FILTER SAAP_FILTER_OVERSAMPLE2x -#define SAAP_NOFILTER 0x00000100 -#define SAAP_44100 0x00000030 -#define SAAP_22050 0x00000020 -#define SAAP_11025 0x00000010 -#define SAAP_16BIT 0x0000000c -#define SAAP_8BIT 0x00000004 -#define SAAP_STEREO 0x00000003 -#define SAAP_MONO 0x00000001 - -// Bitmasks for use with GetCurrentSoundParameters, for example, -// unsigned long CurrentSampleRateParameter = GetCurrentSoundParameters() -#define SAAP_MASK_FILTER 0x00000f00 -#define SAAP_MASK_FILTER_HIGHPASS 0x00000c00 -#define SAAP_MASK_FILTER_OVERSAMPLE 0x00000300 -#define SAAP_MASK_SAMPLERATE 0x000000030 -#define SAAP_MASK_BITDEPTH 0x0000000c -#define SAAP_MASK_CHANNELS 0x00000003 - -typedef unsigned long SAAPARAM; - - -#ifndef BYTE -#define BYTE unsigned char -#endif - -#ifdef WIN32 -#ifndef WINAPI -#define WINAPI __stdcall -#endif -#define EXTAPI __declspec(dllexport) WINAPI -#else // Win32 -#ifndef WINAPI -#define WINAPI /**/ -#endif -#define EXTAPI /**/ -#endif // Win32 - -#endif // SAASOUND_H_INCLUDED - -typedef void * SAASND; - -// the following are implemented as calls, etc, to a class. - -#ifdef __cplusplus -extern "C" { -#endif - -SAASND EXTAPI newSAASND(void); -void EXTAPI deleteSAASND(SAASND object); - -void EXTAPI SAASNDSetSoundParameters(SAASND object, SAAPARAM uParam); -void EXTAPI SAASNDWriteAddress(SAASND object, BYTE nReg); -void EXTAPI SAASNDWriteData(SAASND object, BYTE nData); -void EXTAPI SAASNDWriteAddressData(SAASND object, BYTE nReg, BYTE nData); -void EXTAPI SAASNDClear(SAASND object); -BYTE EXTAPI SAASNDReadAddress(SAASND object); - -SAAPARAM EXTAPI SAASNDGetCurrentSoundParameters(SAASND object); -unsigned short EXTAPI SAASNDGetCurrentBytesPerSample(SAASND object); -unsigned short EXTAPI SAASNDGetBytesPerSample(SAAPARAM uParam); -unsigned long EXTAPI SAASNDGetCurrentSampleRate(SAASND object); -unsigned long EXTAPI SAASNDGetSampleRate(SAAPARAM uParam); - -void EXTAPI SAASNDGenerateMany(SAASND object, BYTE * pBuffer, unsigned long nSamples); - -void EXTAPI SAASNDSetClockRate(SAASND object, unsigned int nClockRate); -void EXTAPI SAASNDSetSampleRate(SAASND object, unsigned int nSampleRate); -void EXTAPI SAASNDSetOversample(SAASND object, unsigned int nOversample); - - -#ifdef __cplusplus -}; // extern "C" -#endif - -#endif // SAASNDC_H_INCLUDED diff --git a/src/sound/saasound/SAASound.cpp b/src/sound/saasound/SAASound.cpp deleted file mode 100755 index c5e33d862..000000000 --- a/src/sound/saasound/SAASound.cpp +++ /dev/null @@ -1,13 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAASound.cpp - dummy function -// -////////////////////////////////////////////////////////////////////// - -#include - -// Provide something so the compiler doesn't optimise us out of existance -int SomeFunction () -{ - return 42; -} diff --git a/src/sound/saasound/SAASound.h b/src/sound/saasound/SAASound.h deleted file mode 100644 index a5e9265ac..000000000 --- a/src/sound/saasound/SAASound.h +++ /dev/null @@ -1,130 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// SAASound.h: interface for the CSAASound class. -// -// This corresponds to the public (exported) DLL interface, so all -// APIs and client factory methods belong here. -// -// Compatibility notes : the intention is for this to be fully backwards -// compatible across minor and patch versions. Any backwards breaking changes -// should be reflected as a major version increment. New functionality can be added -// in minor versions so long as backwards compatiblity is maintained -// -// Version 3.3.0 (4th Dec 2018) -// -////////////////////////////////////////////////////////////////////// - -#ifndef SAASOUND_H_INCLUDED -#define SAASOUND_H_INCLUDED - -// define this if you want to output diagnostic text and PCM files -//#define DEBUGSAA - -// Parameters for use with SetSoundParameters, for example, -// SetSoundParameters(SAAP_NOFILTER | SAAP_44100 | SAA_16BIT | SAA_STEREO); -// SAAP_FILTER_HIGHPASS_SIMPLE can be ORd with SAAP_FILTER_OVERSAMPLE64x/2x -#define SAAP_FILTER_HIGHPASS_SIMPLE 0x00000400 -#define SAAP_FILTER_OVERSAMPLE64x 0x00000300 -#define SAAP_FILTER_OVERSAMPLE2x 0x00000200 -#define SAAP_FILTER SAAP_FILTER_OVERSAMPLE2x -#define SAAP_NOFILTER 0x00000100 -#define SAAP_44100 0x00000030 -#define SAAP_22050 0x00000020 -#define SAAP_11025 0x00000010 -#define SAAP_16BIT 0x0000000c -#define SAAP_8BIT 0x00000004 -#define SAAP_STEREO 0x00000003 -#define SAAP_MONO 0x00000001 - -// Bitmasks for use with GetCurrentSoundParameters, for example, -// unsigned long CurrentSampleRateParameter = GetCurrentSoundParameters() -#define SAAP_MASK_FILTER 0x00000f00 -#define SAAP_MASK_FILTER_HIGHPASS 0x00000c00 -#define SAAP_MASK_FILTER_OVERSAMPLE 0x00000300 -#define SAAP_MASK_SAMPLERATE 0x000000030 -#define SAAP_MASK_BITDEPTH 0x0000000c -#define SAAP_MASK_CHANNELS 0x00000003 - -typedef unsigned long SAAPARAM; - - -#ifndef BYTE -#define BYTE unsigned char -#endif - -#ifdef _WIN32 -#define SAAAPI _stdcall -#else -#define SAAAPI -#endif - - -#ifdef __cplusplus - -class CSAASound -{ -public: - virtual ~CSAASound() { } - - virtual void SetSoundParameters (SAAPARAM uParam) = 0; - virtual void WriteAddress (BYTE nReg) = 0; - virtual void WriteData (BYTE nData) = 0; - virtual void WriteAddressData (BYTE nReg, BYTE nData) = 0; - virtual void Clear () = 0; - virtual BYTE ReadAddress () = 0; - - virtual SAAPARAM GetCurrentSoundParameters () = 0; - virtual unsigned long GetCurrentSampleRate () = 0; - static unsigned long GetSampleRate (SAAPARAM uParam); - virtual unsigned short GetCurrentBytesPerSample () = 0; - static unsigned short GetBytesPerSample (SAAPARAM uParam); - - virtual void GenerateMany (BYTE * pBuffer, unsigned long nSamples) = 0; - - virtual void SetClockRate(unsigned int nClockRate) = 0; - virtual void SetSampleRate(unsigned int nSampleRate) = 0; - virtual void SetOversample(unsigned int nOversample) = 0; -}; - -typedef class CSAASound * LPCSAASOUND; - -LPCSAASOUND SAAAPI CreateCSAASound(void); -void SAAAPI DestroyCSAASound(LPCSAASOUND object); - -#endif // __cplusplus - - -#ifdef __cplusplus -extern "C" { -#endif - -typedef void * SAASND; - -// "C-style" interface for the CSAASound class -SAASND SAAAPI newSAASND(void); -void SAAAPI deleteSAASND(SAASND object); - -void SAAAPI SAASNDSetSoundParameters(SAASND object, SAAPARAM uParam); -void SAAAPI SAASNDWriteAddress(SAASND object, BYTE nReg); -void SAAAPI SAASNDWriteData(SAASND object, BYTE nData); -void SAAAPI SAASNDWriteAddressData(SAASND object, BYTE nReg, BYTE nData); -void SAAAPI SAASNDClear(SAASND object); - -SAAPARAM SAAAPI SAASNDGetCurrentSoundParameters(SAASND object); -unsigned short SAAAPI SAASNDGetCurrentBytesPerSample(SAASND object); -unsigned short SAAAPI SAASNDGetBytesPerSample(SAAPARAM uParam); -unsigned long SAAAPI SAASNDGetCurrentSampleRate(SAASND object); -unsigned long SAAAPI SAASNDGetSampleRate(SAAPARAM uParam); - -void SAAAPI SAASNDGenerateMany(SAASND object, BYTE * pBuffer, unsigned long nSamples); -void SAAAPI SAASNDSetClockRate(SAASND object, unsigned int nClockRate); -void SAAAPI SAASNDSetSampleRate(SAASND object, unsigned int nSampleRate); -void SAAAPI SAASNDSetOversample(SAASND object, unsigned int nOversample); - -BYTE SAAAPI SAASNDReadAddress(SAASND object); - -#ifdef __cplusplus -}; // extern "C" -#endif - -#endif // SAASOUND_H_INCLUDED diff --git a/src/sound/saasound/defns.h b/src/sound/saasound/defns.h deleted file mode 100644 index e81d1c819..000000000 --- a/src/sound/saasound/defns.h +++ /dev/null @@ -1,59 +0,0 @@ -// Part of SAASound copyright 2020 Dave Hooper -// -// defns.h: compile-time configuration parameters -// -////////////////////////////////////////////////////////////////////// - -#ifndef DEFNS_H_INCLUDED -#define DEFNS_H_INCLUDED - -#define HAVE_CONFIG_H -#ifdef HAVE_CONFIG_H -// using CMAKE -#include "saasound_cmake_config.h" -#else - -// initial default SAA1099 crystal clock rate in HZ (can be changed subsequently by calling SetClockRate) -#define EXTERNAL_CLK_HZ 8000000 - -// define SAAFREQ_FIXED_CLOCKRATE if the above external clock rate is the only supported clock rate -// i.e. only support a single compile-time clock rate (=> this also prevents using the SetClockRate method) -#undef SAAFREQ_FIXED_CLOCKRATE -// #define SAAFREQ_FIXED_CLOCKRATE - -// initial default sample rate (audio samplerate) -#define SAMPLE_RATE_HZ 44100 - -// initial default oversample (audio quality) recommend 0<=oversample<=6 -#define DEFAULT_OVERSAMPLE 6 - -// Whether to dump out a log of all register and value changes and raw output pcm -//#define DEBUGSAA -#undef DEBUGSAA - -// the (default) names of the register output and pcm output log files. -// If you're using a config file, you can change these (or, if you enable -// debugging via the config file settings, but leave the filenames unspecified, -// it will use these defaults) -#define DEBUG_SAA_REGISTER_LOG "debugsaa.txt" -#define DEBUG_SAA_PCM_LOG "debugsaa.pcm" -// Whether to include support for these debug logs via config file (only making -// sense if USE_CONFIG_FILE is also defined) - -// Whether to support a startup configuration file that is parsed at load time -// #undef USE_CONFIG_FILE -#define USE_CONFIG_FILE - -// and if so, what is its location -#ifdef USE_CONFIG_FILE -#define CONFIG_FILE_PATH "SAASound.cfg" -#endif // USE_CONFIG_FILE - -#define DEFAULT_UNBOOSTED_MULTIPLIER 11.35 - -#define DEFAULT_BOOST 1 - - -#endif // HAVE_CONFIG_H - -#endif // DEFNS_H_INCLUDED diff --git a/src/sound/saasound/resource.h b/src/sound/saasound/resource.h deleted file mode 100755 index 0b893bf3a..000000000 --- a/src/sound/saasound/resource.h +++ /dev/null @@ -1,15 +0,0 @@ -//{{NO_DEPENDENCIES}} -// Microsoft Developer Studio generated include file. -// Used by SAASound.rc -// - -// Next default values for new objects -// -#ifdef APSTUDIO_INVOKED -#ifndef APSTUDIO_READONLY_SYMBOLS -#define _APS_NEXT_RESOURCE_VALUE 101 -#define _APS_NEXT_COMMAND_VALUE 40001 -#define _APS_NEXT_CONTROL_VALUE 1000 -#define _APS_NEXT_SYMED_VALUE 101 -#endif -#endif diff --git a/src/sound/saasound/saasound_cmake_config.h b/src/sound/saasound/saasound_cmake_config.h deleted file mode 100644 index da914a71b..000000000 --- a/src/sound/saasound/saasound_cmake_config.h +++ /dev/null @@ -1,14 +0,0 @@ -#pragma once - -#define EXTERNAL_CLK_HZ 7159090 -/* #undef SAAFREQ_FIXED_CLOCKRATE */ -#define SAMPLE_RATE_HZ 44100 -#define DEFAULT_OVERSAMPLE 6 -#define DEFAULT_UNBOOSTED_MULTIPLIER 11.3 -#define DEFAULT_BOOST 1 -/* #undef DEBUGSAA */ -#define DEBUG_SAA_REGISTER_LOG "debugsaa.txt" -#define DEBUG_SAA_PCM_LOG "debugsaa.pcm" - -/* #undef USE_CONFIG_FILE */ -#define CONFIG_FILE_PATH "SAASound.cfg" diff --git a/src/sound/saasound/types.h b/src/sound/saasound/types.h deleted file mode 100755 index 4eb62f485..000000000 --- a/src/sound/saasound/types.h +++ /dev/null @@ -1,34 +0,0 @@ -// Part of SAASound copyright 1998-2018 Dave Hooper -// -// handy typedefs -// -////////////////////////////////////////////////////////////////////// - -#ifndef TYPES_H_INCLUDED -#define TYPES_H_INCLUDED - -#if defined(__i386__) || defined(WIN32) || \ - (defined(__alpha__) || defined(__alpha)) || \ - defined(__arm__) || \ - (defined(__mips__) && defined(__MIPSEL__)) -#else -#define __BIG_ENDIAN -#endif - - -#ifndef NULL -#define NULL 0 -#endif - -typedef struct -{ - int nNumberOfPhases; - bool bLooping; - int nLevels[2][2][16]; // [Resolution][Phase][Withinphase] -} ENVDATA; - -#ifdef WIN32 -extern "C" void _stdcall OutputDebugStringA (char*); -#endif - -#endif diff --git a/src/sound/snd_cms.c b/src/sound/snd_cms.c index c6591b1fc..66dff80f3 100644 --- a/src/sound/snd_cms.c +++ b/src/sound/snd_cms.c @@ -8,7 +8,6 @@ #include <86box/86box.h> #include <86box/device.h> #include <86box/io.h> -#include "saasound/SAASound.h" #include <86box/snd_cms.h> #include <86box/sound.h> #include <86box/plat_unused.h> @@ -16,13 +15,62 @@ void cms_update(cms_t *cms) { - if (cms->pos < wavetable_pos_global) { - SAASNDGenerateMany(cms->saasound, (unsigned char*)&cms->buffer[cms->pos], wavetable_pos_global - cms->pos); - cms->pos = wavetable_pos_global; - } - if (cms->pos2 < wavetable_pos_global) { - SAASNDGenerateMany(cms->saasound2, (unsigned char*)&cms->buffer2[cms->pos2], wavetable_pos_global - cms->pos2); - cms->pos2 = wavetable_pos_global; + for (; cms->pos < sound_pos_global; cms->pos++) { + int16_t out_l = 0; + int16_t out_r = 0; + + for (uint8_t c = 0; c < 4; c++) { + switch (cms->noisetype[c >> 1][c & 1]) { + case 0: + cms->noisefreq[c >> 1][c & 1] = MASTER_CLOCK / 256; + break; + case 1: + cms->noisefreq[c >> 1][c & 1] = MASTER_CLOCK / 512; + break; + case 2: + cms->noisefreq[c >> 1][c & 1] = MASTER_CLOCK / 1024; + break; + case 3: + cms->noisefreq[c >> 1][c & 1] = cms->freq[c >> 1][(c & 1) * 3]; + break; + + default: + break; + } + } + for (uint8_t c = 0; c < 2; c++) { + if (cms->regs[c][0x1C] & 1) { + for (uint8_t d = 0; d < 6; d++) { + if (cms->regs[c][0x14] & (1 << d)) { + if (cms->stat[c][d]) + out_l += (cms->vol[c][d][0] * 90); + if (cms->stat[c][d]) + out_r += (cms->vol[c][d][1] * 90); + cms->count[c][d] += cms->freq[c][d]; + if (cms->count[c][d] >= 24000) { + cms->count[c][d] -= 24000; + cms->stat[c][d] ^= 1; + } + } else if (cms->regs[c][0x15] & (1 << d)) { + if (cms->noise[c][d / 3] & 1) + out_l += (cms->vol[c][d][0] * 90); + if (cms->noise[c][d / 3] & 1) + out_r += (cms->vol[c][d][0] * 90); + } + } + for (uint8_t d = 0; d < 2; d++) { + cms->noisecount[c][d] += cms->noisefreq[c][d]; + while (cms->noisecount[c][d] >= 24000) { + cms->noisecount[c][d] -= 24000; + cms->noise[c][d] <<= 1; + if (!(((cms->noise[c][d] & 0x4000) >> 8) ^ (cms->noise[c][d] & 0x40))) + cms->noise[c][d] |= 1; + } + } + } + } + cms->buffer[cms->pos << 1] = out_l; + cms->buffer[(cms->pos << 1) + 1] = out_r; } } @@ -39,39 +87,63 @@ cms_get_buffer(int32_t *buffer, int len, void *priv) cms->pos = 0; } -void -cms_get_buffer_2(int32_t *buffer, int len, void *priv) -{ - cms_t *cms = (cms_t *) priv; - - cms_update(cms); - - for (int c = 0; c < len * 2; c++) - buffer[c] += cms->buffer2[c]; - - cms->pos2 = 0; -} - void cms_write(uint16_t addr, uint8_t val, void *priv) { cms_t *cms = (cms_t *) priv; + int voice; + int chip = (addr & 2) >> 1; switch (addr & 0xf) { case 0x1: /* SAA #1 Register Select Port */ - SAASNDWriteAddress(cms->saasound, val & 31); + cms->addrs[0] = val & 31; break; case 0x3: /* SAA #2 Register Select Port */ - SAASNDWriteAddress(cms->saasound2, val & 31); + cms->addrs[1] = val & 31; break; case 0x0: /* SAA #1 Data Port */ - cms_update(cms); - SAASNDWriteData(cms->saasound, val); - break; case 0x2: /* SAA #2 Data Port */ cms_update(cms); - SAASNDWriteData(cms->saasound2, val); + cms->regs[chip][cms->addrs[chip] & 31] = val; + switch (cms->addrs[chip] & 31) { + case 0x00: + case 0x01: + case 0x02: /*Volume*/ + case 0x03: + case 0x04: + case 0x05: + voice = cms->addrs[chip] & 7; + cms->vol[chip][voice][0] = val & 0xf; + cms->vol[chip][voice][1] = val >> 4; + break; + case 0x08: + case 0x09: + case 0x0A: /*Frequency*/ + case 0x0B: + case 0x0C: + case 0x0D: + voice = cms->addrs[chip] & 7; + cms->latch[chip][voice] = (cms->latch[chip][voice] & 0x700) | val; + cms->freq[chip][voice] = (MASTER_CLOCK / 512 << (cms->latch[chip][voice] >> 8)) / (511 - (cms->latch[chip][voice] & 255)); + break; + case 0x10: + case 0x11: + case 0x12: /*Octave*/ + voice = (cms->addrs[chip] & 3) << 1; + cms->latch[chip][voice] = (cms->latch[chip][voice] & 0xFF) | ((val & 7) << 8); + cms->latch[chip][voice + 1] = (cms->latch[chip][voice + 1] & 0xFF) | ((val & 0x70) << 4); + cms->freq[chip][voice] = (MASTER_CLOCK / 512 << (cms->latch[chip][voice] >> 8)) / (511 - (cms->latch[chip][voice] & 255)); + cms->freq[chip][voice + 1] = (MASTER_CLOCK / 512 << (cms->latch[chip][voice + 1] >> 8)) / (511 - (cms->latch[chip][voice + 1] & 255)); + break; + case 0x16: /*Noise*/ + cms->noisetype[chip][0] = val & 3; + cms->noisetype[chip][1] = (val >> 4) & 3; + break; + + default: + break; + } break; case 0x6: /* GameBlaster Write Port */ @@ -91,9 +163,9 @@ cms_read(uint16_t addr, void *priv) switch (addr & 0xf) { case 0x1: /* SAA #1 Register Select Port */ - return SAASNDReadAddress(cms->saasound); + return cms->addrs[0]; case 0x3: /* SAA #2 Register Select Port */ - return SAASNDReadAddress(cms->saasound2); + return cms->addrs[1]; case 0x4: /* GameBlaster Read port (Always returns 0x7F) */ return 0x7f; case 0xa: /* GameBlaster Read Port */ @@ -113,12 +185,7 @@ cms_init(UNUSED(const device_t *info)) uint16_t addr = device_get_config_hex16("base"); io_sethandler(addr, 0x0010, cms_read, NULL, NULL, cms_write, NULL, NULL, cms); - cms->saasound = newSAASND(); - SAASNDSetSoundParameters(cms->saasound, SAAP_44100 | SAAP_16BIT | SAAP_NOFILTER | SAAP_STEREO); - cms->saasound2 = newSAASND(); - SAASNDSetSoundParameters(cms->saasound2, SAAP_44100 | SAAP_16BIT | SAAP_NOFILTER | SAAP_STEREO); - wavetable_add_handler(cms_get_buffer, cms); - wavetable_add_handler(cms_get_buffer_2, cms); + sound_add_handler(cms_get_buffer, cms); return cms; } @@ -127,9 +194,6 @@ cms_close(void *priv) { cms_t *cms = (cms_t *) priv; - deleteSAASND(cms->saasound); - deleteSAASND(cms->saasound2); - free(cms); }