mirror of
https://github.com/AppleWin/AppleWin.git
synced 2024-06-25 10:29:37 +00:00
4543117f81
Fixed debugger main "data" window behaviour:
. When activating the main data display ("data" command) the cursor keys wouldn't work until the minidump ("md1") was also enabled. NB. The cursor keys should work in the main data window, independently of whether the minidump is active.
964 lines
29 KiB
C++
964 lines
29 KiB
C++
/*
|
|
AppleWin : An Apple //e emulator for Windows
|
|
|
|
Copyright (C) 1994-1996, Michael O'Brien
|
|
Copyright (C) 1999-2001, Oliver Schmidt
|
|
Copyright (C) 2002-2005, Tom Charlesworth
|
|
Copyright (C) 2006-2007, Tom Charlesworth, Michael Pohoreski
|
|
|
|
AppleWin is free software; you can redistribute it and/or modify
|
|
it under the terms of the GNU General Public License as published by
|
|
the Free Software Foundation; either version 2 of the License, or
|
|
(at your option) any later version.
|
|
|
|
AppleWin is distributed in the hope that it will be useful,
|
|
but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
GNU General Public License for more details.
|
|
|
|
You should have received a copy of the GNU General Public License
|
|
along with AppleWin; if not, write to the Free Software
|
|
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
|
*/
|
|
|
|
/* Description: Speaker emulation
|
|
*
|
|
* Author: Various
|
|
*/
|
|
|
|
#include "StdAfx.h"
|
|
|
|
#include "Applewin.h"
|
|
#include "CPU.h"
|
|
#include "Frame.h"
|
|
#include "Log.h"
|
|
#include "Memory.h"
|
|
#include "SoundCore.h"
|
|
#include "Speaker.h"
|
|
#include "Video.h" // VideoRedrawScreen()
|
|
#include "YamlHelper.h"
|
|
#include "Riff.h"
|
|
|
|
#include "Debugger/Debug.h" // For DWORD extbench
|
|
|
|
// Notes:
|
|
//
|
|
// [OLD: 23.191 Apple CLKs == 44100Hz (CLK_6502/44100)]
|
|
// 23 Apple CLKS per PC sample (played back at 44.1KHz)
|
|
//
|
|
//
|
|
// The speaker's wave output drives how much 6502 emulation is done in real-time, eg:
|
|
// If the speaker's wave buffer is running out of sample-data, then more 6502 cycles
|
|
// need to be executed to top-up the wave buffer.
|
|
// This is in contrast to the AY8910 voices, which can simply generate more data if
|
|
// their buffers are running low.
|
|
//
|
|
|
|
static const unsigned short g_nSPKR_NumChannels = 1;
|
|
static const DWORD g_dwDSSpkrBufferSize = MAX_SAMPLES * sizeof(short) * g_nSPKR_NumChannels;
|
|
|
|
//-------------------------------------
|
|
|
|
static short* g_pSpeakerBuffer = NULL;
|
|
|
|
// Globals (SOUND_WAVE)
|
|
const short SPKR_DATA_INIT = (short)0x8000;
|
|
|
|
short g_nSpeakerData = SPKR_DATA_INIT;
|
|
static UINT g_nBufferIdx = 0;
|
|
|
|
static short* g_pRemainderBuffer = NULL;
|
|
static UINT g_nRemainderBufferSize; // Setup in SpkrInitialize()
|
|
static UINT g_nRemainderBufferIdx; // Setup in SpkrInitialize()
|
|
|
|
// Application-wide globals:
|
|
SoundType_e soundtype = SOUND_WAVE;
|
|
double g_fClksPerSpkrSample; // Setup in SetClksPerSpkrSample()
|
|
|
|
// Allow temporary quietening of speaker (8 bit DAC)
|
|
bool g_bQuieterSpeaker = false;
|
|
|
|
// Globals
|
|
static unsigned __int64 g_nSpkrQuietCycleCount = 0;
|
|
static unsigned __int64 g_nSpkrLastCycle = 0;
|
|
static bool g_bSpkrToggleFlag = false;
|
|
static VOICE SpeakerVoice = {0};
|
|
static bool g_bSpkrAvailable = false;
|
|
|
|
//-----------------------------------------------------------------------------
|
|
|
|
// Forward refs:
|
|
static ULONG Spkr_SubmitWaveBuffer_FullSpeed(short* pSpeakerBuffer, ULONG nNumSamples);
|
|
static ULONG Spkr_SubmitWaveBuffer(short* pSpeakerBuffer, ULONG nNumSamples);
|
|
static void Spkr_SetActive(bool bActive);
|
|
static void Spkr_DSUninit();
|
|
|
|
//=============================================================================
|
|
|
|
static void DisplayBenchmarkResults ()
|
|
{
|
|
DWORD totaltime = GetTickCount()-extbench;
|
|
VideoRedrawScreen();
|
|
TCHAR buffer[64];
|
|
wsprintf(buffer,
|
|
TEXT("This benchmark took %u.%02u seconds."),
|
|
(unsigned)(totaltime / 1000),
|
|
(unsigned)((totaltime / 10) % 100));
|
|
MessageBox(g_hFrameWindow,
|
|
buffer,
|
|
TEXT("Benchmark Results"),
|
|
MB_ICONINFORMATION | MB_SETFOREGROUND);
|
|
}
|
|
|
|
//=============================================================================
|
|
//
|
|
// DC filtering V2 (Riccardo Macri May 2015) (GH#275)
|
|
//
|
|
// To prevent loud clicks on Window's sound buffer underruns and constant DC
|
|
// being sent out to amplifiers (some soundcards are DC coupled) which is
|
|
// not good for them, an attenuator slowly drops the speaker output
|
|
// to 0 after the speaker (or 8 bit DAC) has been idle for a couple hundred
|
|
// milliseconds.
|
|
//
|
|
// The approach works as follows:
|
|
// - SpkrToggle() is called when the speaker state is flipped by accessing $C030
|
|
// - This brings audio up to date then calls ResetDCFilter()
|
|
// - ResetDCFilter() sets a counter to a high value
|
|
// - every audio sample is processed by DCFilter() as follows:
|
|
// - if the counter is >= 32768, the speaker has been recently toggled
|
|
// and the samples are unaffected
|
|
// - if the counter is < 32768 but > 0, it is used to scale the
|
|
// sample to reduce +ve or -ve speaker states towards zero
|
|
// - In the two cases above, the counter is decremented
|
|
// - if the counter is zero, the speaker has been silent for a while
|
|
// and the output is 0 regardless of the speaker state.
|
|
//
|
|
// - the initial "high value" is chosen so 10000/44100 = about a
|
|
// quarter of a second of speaker inactivity is needed before attenuation
|
|
// begins.
|
|
//
|
|
// NOTE: The attenuation is not ever reducing the level of audio, just
|
|
// the DC offset at which the speaker has been left.
|
|
//
|
|
// This approach has zero impact on any speaker tones including PWM
|
|
// due to the samples being unchanged for at least 0.25 seconds after
|
|
// any speaker activity.
|
|
//
|
|
|
|
static UINT g_uDCFilterState = 0;
|
|
|
|
inline void ResetDCFilter(void)
|
|
{
|
|
// reset the attenuator with an additional 250ms of full gain
|
|
// (10000 samples) before it starts attenuating
|
|
g_uDCFilterState = 32768 + 10000;
|
|
}
|
|
|
|
inline short DCFilter(short sample_in)
|
|
{
|
|
if (g_uDCFilterState == 0) // no sound for a while, stay 0
|
|
return 0;
|
|
|
|
if (g_uDCFilterState >= 32768) // full gain after recent sound
|
|
{
|
|
g_uDCFilterState--;
|
|
return sample_in;
|
|
}
|
|
|
|
return (((int)sample_in) * (g_uDCFilterState--)) / 32768; // scale & divide by 32768 (NB. Don't ">>15" as undefined behaviour)
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
static void SetClksPerSpkrSample()
|
|
{
|
|
// // 23.191 clks for 44.1Khz (when 6502 CLK=1.0Mhz)
|
|
// g_fClksPerSpkrSample = g_fCurrentCLK6502 / (double)SPKR_SAMPLE_RATE;
|
|
|
|
// Use integer value: Better for MJ Mahon's RT.SYNTH.DSK (integer multiples of 1.023MHz Clk)
|
|
// . 23 clks @ 1.023MHz
|
|
g_fClksPerSpkrSample = (double) (UINT) (g_fCurrentCLK6502 / (double)SPKR_SAMPLE_RATE);
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
static void InitRemainderBuffer()
|
|
{
|
|
delete [] g_pRemainderBuffer;
|
|
|
|
SetClksPerSpkrSample();
|
|
|
|
g_nRemainderBufferSize = (UINT) g_fClksPerSpkrSample;
|
|
if ((double)g_nRemainderBufferSize < g_fClksPerSpkrSample)
|
|
g_nRemainderBufferSize++;
|
|
|
|
g_pRemainderBuffer = new short [g_nRemainderBufferSize];
|
|
memset(g_pRemainderBuffer, 0, g_nRemainderBufferSize);
|
|
|
|
g_nRemainderBufferIdx = 0;
|
|
}
|
|
|
|
//
|
|
// ----- ALL GLOBALLY ACCESSIBLE FUNCTIONS ARE BELOW THIS LINE -----
|
|
//
|
|
|
|
//=============================================================================
|
|
|
|
void SpkrDestroy ()
|
|
{
|
|
Spkr_DSUninit();
|
|
|
|
//
|
|
|
|
if(soundtype == SOUND_WAVE)
|
|
{
|
|
delete [] g_pSpeakerBuffer;
|
|
delete [] g_pRemainderBuffer;
|
|
|
|
g_pSpeakerBuffer = NULL;
|
|
g_pRemainderBuffer = NULL;
|
|
}
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
void SpkrInitialize ()
|
|
{
|
|
if(g_fh)
|
|
{
|
|
fprintf(g_fh, "Spkr Config: soundtype = %d ", (int) soundtype);
|
|
switch(soundtype)
|
|
{
|
|
case SOUND_NONE: fprintf(g_fh, "(NONE)\n"); break;
|
|
case SOUND_WAVE: fprintf(g_fh, "(WAVE)\n"); break;
|
|
default: fprintf(g_fh, "(UNDEFINED!)\n"); break;
|
|
}
|
|
}
|
|
|
|
if(g_bDisableDirectSound)
|
|
{
|
|
SpeakerVoice.bMute = true;
|
|
}
|
|
else
|
|
{
|
|
g_bSpkrAvailable = Spkr_DSInit();
|
|
}
|
|
|
|
//
|
|
|
|
if (soundtype == SOUND_WAVE)
|
|
{
|
|
InitRemainderBuffer();
|
|
|
|
g_pSpeakerBuffer = new short [SPKR_SAMPLE_RATE]; // Buffer can hold a max of 1 seconds worth of samples
|
|
}
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
// NB. Called when /g_fCurrentCLK6502/ changes
|
|
void SpkrReinitialize ()
|
|
{
|
|
if (soundtype == SOUND_WAVE)
|
|
{
|
|
InitRemainderBuffer();
|
|
}
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
void SpkrReset()
|
|
{
|
|
g_nBufferIdx = 0;
|
|
g_nSpkrQuietCycleCount = 0;
|
|
g_bSpkrToggleFlag = false;
|
|
|
|
InitRemainderBuffer();
|
|
Spkr_SubmitWaveBuffer(NULL, 0);
|
|
Spkr_SetActive(false);
|
|
Spkr_Demute();
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
BOOL SpkrSetEmulationType (HWND window, SoundType_e newtype)
|
|
{
|
|
SpkrDestroy(); // GH#295: Destroy for all types (even SOUND_NONE)
|
|
|
|
soundtype = newtype;
|
|
if (soundtype != SOUND_NONE)
|
|
SpkrInitialize();
|
|
|
|
if (soundtype != newtype)
|
|
switch (newtype) {
|
|
|
|
case SOUND_WAVE:
|
|
MessageBox(window,
|
|
TEXT("The emulator is unable to initialize a waveform ")
|
|
TEXT("output device. Make sure you have a sound card ")
|
|
TEXT("and a driver installed and that windows is ")
|
|
TEXT("correctly configured to use the driver. Also ")
|
|
TEXT("ensure that no other program is currently using ")
|
|
TEXT("the device."),
|
|
TEXT("Configuration"),
|
|
MB_ICONEXCLAMATION | MB_SETFOREGROUND);
|
|
return 0;
|
|
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
static void ReinitRemainderBuffer(UINT nCyclesRemaining)
|
|
{
|
|
if(nCyclesRemaining == 0)
|
|
return;
|
|
|
|
for(g_nRemainderBufferIdx=0; g_nRemainderBufferIdx<nCyclesRemaining; g_nRemainderBufferIdx++)
|
|
g_pRemainderBuffer[g_nRemainderBufferIdx] = g_nSpeakerData;
|
|
|
|
_ASSERT(g_nRemainderBufferIdx < g_nRemainderBufferSize);
|
|
}
|
|
|
|
static void UpdateRemainderBuffer(ULONG* pnCycleDiff)
|
|
{
|
|
if(g_nRemainderBufferIdx)
|
|
{
|
|
while((g_nRemainderBufferIdx < g_nRemainderBufferSize) && *pnCycleDiff)
|
|
{
|
|
g_pRemainderBuffer[g_nRemainderBufferIdx] = g_nSpeakerData;
|
|
g_nRemainderBufferIdx++;
|
|
(*pnCycleDiff)--;
|
|
}
|
|
|
|
if(g_nRemainderBufferIdx == g_nRemainderBufferSize)
|
|
{
|
|
g_nRemainderBufferIdx = 0;
|
|
signed long nSampleMean = 0;
|
|
for(UINT i=0; i<g_nRemainderBufferSize; i++)
|
|
nSampleMean += (signed long) g_pRemainderBuffer[i];
|
|
nSampleMean /= (signed long) g_nRemainderBufferSize;
|
|
|
|
if(g_nBufferIdx < SPKR_SAMPLE_RATE-1)
|
|
g_pSpeakerBuffer[g_nBufferIdx++] = DCFilter( (short)nSampleMean );
|
|
}
|
|
}
|
|
}
|
|
|
|
static void UpdateSpkr()
|
|
{
|
|
if(!g_bFullSpeed || SoundCore_GetTimerState())
|
|
{
|
|
ULONG nCycleDiff = (ULONG) (g_nCumulativeCycles - g_nSpkrLastCycle);
|
|
|
|
UpdateRemainderBuffer(&nCycleDiff);
|
|
|
|
ULONG nNumSamples = (ULONG) ((double)nCycleDiff / g_fClksPerSpkrSample);
|
|
|
|
ULONG nCyclesRemaining = (ULONG) ((double)nCycleDiff - (double)nNumSamples * g_fClksPerSpkrSample);
|
|
|
|
while((nNumSamples--) && (g_nBufferIdx < SPKR_SAMPLE_RATE-1))
|
|
g_pSpeakerBuffer[g_nBufferIdx++] = DCFilter(g_nSpeakerData);
|
|
|
|
ReinitRemainderBuffer(nCyclesRemaining); // Partially fill 1Mhz sample buffer
|
|
}
|
|
|
|
g_nSpkrLastCycle = g_nCumulativeCycles;
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
// Called by emulation code when Speaker I/O reg is accessed
|
|
//
|
|
|
|
BYTE __stdcall SpkrToggle (WORD, WORD, BYTE, BYTE, ULONG nExecutedCycles)
|
|
{
|
|
g_bSpkrToggleFlag = true;
|
|
|
|
if(!g_bFullSpeed)
|
|
Spkr_SetActive(true);
|
|
|
|
//
|
|
|
|
|
|
if (extbench)
|
|
{
|
|
DisplayBenchmarkResults();
|
|
extbench = 0;
|
|
}
|
|
|
|
if (soundtype == SOUND_WAVE)
|
|
{
|
|
CpuCalcCycles(nExecutedCycles);
|
|
|
|
UpdateSpkr();
|
|
|
|
short speakerDriveLevel = SPKR_DATA_INIT;
|
|
if (g_bQuieterSpeaker) // quieten the speaker if 8 bit DAC in use
|
|
speakerDriveLevel /= 4; // NB. Don't shift -ve number right: undefined behaviour (MSDN says: implementation-dependent)
|
|
|
|
// When full-speed: Don't ResetDCFilter(), otherwise get occasional clicks when speaker toggled
|
|
if (!g_bFullSpeed)
|
|
ResetDCFilter();
|
|
|
|
if (g_nSpeakerData == speakerDriveLevel)
|
|
g_nSpeakerData = ~speakerDriveLevel;
|
|
else
|
|
g_nSpeakerData = speakerDriveLevel;
|
|
}
|
|
|
|
return MemReadFloatingBus(nExecutedCycles);
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
// Called by ContinueExecution()
|
|
void SpkrUpdate (DWORD totalcycles)
|
|
{
|
|
#ifdef LOG_PERF_TIMINGS
|
|
extern UINT64 g_timeSpeaker;
|
|
PerfMarker perfMarker(g_timeSpeaker);
|
|
#endif
|
|
|
|
if(!g_bSpkrToggleFlag)
|
|
{
|
|
if(!g_nSpkrQuietCycleCount)
|
|
{
|
|
g_nSpkrQuietCycleCount = g_nCumulativeCycles;
|
|
}
|
|
else if(g_nCumulativeCycles - g_nSpkrQuietCycleCount > (unsigned __int64)g_fCurrentCLK6502/5)
|
|
{
|
|
// After 0.2 sec of Apple time, deactivate spkr voice
|
|
// . This allows emulator to auto-switch to full-speed g_nAppMode for fast disk access
|
|
Spkr_SetActive(false);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
g_nSpkrQuietCycleCount = 0;
|
|
g_bSpkrToggleFlag = false;
|
|
}
|
|
|
|
//
|
|
|
|
if (soundtype == SOUND_WAVE)
|
|
{
|
|
UpdateSpkr();
|
|
ULONG nSamplesUsed;
|
|
|
|
if(g_bFullSpeed)
|
|
nSamplesUsed = Spkr_SubmitWaveBuffer_FullSpeed(g_pSpeakerBuffer, g_nBufferIdx);
|
|
else
|
|
nSamplesUsed = Spkr_SubmitWaveBuffer(g_pSpeakerBuffer, g_nBufferIdx);
|
|
|
|
_ASSERT(nSamplesUsed <= g_nBufferIdx);
|
|
memmove(g_pSpeakerBuffer, &g_pSpeakerBuffer[nSamplesUsed], g_nBufferIdx-nSamplesUsed); // FIXME-TC: _Size * 2
|
|
g_nBufferIdx -= nSamplesUsed;
|
|
}
|
|
}
|
|
|
|
// Called from SoundCore_TimerFunc() for FADE_OUT
|
|
void SpkrUpdate_Timer()
|
|
{
|
|
if (soundtype == SOUND_WAVE)
|
|
{
|
|
UpdateSpkr();
|
|
ULONG nSamplesUsed;
|
|
|
|
nSamplesUsed = Spkr_SubmitWaveBuffer_FullSpeed(g_pSpeakerBuffer, g_nBufferIdx);
|
|
|
|
_ASSERT(nSamplesUsed <= g_nBufferIdx);
|
|
memmove(g_pSpeakerBuffer, &g_pSpeakerBuffer[nSamplesUsed], g_nBufferIdx-nSamplesUsed); // FIXME-TC: _Size * 2 (GH#213?)
|
|
g_nBufferIdx -= nSamplesUsed;
|
|
}
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
static DWORD dwByteOffset = (DWORD)-1;
|
|
static int nNumSamplesError = 0;
|
|
static int nDbgSpkrCnt = 0;
|
|
|
|
// FullSpeed g_nAppMode, 2 cases:
|
|
// i) Short burst of full-speed, so PlayCursor doesn't complete sound from previous fixed-speed session.
|
|
// ii) Long burst of full-speed, so PlayCursor completes sound from previous fixed-speed session.
|
|
|
|
// Try to:
|
|
// 1) Output remaining samples from SpeakerBuffer (from previous fixed-speed session)
|
|
// 2) Output pad samples to keep the VoiceBuffer topped-up
|
|
|
|
// If nNumSamples>0 then these are from previous fixed-speed session.
|
|
// - Output these before outputting zero-pad samples.
|
|
|
|
static ULONG Spkr_SubmitWaveBuffer_FullSpeed(short* pSpeakerBuffer, ULONG nNumSamples)
|
|
{
|
|
//char szDbg[200];
|
|
nDbgSpkrCnt++;
|
|
|
|
if(!SpeakerVoice.bActive)
|
|
return nNumSamples;
|
|
|
|
// pSpeakerBuffer can't be NULL, as reset clears g_bFullSpeed, so 1st SpkrUpdate() never calls here
|
|
_ASSERT(pSpeakerBuffer != NULL);
|
|
|
|
//
|
|
|
|
DWORD dwDSLockedBufferSize0, dwDSLockedBufferSize1;
|
|
SHORT *pDSLockedBuffer0, *pDSLockedBuffer1;
|
|
//bool bBufferError = false;
|
|
|
|
DWORD dwCurrentPlayCursor, dwCurrentWriteCursor;
|
|
HRESULT hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
|
|
if(FAILED(hr))
|
|
return nNumSamples;
|
|
|
|
if(dwByteOffset == (DWORD)-1)
|
|
{
|
|
// First time in this func (probably after re-init (Spkr_SubmitWaveBuffer()))
|
|
|
|
dwByteOffset = dwCurrentPlayCursor + (g_dwDSSpkrBufferSize/8)*3; // Ideal: 0.375 is between 0.25 & 0.50 full
|
|
dwByteOffset %= g_dwDSSpkrBufferSize;
|
|
//sprintf(szDbg, "[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X [REINIT]\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples); OutputDebugString(szDbg);
|
|
}
|
|
else
|
|
{
|
|
// Check that our offset isn't between Play & Write positions
|
|
|
|
if(dwCurrentWriteCursor > dwCurrentPlayCursor)
|
|
{
|
|
// |-----PxxxxxW-----|
|
|
if((dwByteOffset > dwCurrentPlayCursor) && (dwByteOffset < dwCurrentWriteCursor))
|
|
{
|
|
//sprintf(szDbg, "[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X xxx\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples); OutputDebugString(szDbg);
|
|
dwByteOffset = dwCurrentWriteCursor;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// |xxW----------Pxxx|
|
|
if((dwByteOffset > dwCurrentPlayCursor) || (dwByteOffset < dwCurrentWriteCursor))
|
|
{
|
|
//sprintf(szDbg, "[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X XXX\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples); OutputDebugString(szDbg);
|
|
dwByteOffset = dwCurrentWriteCursor;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Calc bytes remaining to be played
|
|
int nBytesRemaining = dwByteOffset - dwCurrentPlayCursor;
|
|
if(nBytesRemaining < 0)
|
|
nBytesRemaining += g_dwDSSpkrBufferSize;
|
|
if((nBytesRemaining == 0) && (dwCurrentPlayCursor != dwCurrentWriteCursor))
|
|
nBytesRemaining = g_dwDSSpkrBufferSize; // Case when complete buffer is to be played
|
|
|
|
//
|
|
|
|
UINT nNumPadSamples = 0;
|
|
|
|
if(nBytesRemaining < g_dwDSSpkrBufferSize / 4)
|
|
{
|
|
// < 1/4 of play-buffer remaining (need *more* data)
|
|
nNumPadSamples = ((g_dwDSSpkrBufferSize / 4) - nBytesRemaining) / sizeof(short);
|
|
|
|
if(nNumPadSamples > nNumSamples)
|
|
nNumPadSamples -= nNumSamples;
|
|
else
|
|
nNumPadSamples = 0;
|
|
|
|
// NB. If nNumPadSamples>0 then all nNumSamples are to be used
|
|
}
|
|
|
|
//
|
|
|
|
UINT nBytesFree = g_dwDSSpkrBufferSize - nBytesRemaining; // Calc free buffer space
|
|
ULONG nNumSamplesToUse = nNumSamples + nNumPadSamples;
|
|
|
|
if(nNumSamplesToUse * sizeof(short) > nBytesFree)
|
|
nNumSamplesToUse = nBytesFree / sizeof(short);
|
|
|
|
//
|
|
|
|
if(nNumSamplesToUse >= 128) // Limit the buffer unlock/locking to a minimum
|
|
{
|
|
if(!DSGetLock(SpeakerVoice.lpDSBvoice,
|
|
dwByteOffset, (DWORD)nNumSamplesToUse*sizeof(short),
|
|
&pDSLockedBuffer0, &dwDSLockedBufferSize0,
|
|
&pDSLockedBuffer1, &dwDSLockedBufferSize1))
|
|
return nNumSamples;
|
|
|
|
//
|
|
|
|
DWORD dwBufferSize0 = 0;
|
|
DWORD dwBufferSize1 = 0;
|
|
|
|
if(nNumSamples)
|
|
{
|
|
//sprintf(szDbg, "[Submit_FS] C=%08X, PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X ***\n", nDbgSpkrCnt, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples); OutputDebugString(szDbg);
|
|
|
|
if(nNumSamples*sizeof(short) <= dwDSLockedBufferSize0)
|
|
{
|
|
dwBufferSize0 = nNumSamples*sizeof(short);
|
|
dwBufferSize1 = 0;
|
|
}
|
|
else
|
|
{
|
|
dwBufferSize0 = dwDSLockedBufferSize0;
|
|
dwBufferSize1 = nNumSamples*sizeof(short) - dwDSLockedBufferSize0;
|
|
|
|
if(dwBufferSize1 > dwDSLockedBufferSize1)
|
|
dwBufferSize1 = dwDSLockedBufferSize1;
|
|
}
|
|
|
|
memcpy(pDSLockedBuffer0, &pSpeakerBuffer[0], dwBufferSize0);
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer0, dwBufferSize0/sizeof(short));
|
|
#endif
|
|
nNumSamples = dwBufferSize0/sizeof(short);
|
|
|
|
if(pDSLockedBuffer1 && dwBufferSize1)
|
|
{
|
|
memcpy(pDSLockedBuffer1, &pSpeakerBuffer[dwDSLockedBufferSize0/sizeof(short)], dwBufferSize1);
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer1, dwBufferSize1/sizeof(short));
|
|
#endif
|
|
nNumSamples += dwBufferSize1/sizeof(short);
|
|
}
|
|
}
|
|
|
|
if(nNumPadSamples)
|
|
{
|
|
//sprintf(szDbg, "[Submit_FS] C=%08X, PC=%08X, WC=%08X, Diff=%08X, Off=%08X, PS=%08X, Data=%04X\n", nDbgSpkrCnt, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumPadSamples, g_nSpeakerData); OutputDebugString(szDbg);
|
|
|
|
dwBufferSize0 = dwDSLockedBufferSize0 - dwBufferSize0;
|
|
dwBufferSize1 = dwDSLockedBufferSize1 - dwBufferSize1;
|
|
|
|
if(dwBufferSize0)
|
|
{
|
|
wmemset((wchar_t*)pDSLockedBuffer0, (wchar_t)DCFilter(g_nSpeakerData), dwBufferSize0/sizeof(wchar_t));
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer0, dwBufferSize0/sizeof(short));
|
|
#endif
|
|
}
|
|
|
|
if(pDSLockedBuffer1)
|
|
{
|
|
wmemset((wchar_t*)pDSLockedBuffer1, (wchar_t)DCFilter(g_nSpeakerData), dwBufferSize1/sizeof(wchar_t));
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer1, dwBufferSize1/sizeof(short));
|
|
#endif
|
|
}
|
|
}
|
|
|
|
// Commit sound buffer
|
|
hr = SpeakerVoice.lpDSBvoice->Unlock((void*)pDSLockedBuffer0, dwDSLockedBufferSize0,
|
|
(void*)pDSLockedBuffer1, dwDSLockedBufferSize1);
|
|
if(FAILED(hr))
|
|
return nNumSamples;
|
|
|
|
dwByteOffset = (dwByteOffset + (DWORD)nNumSamplesToUse*sizeof(short)*g_nSPKR_NumChannels) % g_dwDSSpkrBufferSize;
|
|
}
|
|
|
|
return nNumSamples;
|
|
}
|
|
|
|
//-----------------------------------------------------------------------------
|
|
|
|
static ULONG Spkr_SubmitWaveBuffer(short* pSpeakerBuffer, ULONG nNumSamples)
|
|
{
|
|
//char szDbg[200];
|
|
nDbgSpkrCnt++;
|
|
|
|
if(!SpeakerVoice.bActive)
|
|
return nNumSamples;
|
|
|
|
if(pSpeakerBuffer == NULL)
|
|
{
|
|
// Re-init from SpkrReset()
|
|
dwByteOffset = (DWORD)-1;
|
|
nNumSamplesError = 0;
|
|
|
|
// Don't call DSZeroVoiceBuffer() - get noise with "VIA AC'97 Enhanced Audio Controller"
|
|
// . I guess SpeakerVoice.Stop() isn't really working and the new zero buffer causes noise corruption when submitted.
|
|
DSZeroVoiceWritableBuffer(&SpeakerVoice, "Spkr", g_dwDSSpkrBufferSize);
|
|
|
|
return 0;
|
|
}
|
|
|
|
//
|
|
|
|
DWORD dwDSLockedBufferSize0, dwDSLockedBufferSize1;
|
|
SHORT *pDSLockedBuffer0, *pDSLockedBuffer1;
|
|
bool bBufferError = false;
|
|
|
|
DWORD dwCurrentPlayCursor, dwCurrentWriteCursor;
|
|
HRESULT hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
|
|
if(FAILED(hr))
|
|
return nNumSamples;
|
|
|
|
if(dwByteOffset == (DWORD)-1)
|
|
{
|
|
// First time in this func (probably after re-init (above))
|
|
|
|
dwByteOffset = dwCurrentPlayCursor + (g_dwDSSpkrBufferSize/8)*3; // Ideal: 0.375 is between 0.25 & 0.50 full
|
|
dwByteOffset %= g_dwDSSpkrBufferSize;
|
|
//sprintf(szDbg, "[Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X [REINIT]\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset); OutputDebugString(szDbg);
|
|
}
|
|
else
|
|
{
|
|
// Check that our offset isn't between Play & Write positions
|
|
|
|
if(dwCurrentWriteCursor > dwCurrentPlayCursor)
|
|
{
|
|
// |-----PxxxxxW-----|
|
|
if((dwByteOffset > dwCurrentPlayCursor) && (dwByteOffset < dwCurrentWriteCursor))
|
|
{
|
|
double fTicksSecs = (double)GetTickCount() / 1000.0;
|
|
//sprintf(szDbg, "%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X xxx\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
|
|
//OutputDebugString(szDbg);
|
|
//if (g_fh) fprintf(g_fh, szDbg);
|
|
|
|
dwByteOffset = dwCurrentWriteCursor;
|
|
nNumSamplesError = 0;
|
|
bBufferError = true;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// |xxW----------Pxxx|
|
|
if((dwByteOffset > dwCurrentPlayCursor) || (dwByteOffset < dwCurrentWriteCursor))
|
|
{
|
|
double fTicksSecs = (double)GetTickCount() / 1000.0;
|
|
//sprintf(szDbg, "%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X XXX\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
|
|
//OutputDebugString(szDbg);
|
|
//if (g_fh) fprintf(g_fh, "%s", szDbg);
|
|
|
|
dwByteOffset = dwCurrentWriteCursor;
|
|
nNumSamplesError = 0;
|
|
bBufferError = true;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Calc bytes remaining to be played
|
|
int nBytesRemaining = dwByteOffset - dwCurrentPlayCursor;
|
|
if(nBytesRemaining < 0)
|
|
nBytesRemaining += g_dwDSSpkrBufferSize;
|
|
if((nBytesRemaining == 0) && (dwCurrentPlayCursor != dwCurrentWriteCursor))
|
|
nBytesRemaining = g_dwDSSpkrBufferSize; // Case when complete buffer is to be played
|
|
|
|
// Calc correction factor so that play-buffer doesn't under/overflow
|
|
const int nErrorInc = SoundCore_GetErrorInc();
|
|
if(nBytesRemaining < g_dwDSSpkrBufferSize / 4)
|
|
nNumSamplesError += nErrorInc; // < 1/4 of play-buffer remaining (need *more* data)
|
|
else if(nBytesRemaining > g_dwDSSpkrBufferSize / 2)
|
|
nNumSamplesError -= nErrorInc; // > 1/2 of play-buffer remaining (need *less* data)
|
|
else
|
|
nNumSamplesError = 0; // Acceptable amount of data in buffer
|
|
|
|
const int nErrorMax = SoundCore_GetErrorMax(); // Cap feedback to +/-nMaxError units
|
|
if(nNumSamplesError < -nErrorMax) nNumSamplesError = -nErrorMax;
|
|
if(nNumSamplesError > nErrorMax) nNumSamplesError = nErrorMax;
|
|
g_nCpuCyclesFeedback = (int) ((double)nNumSamplesError * g_fClksPerSpkrSample);
|
|
|
|
//
|
|
|
|
UINT nBytesFree = g_dwDSSpkrBufferSize - nBytesRemaining; // Calc free buffer space
|
|
ULONG nNumSamplesToUse = nNumSamples;
|
|
|
|
if(nNumSamplesToUse * sizeof(short) > nBytesFree)
|
|
nNumSamplesToUse = nBytesFree / sizeof(short);
|
|
|
|
if(bBufferError)
|
|
pSpeakerBuffer = &pSpeakerBuffer[nNumSamples - nNumSamplesToUse];
|
|
|
|
//
|
|
|
|
if(nNumSamplesToUse)
|
|
{
|
|
//sprintf(szDbg, "[Submit] C=%08X, PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X +++\n", nDbgSpkrCnt, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamplesToUse); OutputDebugString(szDbg);
|
|
|
|
if(!DSGetLock(SpeakerVoice.lpDSBvoice,
|
|
dwByteOffset, (DWORD)nNumSamplesToUse*sizeof(short),
|
|
&pDSLockedBuffer0, &dwDSLockedBufferSize0,
|
|
&pDSLockedBuffer1, &dwDSLockedBufferSize1))
|
|
return nNumSamples;
|
|
|
|
memcpy(pDSLockedBuffer0, &pSpeakerBuffer[0], dwDSLockedBufferSize0);
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer0, dwDSLockedBufferSize0/sizeof(short));
|
|
#endif
|
|
|
|
if(pDSLockedBuffer1)
|
|
{
|
|
memcpy(pDSLockedBuffer1, &pSpeakerBuffer[dwDSLockedBufferSize0/sizeof(short)], dwDSLockedBufferSize1);
|
|
#ifdef RIFF_SPKR
|
|
RiffPutSamples(pDSLockedBuffer1, dwDSLockedBufferSize1/sizeof(short));
|
|
#endif
|
|
}
|
|
|
|
// Commit sound buffer
|
|
hr = SpeakerVoice.lpDSBvoice->Unlock((void*)pDSLockedBuffer0, dwDSLockedBufferSize0,
|
|
(void*)pDSLockedBuffer1, dwDSLockedBufferSize1);
|
|
if(FAILED(hr))
|
|
return nNumSamples;
|
|
|
|
dwByteOffset = (dwByteOffset + (DWORD)nNumSamplesToUse*sizeof(short)*g_nSPKR_NumChannels) % g_dwDSSpkrBufferSize;
|
|
}
|
|
|
|
return bBufferError ? nNumSamples : nNumSamplesToUse;
|
|
}
|
|
|
|
//-----------------------------------------------------------------------------
|
|
|
|
void Spkr_Mute()
|
|
{
|
|
if(SpeakerVoice.bActive && !SpeakerVoice.bMute)
|
|
{
|
|
SpeakerVoice.lpDSBvoice->SetVolume(DSBVOLUME_MIN);
|
|
SpeakerVoice.bMute = true;
|
|
}
|
|
}
|
|
|
|
void Spkr_Demute()
|
|
{
|
|
if(SpeakerVoice.bActive && SpeakerVoice.bMute)
|
|
{
|
|
SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
|
|
SpeakerVoice.bMute = false;
|
|
}
|
|
}
|
|
|
|
//-----------------------------------------------------------------------------
|
|
|
|
static bool g_bSpkrRecentlyActive = false;
|
|
|
|
static void Spkr_SetActive(bool bActive)
|
|
{
|
|
if(!SpeakerVoice.bActive)
|
|
return;
|
|
|
|
if(bActive)
|
|
{
|
|
// Called by SpkrToggle() or SpkrReset()
|
|
g_bSpkrRecentlyActive = true;
|
|
SpeakerVoice.bRecentlyActive = true;
|
|
}
|
|
else
|
|
{
|
|
// Called by SpkrUpdate() after 0.2s of speaker inactivity
|
|
g_bSpkrRecentlyActive = false;
|
|
SpeakerVoice.bRecentlyActive = false;
|
|
g_bQuieterSpeaker = 0; // undo any muting (for 8 bit DAC)
|
|
}
|
|
}
|
|
|
|
bool Spkr_IsActive()
|
|
{
|
|
return g_bSpkrRecentlyActive;
|
|
}
|
|
|
|
//-----------------------------------------------------------------------------
|
|
|
|
DWORD SpkrGetVolume()
|
|
{
|
|
return SpeakerVoice.dwUserVolume;
|
|
}
|
|
|
|
void SpkrSetVolume(DWORD dwVolume, DWORD dwVolumeMax)
|
|
{
|
|
SpeakerVoice.dwUserVolume = dwVolume;
|
|
|
|
SpeakerVoice.nVolume = NewVolume(dwVolume, dwVolumeMax);
|
|
|
|
if (SpeakerVoice.bActive && !SpeakerVoice.bMute)
|
|
SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
bool Spkr_DSInit()
|
|
{
|
|
//
|
|
// Create single Apple speaker voice
|
|
//
|
|
|
|
if(!g_bDSAvailable)
|
|
return false;
|
|
|
|
SpeakerVoice.bIsSpeaker = true;
|
|
|
|
HRESULT hr = DSGetSoundBuffer(&SpeakerVoice, DSBCAPS_CTRLVOLUME, g_dwDSSpkrBufferSize, SPKR_SAMPLE_RATE, 1);
|
|
if(FAILED(hr))
|
|
{
|
|
if(g_fh) fprintf(g_fh, "Spkr: DSGetSoundBuffer failed (%08X)\n",hr);
|
|
return false;
|
|
}
|
|
|
|
if(!DSZeroVoiceBuffer(&SpeakerVoice, "Spkr", g_dwDSSpkrBufferSize))
|
|
return false;
|
|
|
|
SpeakerVoice.bActive = true;
|
|
|
|
// Volume might've been setup from value in Registry
|
|
if(!SpeakerVoice.nVolume)
|
|
SpeakerVoice.nVolume = DSBVOLUME_MAX;
|
|
|
|
SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
|
|
|
|
//
|
|
|
|
DWORD dwCurrentPlayCursor, dwCurrentWriteCursor;
|
|
hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
|
|
if(SUCCEEDED(hr) && (dwCurrentPlayCursor == dwCurrentWriteCursor))
|
|
{
|
|
// KLUDGE: For my WinXP PC with "VIA AC'97 Enhanced Audio Controller"
|
|
// . Not required for my Win98SE/WinXP PC with PCI "Soundblaster Live!"
|
|
Sleep(200);
|
|
|
|
hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
|
|
char szDbg[100];
|
|
sprintf(szDbg, "[DSInit] PC=%08X, WC=%08X, Diff=%08X\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor); OutputDebugString(szDbg);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
static void Spkr_DSUninit()
|
|
{
|
|
if(SpeakerVoice.lpDSBvoice && SpeakerVoice.bActive)
|
|
{
|
|
SpeakerVoice.lpDSBvoice->Stop();
|
|
SpeakerVoice.bActive = false;
|
|
}
|
|
|
|
DSReleaseSoundBuffer(&SpeakerVoice);
|
|
}
|
|
|
|
//=============================================================================
|
|
|
|
#define SS_YAML_KEY_LASTCYCLE "Last Cycle"
|
|
|
|
static std::string SpkrGetSnapshotStructName(void)
|
|
{
|
|
static const std::string name("Speaker");
|
|
return name;
|
|
}
|
|
|
|
void SpkrSaveSnapshot(YamlSaveHelper& yamlSaveHelper)
|
|
{
|
|
YamlSaveHelper::Label state(yamlSaveHelper, "%s:\n", SpkrGetSnapshotStructName().c_str());
|
|
yamlSaveHelper.SaveHexUint64(SS_YAML_KEY_LASTCYCLE, g_nSpkrLastCycle);
|
|
}
|
|
|
|
void SpkrLoadSnapshot(YamlLoadHelper& yamlLoadHelper)
|
|
{
|
|
if (!yamlLoadHelper.GetSubMap(SpkrGetSnapshotStructName()))
|
|
return;
|
|
|
|
g_nSpkrLastCycle = yamlLoadHelper.LoadUint64(SS_YAML_KEY_LASTCYCLE);
|
|
|
|
yamlLoadHelper.PopMap();
|
|
}
|