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AppleWin/source/Speaker.cpp

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/*
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 "Speaker.h"
#include "Core.h"
#include "CPU.h"
#include "Interface.h"
#include "Log.h"
#include "Memory.h"
#include "SoundCore.h"
#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.
//
// NB. Setting g_nSPKR_NumChannels=1 still works (ie. mono).
// . Retain it for a while in case there are regressions with the new 2-channel code, then remove it.
static const unsigned short g_nSPKR_NumChannels = 2;
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; // Sample index
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;
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 bool g_bSpkrOutputToRiff = false;
void Spkr_OutputToRiff(void)
{
g_bSpkrOutputToRiff = true;
}
UINT Spkr_GetNumChannels(void)
{
return g_nSPKR_NumChannels;
}
//=============================================================================
static void DisplayBenchmarkResults ()
{
DWORD totaltime = GetTickCount()-extbench;
GetFrame().VideoRedrawScreen();
std::string strText = StrFormat("This benchmark took %u.%02u seconds.",
(unsigned)(totaltime / 1000),
(unsigned)((totaltime / 10) % 100));
GetFrame().FrameMessageBox(strText.c_str(),
"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;
LogFileOutput("SpkrInitialize: g_bDisableDirectSound=1... SpeakerVoice.bMute=true\n");
}
else
{
g_bSpkrAvailable = Spkr_DSInit();
LogFileOutput("Spkr_DSInit(), res=%d\n", g_bSpkrAvailable ? 1 : 0);
if (!g_bSpkrAvailable)
{
GetFrame().FrameMessageBox(
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);
}
}
//
if (soundtype == SOUND_WAVE)
{
InitRemainderBuffer();
g_pSpeakerBuffer = new short [SPKR_SAMPLE_RATE * g_nSPKR_NumChannels]; // 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_Unmute();
}
//=============================================================================
void SpkrSetEmulationType (SoundType_e newtype)
{
SpkrDestroy(); // GH#295: Destroy for all types (even SOUND_NONE)
soundtype = newtype;
if (soundtype != SOUND_NONE)
SpkrInitialize();
}
//=============================================================================
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)
{
if (g_nSPKR_NumChannels == 1)
{
g_pSpeakerBuffer[g_nBufferIdx] = DCFilter((short)nSampleMean);
}
else
{
const short sample = DCFilter((short)nSampleMean);
g_pSpeakerBuffer[g_nBufferIdx * 2 + 0] = sample;
g_pSpeakerBuffer[g_nBufferIdx * 2 + 1] = sample;
}
g_nBufferIdx++;
}
}
}
}
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))
{
if (g_nSPKR_NumChannels == 1)
{
g_pSpeakerBuffer[g_nBufferIdx] = DCFilter(g_nSpeakerData);
}
else
{
const short sample = DCFilter(g_nSpeakerData);
g_pSpeakerBuffer[g_nBufferIdx * 2 + 0] = sample;
g_pSpeakerBuffer[g_nBufferIdx * 2 + 1] = sample;
}
g_nBufferIdx++;
}
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) * sizeof(short) * g_nSPKR_NumChannels);
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) * sizeof(short) * g_nSPKR_NumChannels);
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)
{
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;
//LogOutput("[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X [REINIT]\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
}
else
{
// Check that our offset isn't between Play & Write positions
if(dwCurrentWriteCursor > dwCurrentPlayCursor)
{
// |-----PxxxxxW-----|
if((dwByteOffset > dwCurrentPlayCursor) && (dwByteOffset < dwCurrentWriteCursor))
{
//LogOutput("[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X xxx\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
dwByteOffset = dwCurrentWriteCursor;
}
}
else
{
// |xxW----------Pxxx|
if((dwByteOffset > dwCurrentPlayCursor) || (dwByteOffset < dwCurrentWriteCursor))
{
//LogOutput("[Submit_FS] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X XXX\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
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) * g_nSPKR_NumChannels);
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) * g_nSPKR_NumChannels > nBytesFree)
nNumSamplesToUse = nBytesFree / (sizeof(short) * g_nSPKR_NumChannels);
//
if(nNumSamplesToUse >= 128) // Limit the buffer unlock/locking to a minimum
{
hr = DSGetLock(SpeakerVoice.lpDSBvoice,
dwByteOffset, (DWORD)nNumSamplesToUse * sizeof(short) * g_nSPKR_NumChannels,
&pDSLockedBuffer0, &dwDSLockedBufferSize0,
&pDSLockedBuffer1, &dwDSLockedBufferSize1);
if (FAILED(hr))
return nNumSamples;
//
DWORD dwBufferSize0 = 0;
DWORD dwBufferSize1 = 0;
if(nNumSamples)
{
//LogOutput("[Submit_FS] C=%08X, PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X ***\n", nDbgSpkrCnt, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
if (nNumSamples * sizeof(short) * g_nSPKR_NumChannels <= dwDSLockedBufferSize0)
{
dwBufferSize0 = nNumSamples * sizeof(short) * g_nSPKR_NumChannels;
dwBufferSize1 = 0;
}
else
{
dwBufferSize0 = dwDSLockedBufferSize0;
dwBufferSize1 = nNumSamples * sizeof(short) * g_nSPKR_NumChannels - dwDSLockedBufferSize0;
if(dwBufferSize1 > dwDSLockedBufferSize1)
dwBufferSize1 = dwDSLockedBufferSize1;
}
memcpy(pDSLockedBuffer0, &pSpeakerBuffer[0], dwBufferSize0);
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer0, dwBufferSize0 / (sizeof(short) * g_nSPKR_NumChannels));
nNumSamples = dwBufferSize0 / (sizeof(short) * g_nSPKR_NumChannels);
if(pDSLockedBuffer1 && dwBufferSize1)
{
memcpy(pDSLockedBuffer1, &pSpeakerBuffer[dwDSLockedBufferSize0/sizeof(short)], dwBufferSize1);
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer1, dwBufferSize1 / (sizeof(short) * g_nSPKR_NumChannels));
nNumSamples += dwBufferSize1 / (sizeof(short) * g_nSPKR_NumChannels);
}
}
if(nNumPadSamples)
{
//LogOutput("[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);
dwBufferSize0 = dwDSLockedBufferSize0 - dwBufferSize0;
dwBufferSize1 = dwDSLockedBufferSize1 - dwBufferSize1;
if(dwBufferSize0)
{
const UINT numSamples = dwBufferSize0 / (sizeof(short) * g_nSPKR_NumChannels);
if (g_nSPKR_NumChannels == 1)
{
std::fill_n(pDSLockedBuffer0, numSamples, DCFilter(g_nSpeakerData));
}
else
{
for (UINT i = 0; i < numSamples; i++)
{
const short sample = DCFilter(g_nSpeakerData);
pDSLockedBuffer0[i * 2 + 0] = sample;
pDSLockedBuffer0[i * 2 + 1] = sample;
}
}
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer0, numSamples);
}
if(pDSLockedBuffer1)
{
const UINT numSamples = dwBufferSize0 / (sizeof(short) * g_nSPKR_NumChannels);
if (g_nSPKR_NumChannels == 1)
{
std::fill_n(pDSLockedBuffer1, numSamples, DCFilter(g_nSpeakerData));
}
else
{
for (UINT i = 0; i < numSamples; i++)
{
const short sample = DCFilter(g_nSpeakerData);
pDSLockedBuffer1[i * 2 + 0] = sample;
pDSLockedBuffer1[i * 2 + 1] = sample;
}
}
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer1, numSamples);
}
}
// 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)
{
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.
bool res = DSZeroVoiceWritableBuffer(&SpeakerVoice, g_dwDSSpkrBufferSize);
LogFileOutput("Spkr_SubmitWaveBuffer: DSZeroVoiceWritableBuffer, res=%d\n", res ? 1 : 0);
return 0;
}
//
DWORD dwDSLockedBufferSize0, dwDSLockedBufferSize1;
SHORT *pDSLockedBuffer0, *pDSLockedBuffer1;
bool bBufferError = false;
DWORD dwCurrentPlayCursor, dwCurrentWriteCursor;
HRESULT hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
if (FAILED(hr))
{
LogFileOutput("Spkr_SubmitWaveBuffer: GetCurrentPosition failed (%08X)\n", 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;
//LogOutput("[Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X [REINIT]\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset);
}
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;
//LogOutput("%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X xxx\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
//LogFileOutput("%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X xxx\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
dwByteOffset = dwCurrentWriteCursor;
nNumSamplesError = 0;
bBufferError = true;
}
}
else
{
// |xxW----------Pxxx|
if((dwByteOffset > dwCurrentPlayCursor) || (dwByteOffset < dwCurrentWriteCursor))
{
double fTicksSecs = (double)GetTickCount() / 1000.0;
//LogOutput("%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X XXX\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
//LogFileOutput("%010.3f: [Submit] PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X XXX\n", fTicksSecs, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamples);
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)
{
//LogOutput("[Submit] C=%08X, PC=%08X, WC=%08X, Diff=%08X, Off=%08X, NS=%08X +++\n", nDbgSpkrCnt, dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor, dwByteOffset, nNumSamplesToUse);
hr = DSGetLock(SpeakerVoice.lpDSBvoice,
dwByteOffset, (DWORD)nNumSamplesToUse * sizeof(short) * g_nSPKR_NumChannels,
&pDSLockedBuffer0, &dwDSLockedBufferSize0,
&pDSLockedBuffer1, &dwDSLockedBufferSize1);
if (FAILED(hr))
{
LogFileOutput("Spkr_SubmitWaveBuffer: DSGetLock failed\n");
return nNumSamples;
}
memcpy(pDSLockedBuffer0, &pSpeakerBuffer[0], dwDSLockedBufferSize0);
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer0, dwDSLockedBufferSize0 / (sizeof(short) * g_nSPKR_NumChannels));
if(pDSLockedBuffer1)
{
memcpy(pDSLockedBuffer1, &pSpeakerBuffer[dwDSLockedBufferSize0/sizeof(short)], dwDSLockedBufferSize1);
if (g_bSpkrOutputToRiff)
RiffPutSamples(pDSLockedBuffer1, dwDSLockedBufferSize1 / (sizeof(short) * g_nSPKR_NumChannels));
}
// Commit sound buffer
hr = SpeakerVoice.lpDSBvoice->Unlock((void*)pDSLockedBuffer0, dwDSLockedBufferSize0,
(void*)pDSLockedBuffer1, dwDSLockedBufferSize1);
if (FAILED(hr))
{
LogFileOutput("Spkr_SubmitWaveBuffer: Unlock failed (%08X)\n", hr);
return nNumSamples;
}
dwByteOffset = (dwByteOffset + (DWORD)nNumSamplesToUse*sizeof(short)*g_nSPKR_NumChannels) % g_dwDSSpkrBufferSize;
}
return bBufferError ? nNumSamples : nNumSamplesToUse;
}
//-----------------------------------------------------------------------------
// NB. Not currently used
void Spkr_Mute()
{
if(SpeakerVoice.bActive && !SpeakerVoice.bMute)
{
HRESULT hr = SpeakerVoice.lpDSBvoice->SetVolume(DSBVOLUME_MIN);
LogFileOutput("Spkr_Mute: SetVolume(%d) res = %08X\n", DSBVOLUME_MIN, hr);
SpeakerVoice.bMute = true;
}
}
// NB. Only called by SpkrReset()
void Spkr_Unmute()
{
if(SpeakerVoice.bActive && SpeakerVoice.bMute)
{
HRESULT hr = SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
LogFileOutput("Spkr_Unmute: SetVolume(%d) res = %08X\n", SpeakerVoice.nVolume, hr);
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)
{
HRESULT hr = SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
LogFileOutput("SpkrSetVolume: SetVolume(%d) res = %08X\n", SpeakerVoice.nVolume, hr);
}
}
//=============================================================================
bool Spkr_DSInit()
{
//
// Create single Apple speaker voice
//
if (!g_bDSAvailable)
{
LogFileOutput("Spkr_DSInit: g_bDSAvailable=0\n");
return false;
}
SpeakerVoice.bIsSpeaker = true;
HRESULT hr = DSGetSoundBuffer(&SpeakerVoice, DSBCAPS_CTRLVOLUME, g_dwDSSpkrBufferSize, SPKR_SAMPLE_RATE, g_nSPKR_NumChannels, "Spkr");
if (FAILED(hr))
{
LogFileOutput("Spkr_DSInit: DSGetSoundBuffer failed (%08X)\n", hr);
return false;
}
if (!DSZeroVoiceBuffer(&SpeakerVoice, g_dwDSSpkrBufferSize))
{
LogFileOutput("Spkr_DSInit: DSZeroVoiceBuffer failed\n");
return false;
}
SpeakerVoice.bActive = true;
// Volume might've been setup from value in Registry
if(!SpeakerVoice.nVolume)
SpeakerVoice.nVolume = DSBVOLUME_MAX;
hr = SpeakerVoice.lpDSBvoice->SetVolume(SpeakerVoice.nVolume);
LogFileOutput("Spkr_DSInit: SetVolume(%d) res = %08X\n", SpeakerVoice.nVolume, hr);
//
DWORD dwCurrentPlayCursor, dwCurrentWriteCursor;
hr = SpeakerVoice.lpDSBvoice->GetCurrentPosition(&dwCurrentPlayCursor, &dwCurrentWriteCursor);
if (FAILED(hr))
LogFileOutput("Spkr_DSInit: GetCurrentPosition failed (%08X)\n", hr);
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);
LogFileOutput("Spkr_DSInit: GetCurrentPosition kludge (%08X)\n", hr);
LogOutput("[DSInit] PC=%08X, WC=%08X, Diff=%08X\n", dwCurrentPlayCursor, dwCurrentWriteCursor, dwCurrentWriteCursor-dwCurrentPlayCursor);
}
return true;
}
static void Spkr_DSUninit()
{
if(SpeakerVoice.lpDSBvoice && SpeakerVoice.bActive)
DSVoiceStop(&SpeakerVoice);
DSReleaseSoundBuffer(&SpeakerVoice);
}
//=============================================================================
#define SS_YAML_KEY_LASTCYCLE "Last Cycle"
static const 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();
}
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