mirror of
https://github.com/AppleWin/AppleWin.git
synced 2024-12-22 09:30:15 +00:00
1038 lines
32 KiB
C++
1038 lines
32 KiB
C++
/*
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AppleWin : An Apple //e emulator for Windows
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Copyright (C) 1994-1996, Michael O'Brien
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Copyright (C) 1999-2001, Oliver Schmidt
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Copyright (C) 2002-2005, Tom Charlesworth
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Copyright (C) 2006-2007, Tom Charlesworth, Michael Pohoreski
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AppleWin is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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AppleWin is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with AppleWin; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/* Description: Speaker emulation
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*
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* Author: Various
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*/
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#include "StdAfx.h"
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#include "Speaker.h"
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#include "Core.h"
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#include "CPU.h"
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#include "Interface.h"
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#include "Log.h"
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#include "Memory.h"
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#include "SoundCore.h"
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#include "YamlHelper.h"
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#include "Riff.h"
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#include "Debugger/Debug.h" // For DWORD extbench
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// Notes:
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//
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// [OLD: 23.191 Apple CLKs == 44100Hz (CLK_6502/44100)]
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// 23 Apple CLKS per PC sample (played back at 44.1KHz)
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//
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//
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// The speaker's wave output drives how much 6502 emulation is done in real-time, eg:
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// If the speaker's wave buffer is running out of sample-data, then more 6502 cycles
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// need to be executed to top-up the wave buffer.
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// This is in contrast to the AY8910 voices, which can simply generate more data if
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// their buffers are running low.
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//
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// NB. Setting g_nSPKR_NumChannels=1 still works (ie. mono).
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// . Retain it for a while in case there are regressions with the new 2-channel code, then remove it.
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static const unsigned short g_nSPKR_NumChannels = 2;
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static const DWORD g_dwDSSpkrBufferSize = MAX_SAMPLES * sizeof(short) * g_nSPKR_NumChannels;
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//-------------------------------------
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static short* g_pSpeakerBuffer = NULL;
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// Globals (SOUND_WAVE)
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const short SPKR_DATA_INIT = (short)0x8000;
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short g_nSpeakerData = SPKR_DATA_INIT;
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static UINT g_nBufferIdx = 0; // Sample index
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static short* g_pRemainderBuffer = NULL;
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static UINT g_nRemainderBufferSize; // Setup in SpkrInitialize()
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static UINT g_nRemainderBufferIdx; // Setup in SpkrInitialize()
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// Application-wide globals:
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SoundType_e soundtype = SOUND_WAVE;
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double g_fClksPerSpkrSample; // Setup in SetClksPerSpkrSample()
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// Allow temporary quietening of speaker (8 bit DAC)
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bool g_bQuieterSpeaker = false;
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// Globals
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static unsigned __int64 g_nSpkrQuietCycleCount = 0;
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static unsigned __int64 g_nSpkrLastCycle = 0;
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static bool g_bSpkrToggleFlag = false;
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static VOICE SpeakerVoice;
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static bool g_bSpkrAvailable = false;
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//-----------------------------------------------------------------------------
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// Forward refs:
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static ULONG Spkr_SubmitWaveBuffer_FullSpeed(short* pSpeakerBuffer, ULONG nNumSamples);
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static ULONG Spkr_SubmitWaveBuffer(short* pSpeakerBuffer, ULONG nNumSamples);
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static void Spkr_SetActive(bool bActive);
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static void Spkr_DSUninit();
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//-----------------------------------------------------------------------------
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static bool g_bSpkrOutputToRiff = false;
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void Spkr_OutputToRiff(void)
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{
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g_bSpkrOutputToRiff = true;
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}
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UINT Spkr_GetNumChannels(void)
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{
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return g_nSPKR_NumChannels;
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}
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//=============================================================================
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static void DisplayBenchmarkResults ()
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{
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DWORD totaltime = GetTickCount()-extbench;
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GetFrame().VideoRedrawScreen();
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std::string strText = StrFormat("This benchmark took %u.%02u seconds.",
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(unsigned)(totaltime / 1000),
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(unsigned)((totaltime / 10) % 100));
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GetFrame().FrameMessageBox(strText.c_str(),
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"Benchmark Results",
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MB_ICONINFORMATION | MB_SETFOREGROUND);
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}
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//=============================================================================
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//
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// DC filtering V2 (Riccardo Macri May 2015) (GH#275)
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//
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// To prevent loud clicks on Window's sound buffer underruns and constant DC
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// being sent out to amplifiers (some soundcards are DC coupled) which is
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// not good for them, an attenuator slowly drops the speaker output
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// to 0 after the speaker (or 8 bit DAC) has been idle for a couple hundred
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// milliseconds.
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//
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// The approach works as follows:
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// - SpkrToggle() is called when the speaker state is flipped by accessing $C030
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// - This brings audio up to date then calls ResetDCFilter()
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// - ResetDCFilter() sets a counter to a high value
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// - every audio sample is processed by DCFilter() as follows:
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// - if the counter is >= 32768, the speaker has been recently toggled
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// and the samples are unaffected
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// - if the counter is < 32768 but > 0, it is used to scale the
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// sample to reduce +ve or -ve speaker states towards zero
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// - In the two cases above, the counter is decremented
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// - if the counter is zero, the speaker has been silent for a while
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// and the output is 0 regardless of the speaker state.
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//
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// - the initial "high value" is chosen so 10000/44100 = about a
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// quarter of a second of speaker inactivity is needed before attenuation
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// begins.
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//
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// NOTE: The attenuation is not ever reducing the level of audio, just
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// the DC offset at which the speaker has been left.
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//
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// This approach has zero impact on any speaker tones including PWM
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// due to the samples being unchanged for at least 0.25 seconds after
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// any speaker activity.
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//
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static UINT g_uDCFilterState = 0;
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inline void ResetDCFilter(void)
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{
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// reset the attenuator with an additional 250ms of full gain
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// (10000 samples) before it starts attenuating
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g_uDCFilterState = 32768 + 10000;
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}
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inline short DCFilter(short sample_in)
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{
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if (g_uDCFilterState == 0) // no sound for a while, stay 0
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return 0;
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if (g_uDCFilterState >= 32768) // full gain after recent sound
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{
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g_uDCFilterState--;
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return sample_in;
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}
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return (((int)sample_in) * (g_uDCFilterState--)) / 32768; // scale & divide by 32768 (NB. Don't ">>15" as undefined behaviour)
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}
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//=============================================================================
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static void SetClksPerSpkrSample()
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{
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// // 23.191 clks for 44.1Khz (when 6502 CLK=1.0Mhz)
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// g_fClksPerSpkrSample = g_fCurrentCLK6502 / (double)SPKR_SAMPLE_RATE;
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// Use integer value: Better for MJ Mahon's RT.SYNTH.DSK (integer multiples of 1.023MHz Clk)
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// . 23 clks @ 1.023MHz
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g_fClksPerSpkrSample = (double) (UINT) (g_fCurrentCLK6502 / (double)SPKR_SAMPLE_RATE);
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}
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//=============================================================================
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static void InitRemainderBuffer()
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{
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delete [] g_pRemainderBuffer;
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SetClksPerSpkrSample();
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g_nRemainderBufferSize = (UINT) g_fClksPerSpkrSample;
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if ((double)g_nRemainderBufferSize < g_fClksPerSpkrSample)
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g_nRemainderBufferSize++;
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g_pRemainderBuffer = new short [g_nRemainderBufferSize];
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memset(g_pRemainderBuffer, 0, g_nRemainderBufferSize);
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g_nRemainderBufferIdx = 0;
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}
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//
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// ----- ALL GLOBALLY ACCESSIBLE FUNCTIONS ARE BELOW THIS LINE -----
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//
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//=============================================================================
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void SpkrDestroy ()
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{
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Spkr_DSUninit();
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//
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if(soundtype == SOUND_WAVE)
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{
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delete [] g_pSpeakerBuffer;
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delete [] g_pRemainderBuffer;
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g_pSpeakerBuffer = NULL;
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g_pRemainderBuffer = NULL;
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}
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}
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//=============================================================================
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void SpkrInitialize ()
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{
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if(g_fh)
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{
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fprintf(g_fh, "Spkr Config: soundtype = %d ", (int) soundtype);
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switch(soundtype)
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{
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case SOUND_NONE: fprintf(g_fh, "(NONE)\n"); break;
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case SOUND_WAVE: fprintf(g_fh, "(WAVE)\n"); break;
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default: fprintf(g_fh, "(UNDEFINED!)\n"); break;
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}
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}
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if(g_bDisableDirectSound)
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{
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SpeakerVoice.bMute = true;
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LogFileOutput("SpkrInitialize: g_bDisableDirectSound=1... SpeakerVoice.bMute=true\n");
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}
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else
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{
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g_bSpkrAvailable = Spkr_DSInit();
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LogFileOutput("Spkr_DSInit(), res=%d\n", g_bSpkrAvailable ? 1 : 0);
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if (!g_bSpkrAvailable)
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{
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GetFrame().FrameMessageBox(
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TEXT("The emulator is unable to initialize a waveform ")
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TEXT("output device. Make sure you have a sound card ")
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TEXT("and a driver installed and that Windows is ")
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TEXT("correctly configured to use the driver. Also ")
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TEXT("ensure that no other program is currently using ")
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TEXT("the device."),
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TEXT("Configuration"),
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MB_ICONEXCLAMATION | MB_SETFOREGROUND);
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}
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}
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//
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if (soundtype == SOUND_WAVE)
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{
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InitRemainderBuffer();
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g_pSpeakerBuffer = new short [SPKR_SAMPLE_RATE * g_nSPKR_NumChannels]; // Buffer can hold a max of 1 seconds worth of samples
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}
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}
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//=============================================================================
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// NB. Called when /g_fCurrentCLK6502/ changes
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void SpkrReinitialize ()
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{
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if (soundtype == SOUND_WAVE)
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{
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InitRemainderBuffer();
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}
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}
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//=============================================================================
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void SpkrReset()
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{
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g_nBufferIdx = 0;
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g_nSpkrQuietCycleCount = 0;
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g_bSpkrToggleFlag = false;
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InitRemainderBuffer();
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Spkr_SubmitWaveBuffer(NULL, 0);
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Spkr_SetActive(false);
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Spkr_Unmute();
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}
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//=============================================================================
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void SpkrSetEmulationType (SoundType_e newtype)
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{
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SpkrDestroy(); // GH#295: Destroy for all types (even SOUND_NONE)
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soundtype = newtype;
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if (soundtype != SOUND_NONE)
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SpkrInitialize();
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}
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//=============================================================================
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static void ReinitRemainderBuffer(UINT nCyclesRemaining)
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{
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if(nCyclesRemaining == 0)
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return;
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for(g_nRemainderBufferIdx=0; g_nRemainderBufferIdx<nCyclesRemaining; g_nRemainderBufferIdx++)
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g_pRemainderBuffer[g_nRemainderBufferIdx] = g_nSpeakerData;
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_ASSERT(g_nRemainderBufferIdx < g_nRemainderBufferSize);
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}
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static void UpdateRemainderBuffer(ULONG* pnCycleDiff)
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{
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if(g_nRemainderBufferIdx)
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{
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while((g_nRemainderBufferIdx < g_nRemainderBufferSize) && *pnCycleDiff)
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{
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g_pRemainderBuffer[g_nRemainderBufferIdx] = g_nSpeakerData;
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g_nRemainderBufferIdx++;
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(*pnCycleDiff)--;
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}
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if(g_nRemainderBufferIdx == g_nRemainderBufferSize)
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{
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g_nRemainderBufferIdx = 0;
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signed long nSampleMean = 0;
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for(UINT i=0; i<g_nRemainderBufferSize; i++)
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nSampleMean += (signed long) g_pRemainderBuffer[i];
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nSampleMean /= (signed long) g_nRemainderBufferSize;
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if (g_nBufferIdx < SPKR_SAMPLE_RATE - 1)
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{
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if (g_nSPKR_NumChannels == 1)
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{
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g_pSpeakerBuffer[g_nBufferIdx] = DCFilter((short)nSampleMean);
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}
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else
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{
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const short sample = DCFilter((short)nSampleMean);
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g_pSpeakerBuffer[g_nBufferIdx * 2 + 0] = sample;
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g_pSpeakerBuffer[g_nBufferIdx * 2 + 1] = sample;
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}
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g_nBufferIdx++;
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}
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}
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}
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}
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static void UpdateSpkr()
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{
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if(!g_bFullSpeed || SoundCore_GetTimerState())
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{
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ULONG nCycleDiff = (ULONG) (g_nCumulativeCycles - g_nSpkrLastCycle);
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UpdateRemainderBuffer(&nCycleDiff);
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ULONG nNumSamples = (ULONG) ((double)nCycleDiff / g_fClksPerSpkrSample);
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ULONG nCyclesRemaining = (ULONG) ((double)nCycleDiff - (double)nNumSamples * g_fClksPerSpkrSample);
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while ((nNumSamples--) && (g_nBufferIdx < SPKR_SAMPLE_RATE - 1))
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{
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if (g_nSPKR_NumChannels == 1)
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{
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g_pSpeakerBuffer[g_nBufferIdx] = DCFilter(g_nSpeakerData);
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}
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else
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{
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const short sample = DCFilter(g_nSpeakerData);
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g_pSpeakerBuffer[g_nBufferIdx * 2 + 0] = sample;
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g_pSpeakerBuffer[g_nBufferIdx * 2 + 1] = sample;
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}
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g_nBufferIdx++;
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}
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ReinitRemainderBuffer(nCyclesRemaining); // Partially fill 1Mhz sample buffer
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}
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g_nSpkrLastCycle = g_nCumulativeCycles;
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}
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//=============================================================================
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// Called by emulation code when Speaker I/O reg is accessed
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//
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BYTE __stdcall SpkrToggle (WORD, WORD, BYTE, BYTE, ULONG nExecutedCycles)
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{
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g_bSpkrToggleFlag = true;
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if(!g_bFullSpeed)
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Spkr_SetActive(true);
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//
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if (extbench)
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{
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DisplayBenchmarkResults();
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extbench = 0;
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}
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if (soundtype == SOUND_WAVE)
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{
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CpuCalcCycles(nExecutedCycles);
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UpdateSpkr();
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short speakerDriveLevel = SPKR_DATA_INIT;
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if (g_bQuieterSpeaker) // quieten the speaker if 8 bit DAC in use
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speakerDriveLevel /= 4; // NB. Don't shift -ve number right: undefined behaviour (MSDN says: implementation-dependent)
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// When full-speed: Don't ResetDCFilter(), otherwise get occasional clicks when speaker toggled
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if (!g_bFullSpeed)
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ResetDCFilter();
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if (g_nSpeakerData == speakerDriveLevel)
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g_nSpeakerData = ~speakerDriveLevel;
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else
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g_nSpeakerData = speakerDriveLevel;
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}
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return MemReadFloatingBus(nExecutedCycles);
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}
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//=============================================================================
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// Called by ContinueExecution()
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void SpkrUpdate (DWORD totalcycles)
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{
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#ifdef LOG_PERF_TIMINGS
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extern UINT64 g_timeSpeaker;
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PerfMarker perfMarker(g_timeSpeaker);
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#endif
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if(!g_bSpkrToggleFlag)
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{
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if(!g_nSpkrQuietCycleCount)
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{
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g_nSpkrQuietCycleCount = g_nCumulativeCycles;
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}
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else if(g_nCumulativeCycles - g_nSpkrQuietCycleCount > (unsigned __int64)g_fCurrentCLK6502/5)
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{
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// After 0.2 sec of Apple time, deactivate spkr voice
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// . This allows emulator to auto-switch to full-speed g_nAppMode for fast disk access
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Spkr_SetActive(false);
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}
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}
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else
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{
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g_nSpkrQuietCycleCount = 0;
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g_bSpkrToggleFlag = false;
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}
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//
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if (soundtype == SOUND_WAVE)
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{
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UpdateSpkr();
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ULONG nSamplesUsed;
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if(g_bFullSpeed)
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nSamplesUsed = Spkr_SubmitWaveBuffer_FullSpeed(g_pSpeakerBuffer, g_nBufferIdx);
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else
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nSamplesUsed = Spkr_SubmitWaveBuffer(g_pSpeakerBuffer, g_nBufferIdx);
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_ASSERT(nSamplesUsed <= g_nBufferIdx);
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memmove(g_pSpeakerBuffer, &g_pSpeakerBuffer[nSamplesUsed], (g_nBufferIdx - nSamplesUsed) * sizeof(short) * g_nSPKR_NumChannels);
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g_nBufferIdx -= nSamplesUsed;
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}
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}
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// Called from SoundCore_TimerFunc() for FADE_OUT
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void SpkrUpdate_Timer()
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{
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if (soundtype == SOUND_WAVE)
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{
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UpdateSpkr();
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ULONG nSamplesUsed;
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nSamplesUsed = Spkr_SubmitWaveBuffer_FullSpeed(g_pSpeakerBuffer, g_nBufferIdx);
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_ASSERT(nSamplesUsed <= g_nBufferIdx);
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memmove(g_pSpeakerBuffer, &g_pSpeakerBuffer[nSamplesUsed], (g_nBufferIdx - nSamplesUsed) * sizeof(short) * g_nSPKR_NumChannels);
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g_nBufferIdx -= nSamplesUsed;
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}
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}
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//=============================================================================
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static DWORD dwByteOffset = (DWORD)-1;
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static int nNumSamplesError = 0;
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static int nDbgSpkrCnt = 0;
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// FullSpeed g_nAppMode, 2 cases:
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// i) Short burst of full-speed, so PlayCursor doesn't complete sound from previous fixed-speed session.
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// ii) Long burst of full-speed, so PlayCursor completes sound from previous fixed-speed session.
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// Try to:
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// 1) Output remaining samples from SpeakerBuffer (from previous fixed-speed session)
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// 2) Output pad samples to keep the VoiceBuffer topped-up
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// If nNumSamples>0 then these are from previous fixed-speed session.
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// - 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();
|
|
}
|