mirror of
https://github.com/dolphin-emu/dolphin.git
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Some preparations for Wii AX (much work remains)
git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@1101 8ced0084-cf51-0410-be5f-012b33b47a6e
This commit is contained in:
hrydgard
@ -28,7 +28,6 @@
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#include "UCode_AXStructs.h"
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#include "UCode_AX.h"
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// ---------------------------------------------------------------------------------------
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// Externals
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// -----------
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@ -43,7 +42,6 @@ bool gReset = false; // used externally
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extern CDebugger* m_frame;
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// -----------
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CUCode_AX::CUCode_AX(CMailHandler& _rMailHandler, bool wii)
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: IUCode(_rMailHandler)
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, m_addressPBs(0xFFFFFFFF)
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@ -76,7 +74,7 @@ void CUCode_AX::HandleMail(u32 _uMail)
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}
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}
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s16 CUCode_AX::ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac)
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s16 ADPCM_Step(AXParamBlock& pb, u32& samplePos, u32 newSamplePos, u16 frac)
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{
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PBADPCMInfo &adpcm = pb.adpcm;
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@ -158,302 +156,294 @@ u16 ADPCM_Vol(u16 vol, u16 delta, u16 mixer_control)
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}
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// ==============
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void MixAddVoice(AXParamBlock &pb, int *templbuffer, int *temprbuffer, int _iSize)
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{
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#ifdef _WIN32
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ratioFactor = 32000.0f / (float)DSound::DSound_GetSampleRate();
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#else
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ratioFactor = 32000.0f / 44100.0f;
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#endif
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// get necessary values
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const u32 sampleEnd = (pb.audio_addr.end_addr_hi << 16) | pb.audio_addr.end_addr_lo;
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const u32 loopPos = (pb.audio_addr.loop_addr_hi << 16) | pb.audio_addr.loop_addr_lo;
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const u32 updaddr = (u32)(pb.updates.data_hi << 16) | pb.updates.data_lo;
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const u16 updpar = Memory_Read_U16(updaddr);
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const u16 upddata = Memory_Read_U16(updaddr + 2);
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// =======================================================================================
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/*
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Fix problems introduced with the SSBM fix - Sometimes when a music stream ended sampleEnd
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would become extremely high and the game would play random sound data from ARAM resulting in
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a strange noise. This should take care of that. - Some games (Monkey Ball 1 and Tales of
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Symphonia and other) also had one odd last block with a strange high loopPos and strange
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num_updates values, the loopPos limit turns those off also. - Please report any side effects.
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*/
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// ------------
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if (
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(sampleEnd > 0x10000000 || loopPos > 0x10000000)
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&& gSSBMremedy1
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)
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{
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pb.running = 0;
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// also reset all values if it makes any difference
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pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
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pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
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pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
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pb.src.cur_addr_frac = 0; pb.src.ratio_hi = 0; pb.src.ratio_lo = 0;
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pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
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pb.audio_addr.looping = 0;
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pb.adpcm_loop_info.pred_scale = 0;
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pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
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}
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/*
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// the fact that no settings are reset (except running) after a SSBM type music stream or another
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looping block (for example in Battle Stadium DON) has ended could cause loud garbled sound to be
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played from one or more blocks. Perhaps it was in conjunction with the old sequenced music fix below,
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I'm not sure. This was an attempt to prevent that anyway by resetting all. But I'm not sure if this
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is needed anymore. Please try to play SSBM without it and see if it works anyway.
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*/
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if (
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// detect blocks that have recently been running that we should reset
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pb.running == 0 && pb.audio_addr.looping == 1
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//pb.running == 0 && pb.adpcm_loop_info.pred_scale
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// this prevents us from ruining sequenced music blocks, may not be needed
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/*
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&& !(pb.updates.num_updates[0] || pb.updates.num_updates[1] || pb.updates.num_updates[2]
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|| pb.updates.num_updates[3] || pb.updates.num_updates[4])
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*/
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&& !(updpar || upddata)
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&& pb.mixer_control == 0 // only use this in SSBM
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&& gSSBMremedy2 // let us turn this fix on and off
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)
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{
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// reset the detection values
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pb.audio_addr.looping = 0;
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pb.adpcm_loop_info.pred_scale = 0;
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pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
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//pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
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//pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
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//pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
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//pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
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//pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
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}
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// =============
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// =======================================================================================
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// Reset all values
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// ------------
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if (gReset
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&& (pb.running || pb.audio_addr.looping || pb.adpcm_loop_info.pred_scale)
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)
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{
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pb.running = 0;
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pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
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pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
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pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
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pb.src.cur_addr_frac = 0; pb.src.ratio_hi = 0; pb.src.ratio_lo = 0;
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pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
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pb.audio_addr.looping = 0;
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pb.adpcm_loop_info.pred_scale = 0;
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pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
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}
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// =============
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if (pb.running)
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{
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// =======================================================================================
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// Set initial parameters
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// ------------
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//constants
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const u32 ratio = (u32)(((pb.src.ratio_hi << 16) + pb.src.ratio_lo) * ratioFactor);
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//variables
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u32 samplePos = (pb.audio_addr.cur_addr_hi << 16) | pb.audio_addr.cur_addr_lo;
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u32 frac = pb.src.cur_addr_frac;
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// =============
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// =======================================================================================
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// Handle no-src streams - No src streams have pb.src_type == 2 and have pb.src.ratio_hi = 0
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// and pb.src.ratio_lo = 0. We handle that by setting the sampling ratio integer to 1. This
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// makes samplePos update in the correct way. I'm unsure how we are actually supposed to
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// detect that this setting. Updates did not fix this automatically.
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// ---------------------------------------------------------------------------------------
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// Stream settings
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// src_type = 2 (most other games have src_type = 0)
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// ------------
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// Affected games:
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// Baten Kaitos - Eternal Wings (2003)
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// Baten Kaitos - Origins (2006)?
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// Soul Calibur 2: The movie music use src_type 2 but it needs no adjustment, perhaps
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// the sound format plays in to, Baten use ADPCM SC2 use PCM16
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// ------------
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if(pb.src_type == 2 && (pb.src.ratio_hi == 0 && pb.src.ratio_lo == 0))
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{
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pb.src.ratio_hi = 1;
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}
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// =============
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// =======================================================================================
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// Games that use looping to play non-looping music streams - SSBM has info in all
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// pb.adpcm_loop_info parameters but has pb.audio_addr.looping = 0. If we treat these streams
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// like any other looping streams the music works. I'm unsure how we are actually supposed to
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// detect that these kinds of blocks should be looping. It seems like pb.mixer_control == 0 may
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// identify these types of blocks. Updates did not write any looping values.
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// --------------
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if(
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(pb.adpcm_loop_info.pred_scale || pb.adpcm_loop_info.yn1 || pb.adpcm_loop_info.yn2)
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&& pb.mixer_control == 0
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&& gSSBM
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)
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{
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pb.audio_addr.looping = 1;
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}
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// ==============
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// =======================================================================================
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// Walk through _iSize. _iSize = numSamples. If the game goes slow _iSize will be higher to
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// compensate for that. _iSize can be as low as 100 or as high as 2000 some cases.
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for (int s = 0; s < _iSize; s++)
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{
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int sample = 0;
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frac += ratio;
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u32 newSamplePos = samplePos + (frac >> 16); //whole number of frac
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// =======================================================================================
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// Process sample format
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// --------------
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switch (pb.audio_addr.sample_format)
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{
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case AUDIOFORMAT_PCM8:
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pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
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pb.adpcm.yn1 = ((s8)g_dspInitialize.pARAM_Read_U8(samplePos)) << 8;
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if (pb.src_type == SRCTYPE_NEAREST)
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{
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sample = pb.adpcm.yn1;
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}
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else //linear interpolation
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{
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sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;
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}
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samplePos = newSamplePos;
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break;
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case AUDIOFORMAT_PCM16:
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pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
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pb.adpcm.yn1 = (s16)(u16)((g_dspInitialize.pARAM_Read_U8(samplePos * 2) << 8) | (g_dspInitialize.pARAM_Read_U8((samplePos * 2 + 1))));
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if (pb.src_type == SRCTYPE_NEAREST)
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sample = pb.adpcm.yn1;
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else //linear interpolation
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sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;
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samplePos = newSamplePos;
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break;
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case AUDIOFORMAT_ADPCM:
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sample = ADPCM_Step(pb, samplePos, newSamplePos, frac);
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break;
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default:
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break;
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}
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// ================
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// =======================================================================================
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// Volume control
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frac &= 0xffff;
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int vol = pb.vol_env.cur_volume >> 9;
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sample = sample * vol >> 8;
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if (pb.mixer_control & MIXCONTROL_RAMPING)
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{
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int x = pb.vol_env.cur_volume;
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x += pb.vol_env.cur_volume_delta; // I'm not sure about this, can anybody find a game
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// that use this? Or how does it work?
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if (x < 0) x = 0;
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if (x >= 0x7fff) x = 0x7fff;
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pb.vol_env.cur_volume = x; // maybe not per sample?? :P
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}
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int leftmix = pb.mixer.volume_left >> 5;
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int rightmix = pb.mixer.volume_right >> 5;
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// ===============
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int left = sample * leftmix >> 8;
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int right = sample * rightmix >> 8;
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//adpcm has to walk from oldSamplePos to samplePos here
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templbuffer[s] += left;
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temprbuffer[s] += right;
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if (samplePos >= sampleEnd)
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{
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if (pb.audio_addr.looping == 1)
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{
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samplePos = loopPos;
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if (pb.audio_addr.sample_format == AUDIOFORMAT_ADPCM)
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ADPCM_Loop(pb);
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}
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else
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{
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pb.running = 0;
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break;
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}
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}
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} // end of the _iSize loop
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// ============
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if (gVolume) // allow us to turn this off in the debugger
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{
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pb.mixer.volume_left = ADPCM_Vol(pb.mixer.volume_left, pb.mixer.unknown, pb.mixer_control);
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pb.mixer.volume_right = ADPCM_Vol(pb.mixer.volume_right, pb.mixer.unknown2, pb.mixer_control);
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}
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pb.src.cur_addr_frac = (u16)frac;
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pb.audio_addr.cur_addr_hi = samplePos >> 16;
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pb.audio_addr.cur_addr_lo = (u16)samplePos;
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}
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}
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void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
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{
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AXParamBlock PBs[NUMBER_OF_PBS];
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// read out pbs
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int numberOfPBs = ReadOutPBs(1, PBs, NUMBER_OF_PBS);
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if (_iSize > 1024 * 1024)
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_iSize = 1024 * 1024;
|
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memset(templbuffer, 0, _iSize * sizeof(int));
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memset(temprbuffer, 0, _iSize * sizeof(int));
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// read out pbs
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int numberOfPBs = ReadOutPBs(1, PBs, NUMBER_OF_PBS);
|
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|
||||
#ifdef _WIN32
|
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ratioFactor = 32000.0f / (float)DSound::DSound_GetSampleRate();
|
||||
#else
|
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ratioFactor = 32000.0f / 44100.0f;
|
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#endif
|
||||
|
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// write logging data to debugger
|
||||
if(m_frame)
|
||||
if (m_frame)
|
||||
{
|
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CUCode_AX::Logging(_pBuffer, _iSize, 0);
|
||||
}
|
||||
|
||||
|
||||
|
||||
for (int i = 0; i < numberOfPBs; i++)
|
||||
{
|
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AXParamBlock& pb = PBs[i];
|
||||
|
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// get necessary values
|
||||
const u32 sampleEnd = (pb.audio_addr.end_addr_hi << 16) | pb.audio_addr.end_addr_lo;
|
||||
const u32 loopPos = (pb.audio_addr.loop_addr_hi << 16) | pb.audio_addr.loop_addr_lo;
|
||||
const u32 updaddr = (u32)(pb.updates.data_hi << 16) | pb.updates.data_lo;
|
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const u16 updpar = Memory_Read_U16(updaddr);
|
||||
const u16 upddata = Memory_Read_U16(updaddr + 2);
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
/*
|
||||
Fix problems introduced with the SSBM fix - Sometimes when a music stream ended sampleEnd
|
||||
would become extremely high and the game would play random sound data from ARAM resulting in
|
||||
a strange noise. This should take care of that. - Some games (Monkey Ball 1 and Tales of
|
||||
Symphonia and other) also had one odd last block with a strange high loopPos and strange
|
||||
num_updates values, the loopPos limit turns those off also. - Please report any side effects.
|
||||
*/
|
||||
// ------------
|
||||
if (
|
||||
(sampleEnd > 0x10000000 || loopPos > 0x10000000)
|
||||
&& gSSBMremedy1
|
||||
)
|
||||
{
|
||||
pb.running = 0;
|
||||
|
||||
// also reset all values if it makes any difference
|
||||
pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
|
||||
pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
|
||||
pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
|
||||
|
||||
pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
|
||||
pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
|
||||
|
||||
pb.audio_addr.looping = 0;
|
||||
pb.adpcm_loop_info.pred_scale = 0;
|
||||
pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
|
||||
}
|
||||
|
||||
/*
|
||||
// the fact that no settings are reset (except running) after a SSBM type music stream or another
|
||||
looping block (for example in Battle Stadium DON) has ended could cause loud garbled sound to be
|
||||
played from one or more blocks. Perhaps it was in conjunction with the old sequenced music fix below,
|
||||
I'm not sure. This was an attempt to prevent that anyway by resetting all. But I'm not sure if this
|
||||
is needed anymore. Please try to play SSBM without it and see if it works anyway.
|
||||
*/
|
||||
if (
|
||||
// detect blocks that have recently been running that we should reset
|
||||
pb.running == 0 && pb.audio_addr.looping == 1
|
||||
//pb.running == 0 && pb.adpcm_loop_info.pred_scale
|
||||
|
||||
// this prevents us from ruining sequenced music blocks, may not be needed
|
||||
/*
|
||||
&& !(pb.updates.num_updates[0] || pb.updates.num_updates[1] || pb.updates.num_updates[2]
|
||||
|| pb.updates.num_updates[3] || pb.updates.num_updates[4])
|
||||
*/
|
||||
&& !(updpar || upddata)
|
||||
|
||||
&& pb.mixer_control == 0 // only use this in SSBM
|
||||
|
||||
&& gSSBMremedy2 // let us turn this fix on and off
|
||||
)
|
||||
{
|
||||
// reset the detection values
|
||||
pb.audio_addr.looping = 0;
|
||||
pb.adpcm_loop_info.pred_scale = 0;
|
||||
pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
|
||||
|
||||
//pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
|
||||
//pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
|
||||
//pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
|
||||
|
||||
//pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
|
||||
//pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
|
||||
}
|
||||
|
||||
// =============
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Reset all values
|
||||
// ------------
|
||||
if (gReset
|
||||
&& (pb.running || pb.audio_addr.looping || pb.adpcm_loop_info.pred_scale)
|
||||
)
|
||||
{
|
||||
pb.running = 0;
|
||||
|
||||
pb.audio_addr.cur_addr_hi = 0; pb.audio_addr.cur_addr_lo = 0;
|
||||
pb.audio_addr.end_addr_hi = 0; pb.audio_addr.end_addr_lo = 0;
|
||||
pb.audio_addr.loop_addr_hi = 0; pb.audio_addr.loop_addr_lo = 0;
|
||||
|
||||
pb.src.cur_addr_frac = 0; PBs[i].src.ratio_hi = 0; PBs[i].src.ratio_lo = 0;
|
||||
pb.adpcm.pred_scale = 0; pb.adpcm.yn1 = 0; pb.adpcm.yn2 = 0;
|
||||
|
||||
pb.audio_addr.looping = 0;
|
||||
pb.adpcm_loop_info.pred_scale = 0;
|
||||
pb.adpcm_loop_info.yn1 = 0; pb.adpcm_loop_info.yn2 = 0;
|
||||
}
|
||||
// =============
|
||||
|
||||
|
||||
if (pb.running)
|
||||
{
|
||||
// =======================================================================================
|
||||
// Set initial parameters
|
||||
// ------------
|
||||
//constants
|
||||
const u32 ratio = (u32)(((pb.src.ratio_hi << 16) + pb.src.ratio_lo) * ratioFactor);
|
||||
|
||||
//variables
|
||||
u32 samplePos = (pb.audio_addr.cur_addr_hi << 16) | pb.audio_addr.cur_addr_lo;
|
||||
u32 frac = pb.src.cur_addr_frac;
|
||||
// =============
|
||||
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Handle no-src streams - No src streams have pb.src_type == 2 and have pb.src.ratio_hi = 0
|
||||
// and pb.src.ratio_lo = 0. We handle that by setting the sampling ratio integer to 1. This
|
||||
// makes samplePos update in the correct way. I'm unsure how we are actually supposed to
|
||||
// detect that this setting. Updates did not fix this automatically.
|
||||
// ---------------------------------------------------------------------------------------
|
||||
// Stream settings
|
||||
// src_type = 2 (most other games have src_type = 0)
|
||||
// ------------
|
||||
// Affected games:
|
||||
// Baten Kaitos - Eternal Wings (2003)
|
||||
// Baten Kaitos - Origins (2006)?
|
||||
// Soul Calibur 2: The movie music use src_type 2 but it needs no adjustment, perhaps
|
||||
// the sound format plays in to, Baten use ADPCM SC2 use PCM16
|
||||
// ------------
|
||||
if(pb.src_type == 2 && (pb.src.ratio_hi == 0 && pb.src.ratio_lo == 0))
|
||||
{
|
||||
pb.src.ratio_hi = 1;
|
||||
}
|
||||
// =============
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Games that use looping to play non-looping music streams - SSBM has info in all
|
||||
// pb.adpcm_loop_info parameters but has pb.audio_addr.looping = 0. If we treat these streams
|
||||
// like any other looping streams the music works. I'm unsure how we are actually supposed to
|
||||
// detect that these kinds of blocks should be looping. It seems like pb.mixer_control == 0 may
|
||||
// identify these types of blocks. Updates did not write any looping values.
|
||||
// --------------
|
||||
if(
|
||||
(pb.adpcm_loop_info.pred_scale || pb.adpcm_loop_info.yn1 || pb.adpcm_loop_info.yn2)
|
||||
&& pb.mixer_control == 0
|
||||
&& gSSBM
|
||||
)
|
||||
{
|
||||
pb.audio_addr.looping = 1;
|
||||
}
|
||||
// ==============
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Walk through _iSize. _iSize = numSamples. If the game goes slow _iSize will be higher to
|
||||
// compensate for that. _iSize can be as low as 100 or as high as 2000 some cases.
|
||||
for (int s = 0; s < _iSize; s++)
|
||||
{
|
||||
int sample = 0;
|
||||
frac += ratio;
|
||||
u32 newSamplePos = samplePos + (frac >> 16); //whole number of frac
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Process sample format
|
||||
// --------------
|
||||
switch (pb.audio_addr.sample_format)
|
||||
{
|
||||
case AUDIOFORMAT_PCM8:
|
||||
pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
|
||||
pb.adpcm.yn1 = ((s8)g_dspInitialize.pARAM_Read_U8(samplePos)) << 8;
|
||||
|
||||
if (pb.src_type == SRCTYPE_NEAREST)
|
||||
{
|
||||
sample = pb.adpcm.yn1;
|
||||
}
|
||||
else //linear interpolation
|
||||
{
|
||||
sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;
|
||||
}
|
||||
|
||||
samplePos = newSamplePos;
|
||||
break;
|
||||
|
||||
case AUDIOFORMAT_PCM16:
|
||||
pb.adpcm.yn2 = pb.adpcm.yn1; //save last sample
|
||||
pb.adpcm.yn1 = (s16)(u16)((g_dspInitialize.pARAM_Read_U8(samplePos * 2) << 8) | (g_dspInitialize.pARAM_Read_U8((samplePos * 2 + 1))));
|
||||
if (pb.src_type == SRCTYPE_NEAREST)
|
||||
sample = pb.adpcm.yn1;
|
||||
else //linear interpolation
|
||||
sample = (pb.adpcm.yn1 * (u16)frac + pb.adpcm.yn2 * (u16)(0xFFFF - frac)) >> 16;
|
||||
|
||||
samplePos = newSamplePos;
|
||||
break;
|
||||
|
||||
case AUDIOFORMAT_ADPCM:
|
||||
sample = ADPCM_Step(pb, samplePos, newSamplePos, frac);
|
||||
break;
|
||||
|
||||
default:
|
||||
break;
|
||||
}
|
||||
// ================
|
||||
|
||||
|
||||
// =======================================================================================
|
||||
// Volume control
|
||||
frac &= 0xffff;
|
||||
|
||||
int vol = pb.vol_env.cur_volume >> 9;
|
||||
sample = sample * vol >> 8;
|
||||
|
||||
if (pb.mixer_control & MIXCONTROL_RAMPING)
|
||||
{
|
||||
int x = pb.vol_env.cur_volume;
|
||||
x += pb.vol_env.cur_volume_delta; // I'm not sure about this, can anybody find a game
|
||||
// that use this? Or how does it work?
|
||||
if (x < 0) x = 0;
|
||||
if (x >= 0x7fff) x = 0x7fff;
|
||||
pb.vol_env.cur_volume = x; // maybe not per sample?? :P
|
||||
}
|
||||
|
||||
int leftmix = pb.mixer.volume_left >> 5;
|
||||
int rightmix = pb.mixer.volume_right >> 5;
|
||||
// ===============
|
||||
|
||||
|
||||
int left = sample * leftmix >> 8;
|
||||
int right = sample * rightmix >> 8;
|
||||
|
||||
//adpcm has to walk from oldSamplePos to samplePos here
|
||||
templbuffer[s] += left;
|
||||
temprbuffer[s] += right;
|
||||
|
||||
if (samplePos >= sampleEnd)
|
||||
{
|
||||
if (pb.audio_addr.looping == 1)
|
||||
{
|
||||
samplePos = loopPos;
|
||||
if (pb.audio_addr.sample_format == AUDIOFORMAT_ADPCM)
|
||||
ADPCM_Loop(pb);
|
||||
}
|
||||
else
|
||||
{
|
||||
pb.running = 0;
|
||||
break;
|
||||
}
|
||||
}
|
||||
} // end of the _iSize loop
|
||||
// ============
|
||||
|
||||
if (gVolume) // allow us to turn this off in the debugger
|
||||
{
|
||||
pb.mixer.volume_left = ADPCM_Vol(pb.mixer.volume_left, pb.mixer.unknown, pb.mixer_control);
|
||||
pb.mixer.volume_right = ADPCM_Vol(pb.mixer.volume_right, pb.mixer.unknown2, pb.mixer_control);
|
||||
}
|
||||
|
||||
pb.src.cur_addr_frac = (u16)frac;
|
||||
pb.audio_addr.cur_addr_hi = samplePos >> 16;
|
||||
pb.audio_addr.cur_addr_lo = (u16)samplePos;
|
||||
}
|
||||
MixAddVoice(pb, templbuffer, temprbuffer, _iSize);
|
||||
}
|
||||
|
||||
// write back out pbs
|
||||
WriteBackPBs(PBs, numberOfPBs);
|
||||
|
||||
for (int i = 0; i < _iSize; i++)
|
||||
{
|
||||
// Clamp into 16-bit. Maybe we should add a volume compressor here.
|
||||
@ -467,9 +457,6 @@ void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
|
||||
*_pBuffer++ = right;
|
||||
}
|
||||
|
||||
// write back out pbs
|
||||
WriteBackPBs(PBs, numberOfPBs);
|
||||
|
||||
// write logging data to debugger again after the update
|
||||
if (m_frame)
|
||||
{
|
||||
@ -477,6 +464,7 @@ void CUCode_AX::MixAdd(short* _pBuffer, int _iSize)
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
void CUCode_AX::Update()
|
||||
{
|
||||
// check if we have to sent something
|
||||
|
||||
Reference in New Issue
Block a user