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source-engine/engine/audio/snd_mix.cpp
nillerusr bca0ae8c59 engine/audio: fix build
2022-10-12 05:31:18 +03:00

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//========= Copyright Valve Corporation, All rights reserved. ============//
//
// Purpose: Portable code to mix sounds for snd_dma.cpp.
//
//=============================================================================//
#include "audio_pch.h"
#include "mouthinfo.h"
#include "cl_main.h"
#include "icliententitylist.h"
#include "icliententity.h"
#include "sys_dll.h"
#include "video/ivideoservices.h"
#include "engine/IEngineSound.h"
#if defined( REPLAY_ENABLED )
#include "demo.h"
#include "replay_internal.h"
#endif
#ifdef GNUC
// we don't suport the ASM in this file right now under GCC, fallback to C libs
#undef id386
#endif
// memdbgon must be the last include file in a .cpp file!!!
#include "tier0/memdbgon.h"
#if defined( REPLAY_ENABLED )
extern IReplayMovieManager *g_pReplayMovieManager;
#endif
#if defined(_WIN32) && id386
// warning C4731: frame pointer register 'ebp' modified by inline assembly code
#pragma warning(disable : 4731)
#endif
// NOTE: !!!!!! YOU MUST UPDATE SND_MIXA.S IF THIS VALUE IS CHANGED !!!!!
#define SND_SCALE_BITS 7
#define SND_SCALE_SHIFT (8-SND_SCALE_BITS)
#define SND_SCALE_LEVELS (1<<SND_SCALE_BITS)
#define SND_SCALE_BITS16 8
#define SND_SCALE_SHIFT16 (8-SND_SCALE_BITS16)
#define SND_SCALE_LEVELS16 (1<<SND_SCALE_BITS16)
void Snd_WriteLinearBlastStereo16(void);
void SND_PaintChannelFrom8( portable_samplepair_t *pOutput, int *volume, byte *pData8, int count );
bool Con_IsVisible( void );
void SND_RecordBuffer( void );
bool DSP_RoomDSPIsOff( void );
bool BChannelLowVolume( channel_t *pch, int vol_min );
void ChannelCopyVolumes( channel_t *pch, int *pvolume_dest, int ivol_start, int cvol );
float ChannelLoudestCurVolume( const channel_t * RESTRICT pch );
extern int g_soundtime;
extern float host_frametime;
extern float host_frametime_unbounded;
#if !defined( NO_VOICE )
extern int g_SND_VoiceOverdriveInt;
#endif
extern ConVar dsp_room;
extern ConVar dsp_water;
extern ConVar dsp_player;
extern ConVar dsp_facingaway;
extern ConVar snd_showstart;
extern ConVar dsp_automatic;
extern ConVar snd_pitchquality;
extern float DSP_ROOM_MIX;
extern float DSP_NOROOM_MIX;
portable_samplepair_t *g_paintbuffer;
// temp paintbuffer - not included in main list of paintbuffers
// NOTE: this paintbuffer is also used as a copy buffer by interpolating pitch
// shift routines. Decreasing TEMP_COPY_BUFFER_SIZE (or PAINTBUFFER_MEM_SIZE)
// will decrease the maximum pitch level (current 4.0)!
portable_samplepair_t *g_temppaintbuffer = NULL;
CUtlVector< paintbuffer_t > g_paintBuffers;
// pointer to current paintbuffer (front and reare), used by all mixing, upsampling and dsp routines
portable_samplepair_t *g_curpaintbuffer = NULL;
portable_samplepair_t *g_currearpaintbuffer = NULL;
portable_samplepair_t *g_curcenterpaintbuffer = NULL;
bool g_bdirectionalfx;
bool g_bDspOff;
float g_dsp_volume;
// dsp performance timing
unsigned g_snd_call_time_debug = 0;
unsigned g_snd_time_debug = 0;
unsigned g_snd_count_debug = 0;
unsigned g_snd_samplecount = 0;
unsigned g_snd_frametime = 0;
unsigned g_snd_frametime_total = 0;
int g_snd_profile_type = 0; // type 1 dsp, type 2 mixer, type 3 load sound, type 4 all sound
#define FILTERTYPE_NONE 0
#define FILTERTYPE_LINEAR 1
#define FILTERTYPE_CUBIC 2
// filter memory for upsampling
portable_samplepair_t cubicfilter1[3] = {{0,0},{0,0},{0,0}};
portable_samplepair_t cubicfilter2[3] = {{0,0},{0,0},{0,0}};
portable_samplepair_t linearfilter1[1] = {{0,0}};
portable_samplepair_t linearfilter2[1] = {{0,0}};
portable_samplepair_t linearfilter3[1] = {{0,0}};
portable_samplepair_t linearfilter4[1] = {{0,0}};
portable_samplepair_t linearfilter5[1] = {{0,0}};
portable_samplepair_t linearfilter6[1] = {{0,0}};
portable_samplepair_t linearfilter7[1] = {{0,0}};
portable_samplepair_t linearfilter8[1] = {{0,0}};
int snd_scaletable[SND_SCALE_LEVELS][256]; // 32k*4 = 128K
int *snd_p, snd_linear_count, snd_vol;
short *snd_out;
extern int DSP_Alloc( int ipset, float xfade, int cchan );
bool DSP_CheckDspAutoEnabled( void );
int Get_idsp_room ( void );
int dsp_room_GetInt ( void );
void DSP_SetDspAuto( int dsp_preset );
bool DSP_CheckDspAutoEnabled( void );
void MIX_ScalePaintBuffer( int bufferIndex, int count, float fgain );
bool IsReplayRendering()
{
#if defined( REPLAY_ENABLED )
return g_pReplayMovieManager && g_pReplayMovieManager->IsRendering();
#else
return false;
#endif
}
//-----------------------------------------------------------------------------
// Free allocated memory buffers
//-----------------------------------------------------------------------------
void MIX_FreeAllPaintbuffers(void)
{
if ( g_paintBuffers.Count() )
{
if ( g_temppaintbuffer )
{
_aligned_free( g_temppaintbuffer );
g_temppaintbuffer = NULL;
}
for ( int i = 0; i < g_paintBuffers.Count(); i++ )
{
if ( g_paintBuffers[i].pbuf )
{
_aligned_free( g_paintBuffers[i].pbuf );
}
if ( g_paintBuffers[i].pbufrear )
{
_aligned_free( g_paintBuffers[i].pbufrear );
}
if ( g_paintBuffers[i].pbufcenter )
{
_aligned_free( g_paintBuffers[i].pbufcenter );
}
}
g_paintBuffers.RemoveAll();
}
}
void MIX_InitializePaintbuffer( paintbuffer_t *pPaintBuffer, bool bSurround, bool bSurroundCenter )
{
V_memset( pPaintBuffer, 0, sizeof( *pPaintBuffer ) );
pPaintBuffer->pbuf = (portable_samplepair_t *)_aligned_malloc( PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t), 16 );
V_memset( pPaintBuffer->pbuf, 0, PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t) );
if ( bSurround )
{
pPaintBuffer->pbufrear = (portable_samplepair_t *)_aligned_malloc( PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t), 16 );
V_memset( pPaintBuffer->pbufrear, 0, PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t) );
}
if ( bSurroundCenter )
{
pPaintBuffer->pbufcenter = (portable_samplepair_t *)_aligned_malloc( PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t), 16 );
V_memset( pPaintBuffer->pbufcenter, 0, PAINTBUFFER_MEM_SIZE*sizeof(portable_samplepair_t) );
}
}
//-----------------------------------------------------------------------------
// Allocate memory buffers
// Initialize paintbuffers array, set current paint buffer to main output buffer SOUND_BUFFER_PAINT
//-----------------------------------------------------------------------------
bool MIX_InitAllPaintbuffers(void)
{
bool bSurround;
bool bSurroundCenter;
bSurroundCenter = g_AudioDevice->IsSurroundCenter();
bSurround = g_AudioDevice->IsSurround() || bSurroundCenter;
g_temppaintbuffer = (portable_samplepair_t*)_aligned_malloc( TEMP_COPY_BUFFER_SIZE*sizeof(portable_samplepair_t), 16 );
V_memset( g_temppaintbuffer, 0, TEMP_COPY_BUFFER_SIZE*sizeof(portable_samplepair_t) );
while ( g_paintBuffers.Count() < SOUND_BUFFER_BASETOTAL )
{
int nIndex = g_paintBuffers.AddToTail();
MIX_InitializePaintbuffer( &(g_paintBuffers[ nIndex ]), bSurround, bSurroundCenter );
}
g_paintbuffer = g_paintBuffers[SOUND_BUFFER_PAINT].pbuf;
// buffer flags
g_paintBuffers[SOUND_BUFFER_ROOM].flags = SOUND_BUSS_ROOM;
g_paintBuffers[SOUND_BUFFER_FACING].flags = SOUND_BUSS_FACING;
g_paintBuffers[SOUND_BUFFER_FACINGAWAY].flags = SOUND_BUSS_FACINGAWAY;
g_paintBuffers[SOUND_BUFFER_SPEAKER].flags = SOUND_BUSS_SPEAKER;
g_paintBuffers[SOUND_BUFFER_DRY].flags = SOUND_BUSS_DRY;
// buffer surround sound flag
g_paintBuffers[SOUND_BUFFER_PAINT].fsurround = bSurround;
g_paintBuffers[SOUND_BUFFER_FACING].fsurround = bSurround;
g_paintBuffers[SOUND_BUFFER_FACINGAWAY].fsurround = bSurround;
g_paintBuffers[SOUND_BUFFER_DRY].fsurround = bSurround;
// buffer 5 channel surround sound flag
g_paintBuffers[SOUND_BUFFER_PAINT].fsurround_center = bSurroundCenter;
g_paintBuffers[SOUND_BUFFER_FACING].fsurround_center = bSurroundCenter;
g_paintBuffers[SOUND_BUFFER_FACINGAWAY].fsurround_center = bSurroundCenter;
g_paintBuffers[SOUND_BUFFER_DRY].fsurround_center = bSurroundCenter;
// room buffer mixes down to mono or stereo, never to 4 or 5 ch
g_paintBuffers[SOUND_BUFFER_ROOM].fsurround = false;
g_paintBuffers[SOUND_BUFFER_ROOM].fsurround_center = false;
// speaker buffer mixes to mono
g_paintBuffers[SOUND_BUFFER_SPEAKER].fsurround = false;
g_paintBuffers[SOUND_BUFFER_SPEAKER].fsurround_center = false;
MIX_SetCurrentPaintbuffer( SOUND_BUFFER_PAINT );
return true;
}
// called before loading samples to mix - cap the mix rate (ie: pitch) so that
// we never overflow the mix copy buffer.
double MIX_GetMaxRate( double rate, int sampleCount )
{
if (rate <= 2.0)
return rate;
// copybuf_bytes = rate_max * samples_max * samplesize_max
// so:
// rate_max = copybuf_bytes / (samples_max * samplesize_max )
double samplesize_max = 4.0; // stereo 16bit samples
double copybuf_bytes = (double)(TEMP_COPY_BUFFER_SIZE * sizeof(portable_samplepair_t));
double samples_max = (double)(PAINTBUFFER_SIZE);
double rate_max = copybuf_bytes / (samples_max * samplesize_max);
// make sure sampleCount is never greater than paintbuffer samples
// (this should have been set up in MIX_PaintChannels)
Assert (sampleCount <= PAINTBUFFER_SIZE);
return fpmin( rate, rate_max );
}
// Transfer (endtime - lpaintedtime) stereo samples in pfront out to hardware
// pfront - pointer to stereo paintbuffer - 32 bit samples, interleaved stereo
// lpaintedtime - total number of 32 bit stereo samples previously output to hardware
// endtime - total number of 32 bit stereo samples currently mixed in paintbuffer
void S_TransferStereo16( void *pOutput, const portable_samplepair_t *pfront, int lpaintedtime, int endtime )
{
int lpos;
if ( IsX360() )
{
// not the right path for 360
Assert( 0 );
return;
}
Assert( pOutput );
snd_vol = S_GetMasterVolume()*256;
snd_p = (int *)pfront;
// get size of output buffer in full samples (LR pairs)
int samplePairCount = g_AudioDevice->DeviceSampleCount() >> 1;
int sampleMask = samplePairCount - 1;
bool bShouldPlaySound = !cl_movieinfo.IsRecording() && !IsReplayRendering();
while ( lpaintedtime < endtime )
{
// pbuf can hold 16384, 16 bit L/R samplepairs.
// lpaintedtime - where to start painting into dma buffer.
// (modulo size of dma buffer for current position).
// handle recirculating buffer issues
// lpos - samplepair index into dma buffer. First samplepair from paintbuffer to be xfered here.
lpos = lpaintedtime & sampleMask;
// snd_out is L/R sample index into dma buffer. First L sample from paintbuffer goes here.
snd_out = (short *)pOutput + (lpos<<1);
// snd_linear_count is number of samplepairs between end of dma buffer and xfer start index.
snd_linear_count = samplePairCount - lpos;
// clamp snd_linear_count to be only as many samplepairs premixed
if ( snd_linear_count > endtime - lpaintedtime )
{
// endtime - lpaintedtime = number of premixed sample pairs ready for xfer.
snd_linear_count = endtime - lpaintedtime;
}
// snd_linear_count is now number of mono 16 bit samples (L and R) to xfer.
snd_linear_count <<= 1;
// write a linear blast of samples
SND_RecordBuffer();
if ( bShouldPlaySound )
{
// transfer 16bit samples from snd_p into snd_out, multiplying each sample by volume.
Snd_WriteLinearBlastStereo16();
}
// advance paintbuffer pointer
snd_p += snd_linear_count;
// advance lpaintedtime by number of samplepairs just xfered.
lpaintedtime += (snd_linear_count>>1);
}
}
// Transfer contents of main paintbuffer pfront out to
// device. Perform volume multiply on each sample.
void S_TransferPaintBuffer(void *pOutput, const portable_samplepair_t *pfront, int lpaintedtime, int endtime)
{
int out_idx; // mono sample index
int count; // number of mono samples to output
int out_mask;
int step;
int val;
int nSoundVol;
const int *p;
if ( IsX360() )
{
// not the right path for 360
Assert( 0 );
return;
}
Assert( pOutput );
p = (const int *) pfront;
count = ((endtime - lpaintedtime) * g_AudioDevice->DeviceChannels());
out_mask = g_AudioDevice->DeviceSampleCount() - 1;
// 44k: remove old 22k sound support << HISPEED_DMA
// out_idx = ((paintedtime << HISPEED_DMA) * g_AudioDevice->DeviceChannels()) & out_mask;
out_idx = (lpaintedtime * g_AudioDevice->DeviceChannels()) & out_mask;
step = 3 - g_AudioDevice->DeviceChannels(); // mono output buffer - step 2, stereo - step 1
nSoundVol = S_GetMasterVolume()*256;
if (g_AudioDevice->DeviceSampleBits() == 16)
{
short *out = (short *) pOutput;
while (count--)
{
val = (*p * nSoundVol) >> 8;
p+= step;
val = CLIP(val);
out[out_idx] = val;
out_idx = (out_idx + 1) & out_mask;
}
}
else if (g_AudioDevice->DeviceSampleBits() == 8)
{
unsigned char *out = (unsigned char *) pOutput;
while (count--)
{
val = (*p * nSoundVol) >> 8;
p+= step;
val = CLIP(val);
out[out_idx] = (val>>8) + 128;
out_idx = (out_idx + 1) & out_mask;
}
}
}
/*
===============================================================================
CHANNEL MIXING
===============================================================================
*/
// free channel so that it may be allocated by the
// next request to play a sound. If sound is a
// word in a sentence, release the sentence.
// Works for static, dynamic, sentence and stream sounds
void S_FreeChannel(channel_t *ch)
{
// Don't reenter in here (can happen inside voice code).
if ( ch->flags.m_bIsFreeingChannel )
return;
ch->flags.m_bIsFreeingChannel = true;
SND_CloseMouth(ch);
g_pSoundServices->OnSoundStopped( ch->guid, ch->soundsource, ch->entchannel, ch->sfx->getname() );
ch->flags.isSentence = false;
// Msg("End sound %s\n", ch->sfx->getname() );
delete ch->pMixer;
ch->pMixer = NULL;
ch->sfx = NULL;
// zero all data in channel
g_ActiveChannels.Remove( ch );
Q_memset(ch, 0, sizeof(channel_t));
}
// Mix all channels into active paintbuffers until paintbuffer is full or 'endtime' is reached.
// endtime: time in 44khz samples to mix
// rate: ignore samples which are not natively at this rate (for multipass mixing/filtering)
// if rate == SOUND_ALL_RATES then mix all samples this pass
// flags: if SOUND_MIX_DRY, then mix only samples with channel flagged as 'dry'
// outputRate: target mix rate for all samples. Note, if outputRate = SOUND_DMA_SPEED, then
// this routine will fill the paintbuffer to endtime. Otherwise, fewer samples are mixed.
// if (endtime - paintedtime) is not aligned on boundaries of 4,
// we'll miss data if outputRate < SOUND_DMA_SPEED!
void MIX_MixChannelsToPaintbuffer( CChannelList &list, int endtime, int flags, int rate, int outputRate )
{
VPROF( "MixChannelsToPaintbuffer" );
int i;
int sampleCount;
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s c:%d %d/%d", __FUNCTION__, list.Count(), rate, outputRate );
// mix each channel into paintbuffer
// validate parameters
Assert( outputRate <= SOUND_DMA_SPEED );
Assert( !((endtime - g_paintedtime) & 0x3) || (outputRate == SOUND_DMA_SPEED) ); // make sure we're not discarding data
// 44k: try to mix this many samples at outputRate
sampleCount = ( endtime - g_paintedtime ) / ( SOUND_DMA_SPEED / outputRate );
if ( sampleCount <= 0 )
return;
// Apply a global pitch shift if we're playing back a time-scaled replay
float flGlobalPitchScale = 1.0f;
#if defined( REPLAY_ENABLED )
extern IDemoPlayer *g_pReplayDemoPlayer;
if ( demoplayer->IsPlayingBack() && demoplayer == g_pReplayDemoPlayer )
{
// adjust time scale if playing back demo
flGlobalPitchScale = demoplayer->GetPlaybackTimeScale();
}
#endif
for ( i = list.Count(); --i >= 0; )
{
channel_t *ch = list.GetChannel( i );
Assert( ch->sfx );
// must never have a 'dry' and 'speaker' set - causes double mixing & double data reading
Assert ( !( ( ch->flags.bdry && ch->flags.bSpeaker ) || ( ch->flags.bdry && ch->special_dsp != 0 ) ) );
// if mixing with SOUND_MIX_DRY flag, ignore (don't even load) all channels not flagged as 'dry'
if ( flags == SOUND_MIX_DRY )
{
if ( !ch->flags.bdry )
continue;
}
// if mixing with SOUND_MIX_WET flag, ignore (don't even load) all channels flagged as 'dry' or 'speaker'
if ( flags == SOUND_MIX_WET )
{
if ( ch->flags.bdry || ch->flags.bSpeaker || ch->special_dsp != 0 )
continue;
}
// if mixing with SOUND_MIX_SPEAKER flag, ignore (don't even load) all channels not flagged as 'speaker'
if ( flags == SOUND_MIX_SPEAKER )
{
if ( !ch->flags.bSpeaker )
continue;
}
// if mixing with SOUND_MIX_SPEAKER flag, ignore (don't even load) all channels not flagged as 'speaker'
if ( flags == SOUND_MIX_SPECIAL_DSP )
{
if ( ch->special_dsp == 0 )
continue;
}
// multipass mixing - only mix samples of specified sample rate
switch ( rate )
{
case SOUND_11k:
case SOUND_22k:
case SOUND_44k:
if ( rate != ch->sfx->pSource->SampleRate() )
continue;
break;
default:
case SOUND_ALL_RATES:
break;
}
// Tracker 20771, if breen is speaking through the monitor, the client doesn't have an entity
// for the "soundsource" but we still need the lipsync to pause if the game is paused. Therefore
// I changed SND_IsMouth to look for any .wav on any channels which has sentence data
bool bIsMouth = SND_IsMouth(ch);
bool bShouldPause = IsX360() ? !ch->sfx->m_bIsUISound : bIsMouth;
// Tracker 14637: Pausing the game pauses voice sounds, but not other sounds...
if ( bShouldPause && g_pSoundServices->IsGamePaused() )
{
continue;
}
if ( bIsMouth )
{
if ( ( ch->soundsource == SOUND_FROM_UI_PANEL ) || entitylist->GetClientEntity(ch->soundsource) ||
( ch->flags.bSpeaker && entitylist->GetClientEntity( ch->speakerentity ) ) )
{
// UNDONE: recode this as a member function of CAudioMixer
SND_MoveMouth8(ch, ch->sfx->pSource, sampleCount);
}
}
// mix channel to all active paintbuffers:
// mix 'dry' sounds only to dry paintbuffer.
// mix 'speaker' sounds only to speaker paintbuffer.
// mix all other sounds between room, facing & facingaway paintbuffers
// NOTE: must be called once per channel only - consecutive calls retrieve additional data.
float flPitch = ch->pitch;
ch->pitch *= flGlobalPitchScale;
if (list.IsQuashed(i))
{
// If the sound has been silenced as a performance heuristic, quash it.
ch->pMixer->SkipSamples( ch, sampleCount, outputRate, 0 );
// DevMsg("Quashed channel %d (%s)\n", i, ch->sfx->GetFileName());
}
else
{
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "MixDataToDevice" );
ch->pMixer->MixDataToDevice( g_AudioDevice, ch, sampleCount, outputRate, 0 );
}
// restore to original pitch settings
ch->pitch = flPitch;
if ( !ch->pMixer->ShouldContinueMixing() )
{
S_FreeChannel( ch );
list.RemoveChannelFromList(i);
}
if ( (ch->nFreeChannelAtSampleTime > 0 && (int)ch->nFreeChannelAtSampleTime <= endtime) )
{
S_FreeChannel( ch );
list.RemoveChannelFromList(i);
}
}
}
// pass in index -1...count+2, return pointer to source sample in either paintbuffer or delay buffer
inline portable_samplepair_t * S_GetNextpFilter(int i, portable_samplepair_t *pbuffer, portable_samplepair_t *pfiltermem)
{
// The delay buffer is assumed to precede the paintbuffer by 6 duplicated samples
if (i == -1)
return (&(pfiltermem[0]));
if (i == 0)
return (&(pfiltermem[1]));
if (i == 1)
return (&(pfiltermem[2]));
// return from paintbuffer, where samples are doubled.
// even samples are to be replaced with interpolated value.
return (&(pbuffer[(i-2)*2 + 1]));
}
// pass forward over passed in buffer and cubic interpolate all odd samples
// pbuffer: buffer to filter (in place)
// prevfilter: filter memory. NOTE: this must match the filtertype ie: filtercubic[] for FILTERTYPE_CUBIC
// if NULL then perform no filtering. UNDONE: should have a filter memory array type
// count: how many samples to upsample. will become count*2 samples in buffer, in place.
void S_Interpolate2xCubic( portable_samplepair_t *pbuffer, portable_samplepair_t *pfiltermem, int cfltmem, int count )
{
// implement cubic interpolation on 2x upsampled buffer. Effectively delays buffer contents by 2 samples.
// pbuffer: contains samples at 0, 2, 4, 6...
// temppaintbuffer is temp buffer, of same or larger size than a paintbuffer, used to store processed values
// count: number of samples to process in buffer ie: how many samples at 0, 2, 4, 6...
// finpos is the fractional, inpos the integer part.
// finpos = 0.5 for upsampling by 2x
// inpos is the position of the sample
// xm1 = x [inpos - 1];
// x0 = x [inpos + 0];
// x1 = x [inpos + 1];
// x2 = x [inpos + 2];
// a = (3 * (x0-x1) - xm1 + x2) / 2;
// b = 2*x1 + xm1 - (5*x0 + x2) / 2;
// c = (x1 - xm1) / 2;
// y [outpos] = (((a * finpos) + b) * finpos + c) * finpos + x0;
int i, upCount = count << 1;
int a, b, c;
int xm1, x0, x1, x2;
portable_samplepair_t *psamp0;
portable_samplepair_t *psamp1;
portable_samplepair_t *psamp2;
portable_samplepair_t *psamp3;
int outpos = 0;
Assert (upCount <= PAINTBUFFER_SIZE);
// pfiltermem holds 6 samples from previous buffer pass
// process 'count' samples
for ( i = 0; i < count; i++)
{
// get source sample pointer
psamp0 = S_GetNextpFilter(i-1, pbuffer, pfiltermem);
psamp1 = S_GetNextpFilter(i, pbuffer, pfiltermem);
psamp2 = S_GetNextpFilter(i+1, pbuffer, pfiltermem);
psamp3 = S_GetNextpFilter(i+2, pbuffer, pfiltermem);
// write out original sample to interpolation buffer
g_temppaintbuffer[outpos++] = *psamp1;
// get all left samples for interpolation window
xm1 = psamp0->left;
x0 = psamp1->left;
x1 = psamp2->left;
x2 = psamp3->left;
// interpolate
a = (3 * (x0-x1) - xm1 + x2) / 2;
b = 2*x1 + xm1 - (5*x0 + x2) / 2;
c = (x1 - xm1) / 2;
// write out interpolated sample
g_temppaintbuffer[outpos].left = a/8 + b/4 + c/2 + x0;
// get all right samples for window
xm1 = psamp0->right;
x0 = psamp1->right;
x1 = psamp2->right;
x2 = psamp3->right;
// interpolate
a = (3 * (x0-x1) - xm1 + x2) / 2;
b = 2*x1 + xm1 - (5*x0 + x2) / 2;
c = (x1 - xm1) / 2;
// write out interpolated sample, increment output counter
g_temppaintbuffer[outpos++].right = a/8 + b/4 + c/2 + x0;
Assert( outpos <= TEMP_COPY_BUFFER_SIZE );
}
Assert(cfltmem >= 3);
// save last 3 samples from paintbuffer
pfiltermem[0] = pbuffer[upCount - 5];
pfiltermem[1] = pbuffer[upCount - 3];
pfiltermem[2] = pbuffer[upCount - 1];
// copy temppaintbuffer back into paintbuffer
for (i = 0; i < upCount; i++)
pbuffer[i] = g_temppaintbuffer[i];
}
// pass forward over passed in buffer and linearly interpolate all odd samples
// pbuffer: buffer to filter (in place)
// prevfilter: filter memory. NOTE: this must match the filtertype ie: filterlinear[] for FILTERTYPE_LINEAR
// if NULL then perform no filtering.
// count: how many samples to upsample. will become count*2 samples in buffer, in place.
void S_Interpolate2xLinear( portable_samplepair_t *pbuffer, portable_samplepair_t *pfiltermem, int cfltmem, int count )
{
int i, upCount = count<<1;
Assert (upCount <= PAINTBUFFER_SIZE);
Assert (cfltmem >= 1);
// use interpolation value from previous mix
pbuffer[0].left = (pfiltermem->left + pbuffer[0].left) >> 1;
pbuffer[0].right = (pfiltermem->right + pbuffer[0].right) >> 1;
for ( i = 2; i < upCount; i+=2)
{
// use linear interpolation for upsampling
pbuffer[i].left = (pbuffer[i].left + pbuffer[i-1].left) >> 1;
pbuffer[i].right = (pbuffer[i].right + pbuffer[i-1].right) >> 1;
}
// save last value to be played out in buffer
*pfiltermem = pbuffer[upCount - 1];
}
// Optimized routine. 2.27X faster than the above routine
void S_Interpolate2xLinear_2( int count, portable_samplepair_t *pbuffer, portable_samplepair_t *pfiltermem, int cfltmem )
{
Assert (cfltmem >= 1);
int sample = count-1;
int end = (count*2)-1;
portable_samplepair_t *pwrite = &pbuffer[end];
portable_samplepair_t *pread = &pbuffer[sample];
portable_samplepair_t last = pread[0];
pread--;
// PERFORMANCE: Unroll the loop 8 times. This improves speed quite a bit
for ( ;sample >= 8; sample -= 8 )
{
pwrite[0] = last;
pwrite[-1].left = (pread[0].left + last.left)>>1;
pwrite[-1].right = (pread[0].right + last.right)>>1;
last = pread[0];
pwrite[-2] = last;
pwrite[-3].left = (pread[-1].left + last.left)>>1;
pwrite[-3].right = (pread[-1].right + last.right)>>1;
last = pread[-1];
pwrite[-4] = last;
pwrite[-5].left = (pread[-2].left + last.left)>>1;
pwrite[-5].right = (pread[-2].right + last.right)>>1;
last = pread[-2];
pwrite[-6] = last;
pwrite[-7].left = (pread[-3].left + last.left)>>1;
pwrite[-7].right = (pread[-3].right + last.right)>>1;
last = pread[-3];
pwrite[-8] = last;
pwrite[-9].left = (pread[-4].left + last.left)>>1;
pwrite[-9].right = (pread[-4].right + last.right)>>1;
last = pread[-4];
pwrite[-10] = last;
pwrite[-11].left = (pread[-5].left + last.left)>>1;
pwrite[-11].right = (pread[-5].right + last.right)>>1;
last = pread[-5];
pwrite[-12] = last;
pwrite[-13].left = (pread[-6].left + last.left)>>1;
pwrite[-13].right = (pread[-6].right + last.right)>>1;
last = pread[-6];
pwrite[-14] = last;
pwrite[-15].left = (pread[-7].left + last.left)>>1;
pwrite[-15].right = (pread[-7].right + last.right)>>1;
last = pread[-7];
pread -= 8;
pwrite -= 16;
}
while ( pread >= pbuffer )
{
pwrite[0] = last;
pwrite[-1].left = (pread[0].left + last.left)>>1;
pwrite[-1].right = (pread[0].right + last.right)>>1;
last = pread[0];
pread--;
pwrite-=2;
}
pbuffer[1] = last;
pbuffer[0].left = (pfiltermem->left + last.left) >> 1;
pbuffer[0].right = (pfiltermem->right + last.right) >> 1;
*pfiltermem = pbuffer[end];
}
// upsample by 2x, optionally using interpolation
// count: how many samples to upsample. will become count*2 samples in buffer, in place.
// pbuffer: buffer to upsample into (in place)
// pfiltermem: filter memory. NOTE: this must match the filtertype ie: filterlinear[] for FILTERTYPE_LINEAR
// if NULL then perform no filtering.
// cfltmem: max number of sample pairs filter can use
// filtertype: FILTERTYPE_NONE, _LINEAR, _CUBIC etc. Must match prevfilter.
void S_MixBufferUpsample2x( int count, portable_samplepair_t *pbuffer, portable_samplepair_t *pfiltermem, int cfltmem, int filtertype )
{
// JAY: Optimized this routine. Test then remove old routine.
// NOTE: Has been proven equivalent by comparing output.
if ( filtertype == FILTERTYPE_LINEAR )
{
S_Interpolate2xLinear_2( count, pbuffer, pfiltermem, cfltmem );
return;
}
int i, j, upCount = count<<1;
// reverse through buffer, duplicating contents for 'count' samples
for (i = upCount - 1, j = count - 1; j >= 0; i-=2, j--)
{
pbuffer[i] = pbuffer[j];
pbuffer[i-1] = pbuffer[j];
}
// pass forward through buffer, interpolate all even slots
switch (filtertype)
{
default:
break;
case FILTERTYPE_LINEAR:
S_Interpolate2xLinear(pbuffer, pfiltermem, cfltmem, count);
break;
case FILTERTYPE_CUBIC:
S_Interpolate2xCubic(pbuffer, pfiltermem, cfltmem, count);
break;
}
}
//===============================================================================
// PAINTBUFFER ROUTINES
//===============================================================================
// Set current paintbuffer to pbuf.
// The set paintbuffer is used by all subsequent mixing, upsampling and dsp routines.
// Also sets the rear paintbuffer if paintbuffer has fsurround true.
// (otherwise, rearpaintbuffer is NULL)
void MIX_SetCurrentPaintbuffer(int ipaintbuffer)
{
// set front and rear paintbuffer
Assert(ipaintbuffer < g_paintBuffers.Count());
g_curpaintbuffer = g_paintBuffers[ipaintbuffer].pbuf;
if ( g_paintBuffers[ipaintbuffer].fsurround )
{
g_currearpaintbuffer = g_paintBuffers[ipaintbuffer].pbufrear;
g_curcenterpaintbuffer = NULL;
if ( g_paintBuffers[ipaintbuffer].fsurround_center )
g_curcenterpaintbuffer = g_paintBuffers[ipaintbuffer].pbufcenter;
}
else
{
g_currearpaintbuffer = NULL;
g_curcenterpaintbuffer = NULL;
}
Assert(g_curpaintbuffer != NULL);
}
// return index to current paintbuffer
int MIX_GetCurrentPaintbufferIndex( void )
{
int i;
for ( i = 0; i < g_paintBuffers.Count(); i++ )
{
if (g_curpaintbuffer == g_paintBuffers[i].pbuf)
return i;
}
return 0;
}
// return pointer to current paintbuffer struct
paintbuffer_t *MIX_GetCurrentPaintbufferPtr( void )
{
int ipaint = MIX_GetCurrentPaintbufferIndex();
Assert( ipaint < g_paintBuffers.Count() );
return &g_paintBuffers[ipaint];
}
// return pointer to front paintbuffer pbuf, given index
inline portable_samplepair_t *MIX_GetPFrontFromIPaint(int ipaintbuffer)
{
return g_paintBuffers[ipaintbuffer].pbuf;
}
paintbuffer_t *MIX_GetPPaintFromIPaint( int ipaintbuffer )
{
Assert( ipaintbuffer < g_paintBuffers.Count() );
return &g_paintBuffers[ipaintbuffer];
}
// return pointer to rear buffer, given index.
// returns null if fsurround is false;
inline portable_samplepair_t *MIX_GetPRearFromIPaint(int ipaintbuffer)
{
if ( g_paintBuffers[ipaintbuffer].fsurround )
return g_paintBuffers[ipaintbuffer].pbufrear;
return NULL;
}
// return pointer to center buffer, given index.
// returns null if fsurround_center is false;
inline portable_samplepair_t *MIX_GetPCenterFromIPaint(int ipaintbuffer)
{
if ( g_paintBuffers[ipaintbuffer].fsurround_center )
return g_paintBuffers[ipaintbuffer].pbufcenter;
return NULL;
}
// return index to paintbuffer, given buffer pointer
inline int MIX_GetIPaintFromPFront( portable_samplepair_t *pbuf )
{
int i;
for ( i = 0; i < g_paintBuffers.Count(); i++ )
{
if ( pbuf == g_paintBuffers[i].pbuf )
return i;
}
return 0;
}
// return pointer to paintbuffer struct, given ptr to buffer data
inline paintbuffer_t *MIX_GetPPaintFromPFront( portable_samplepair_t *pbuf )
{
int i;
i = MIX_GetIPaintFromPFront( pbuf );
return &g_paintBuffers[i];
}
// up convert mono buffer to full surround
inline void MIX_ConvertBufferToSurround( int ipaintbuffer )
{
paintbuffer_t *ppaint = &g_paintBuffers[ipaintbuffer];
// duplicate channel data as needed
if ( g_AudioDevice->IsSurround() )
{
// set buffer flags
ppaint->fsurround = g_AudioDevice->IsSurround();
ppaint->fsurround_center = g_AudioDevice->IsSurroundCenter();
portable_samplepair_t *pfront = MIX_GetPFrontFromIPaint( ipaintbuffer );
portable_samplepair_t *prear = MIX_GetPRearFromIPaint( ipaintbuffer );
portable_samplepair_t *pcenter = MIX_GetPCenterFromIPaint( ipaintbuffer );
// copy front to rear
Q_memcpy(prear, pfront, sizeof(portable_samplepair_t) * PAINTBUFFER_SIZE);
// copy front to center
if ( g_AudioDevice->IsSurroundCenter() )
Q_memcpy(pcenter, pfront, sizeof(portable_samplepair_t) * PAINTBUFFER_SIZE);
}
}
// Activate a paintbuffer. All active paintbuffers are mixed in parallel within
// MIX_MixChannelsToPaintbuffer, according to flags
inline void MIX_ActivatePaintbuffer(int ipaintbuffer)
{
Assert( ipaintbuffer < g_paintBuffers.Count() );
g_paintBuffers[ipaintbuffer].factive = true;
}
// Don't mix into this paintbuffer
inline void MIX_DeactivatePaintbuffer(int ipaintbuffer)
{
Assert( ipaintbuffer < g_paintBuffers.Count() );
g_paintBuffers[ipaintbuffer].factive = false;
}
// Don't mix into any paintbuffers
inline void MIX_DeactivateAllPaintbuffers(void)
{
int i;
for ( i = 0; i < g_paintBuffers.Count(); i++ )
g_paintBuffers[i].factive = false;
}
// set upsampling filter indexes back to 0
inline void MIX_ResetPaintbufferFilterCounters( void )
{
int i;
for ( i = 0; i < g_paintBuffers.Count(); i++ )
g_paintBuffers[i].ifilter = 0;
}
inline void MIX_ResetPaintbufferFilterCounter( int ipaintbuffer )
{
Assert ( ipaintbuffer < g_paintBuffers.Count() );
g_paintBuffers[ipaintbuffer].ifilter = 0;
}
// Change paintbuffer's flags
inline void MIX_SetPaintbufferFlags(int ipaintbuffer, int flags)
{
Assert( ipaintbuffer < g_paintBuffers.Count() );
g_paintBuffers[ipaintbuffer].flags = flags;
}
// zero out all paintbuffers
void MIX_ClearAllPaintBuffers( int SampleCount, bool clearFilters )
{
// g_paintBuffers can be NULL with -nosound
if ( g_paintBuffers.Count() <= 0 )
{
return;
}
int i;
int count = min(SampleCount, PAINTBUFFER_SIZE);
// zero out all paintbuffer data (ignore sampleCount)
for ( i = 0; i < g_paintBuffers.Count(); i++ )
{
if (g_paintBuffers[i].pbuf != NULL)
Q_memset(g_paintBuffers[i].pbuf, 0, (count+1) * sizeof(portable_samplepair_t));
if (g_paintBuffers[i].pbufrear != NULL)
Q_memset(g_paintBuffers[i].pbufrear, 0, (count+1) * sizeof(portable_samplepair_t));
if (g_paintBuffers[i].pbufcenter != NULL)
Q_memset(g_paintBuffers[i].pbufcenter, 0, (count+1) * sizeof(portable_samplepair_t));
if ( clearFilters )
{
Q_memset( g_paintBuffers[i].fltmem, 0, sizeof(g_paintBuffers[i].fltmem) );
Q_memset( g_paintBuffers[i].fltmemrear, 0, sizeof(g_paintBuffers[i].fltmemrear) );
Q_memset( g_paintBuffers[i].fltmemcenter, 0, sizeof(g_paintBuffers[i].fltmemcenter) );
}
}
if ( clearFilters )
{
MIX_ResetPaintbufferFilterCounters();
}
}
#define SWAP(a,b,t) {(t) = (a); (a) = (b); (b) = (t);}
#define AVG(a,b) (((a) + (b)) >> 1 )
#define AVG4(a,b,c,d) (((a) + (b) + (c) + (d)) >> 2 )
// Synthesize center channel from left/right values (average).
// Currently just averages, but could actually remove
// the center signal from the l/r channels...
inline void MIX_CenterFromLeftRight( int *pl, int *pr, int *pc )
{
int l = *pl;
int r = *pr;
int c = 0;
c = (l + r) / 2;
/*
l = l - c/2;
r = r - c/2;
if (l < 0)
{
l = 0;
r += (-l);
c += (-l);
}
else if (r < 0)
{
r = 0;
l += (-r);
c += (-r);
}
*/
*pc = c;
// *pl = l;
// *pr = r;
}
// mixes pbuf1 + pbuf2 into pbuf3, count samples
// fgain is output gain 0-1.0
// NOTE: pbuf3 may equal pbuf1 or pbuf2!
// mixing algorithms:
// destination 2ch:
// pb1 2ch + pb2 2ch -> pb3 2ch
// pb1 (4ch->2ch) + pb2 2ch -> pb3 2ch
// pb1 2ch + pb2 (4ch->2ch) -> pb3 2ch
// pb1 (4ch->2ch) + pb2 (4ch->2ch) -> pb3 2ch
// destination 4ch:
// pb1 4ch + pb2 4ch -> pb3 4ch
// pb1 (2ch->4ch) + pb2 4ch -> pb3 4ch
// pb1 4ch + pb2 (2ch->4ch) -> pb3 4ch
// pb1 (2ch->4ch) + pb2 (2ch->4ch) -> pb3 4ch
// if all buffers are 4 or 5 ch surround, mix rear & center channels into ibuf3 as well.
// NOTE: for performance, conversion and mixing are done in a single pass instead of
// a two pass channel convert + mix scheme.
void MIX_MixPaintbuffers(int ibuf1, int ibuf2, int ibuf3, int count, float fgain_out)
{
VPROF("Mixpaintbuffers");
int i;
portable_samplepair_t *pbuf1, *pbuf2, *pbuf3, *pbuft;
portable_samplepair_t *pbufrear1, *pbufrear2, *pbufrear3, *pbufreart;
portable_samplepair_t *pbufcenter1, *pbufcenter2, *pbufcenter3, *pbufcentert;
int cchan1, cchan2, cchan3, cchant;
int xl,xr;
int l,r,l2,r2,c, c2;
int gain_out;
gain_out = 256 * fgain_out;
Assert (count <= PAINTBUFFER_SIZE);
Assert (ibuf1 < g_paintBuffers.Count());
Assert (ibuf2 < g_paintBuffers.Count());
Assert (ibuf3 < g_paintBuffers.Count());
pbuf1 = g_paintBuffers[ibuf1].pbuf;
pbuf2 = g_paintBuffers[ibuf2].pbuf;
pbuf3 = g_paintBuffers[ibuf3].pbuf;
pbufrear1 = g_paintBuffers[ibuf1].pbufrear;
pbufrear2 = g_paintBuffers[ibuf2].pbufrear;
pbufrear3 = g_paintBuffers[ibuf3].pbufrear;
pbufcenter1 = g_paintBuffers[ibuf1].pbufcenter;
pbufcenter2 = g_paintBuffers[ibuf2].pbufcenter;
pbufcenter3 = g_paintBuffers[ibuf3].pbufcenter;
cchan1 = 2 + (g_paintBuffers[ibuf1].fsurround ? 2 : 0) + (g_paintBuffers[ibuf1].fsurround_center ? 1 : 0);
cchan2 = 2 + (g_paintBuffers[ibuf2].fsurround ? 2 : 0) + (g_paintBuffers[ibuf2].fsurround_center ? 1 : 0);
cchan3 = 2 + (g_paintBuffers[ibuf3].fsurround ? 2 : 0) + (g_paintBuffers[ibuf3].fsurround_center ? 1 : 0);
// make sure pbuf1 always has fewer or equal channels than pbuf2
// NOTE: pbuf3 may equal pbuf1 or pbuf2!
if ( cchan2 < cchan1 )
{
SWAP( cchan1, cchan2, cchant );
SWAP( pbuf1, pbuf2, pbuft );
SWAP( pbufrear1, pbufrear2, pbufreart );
SWAP( pbufcenter1, pbufcenter2, pbufcentert);
}
// UNDONE: implement fast mixing routines for each of the following sections
// destination buffer stereo - average n chans down to stereo
if ( cchan3 == 2 )
{
// destination 2ch:
// pb1 2ch + pb2 2ch -> pb3 2ch
// pb1 2ch + pb2 (4ch->2ch) -> pb3 2ch
// pb1 (4ch->2ch) + pb2 (4ch->2ch) -> pb3 2ch
if ( cchan1 == 2 && cchan2 == 2 )
{
// mix front channels
for (i = 0; i < count; i++)
{
pbuf3[i].left = pbuf1[i].left + pbuf2[i].left;
pbuf3[i].right = pbuf1[i].right + pbuf2[i].right;
}
goto gain2ch;
}
if ( cchan1 == 2 && cchan2 == 4 )
{
// avg rear chan l/r
for (i = 0; i < count; i++)
{
pbuf3[i].left = pbuf1[i].left + AVG( pbuf2[i].left, pbufrear2[i].left );
pbuf3[i].right = pbuf1[i].right + AVG( pbuf2[i].right, pbufrear2[i].right );
}
goto gain2ch;
}
if ( cchan1 == 4 && cchan2 == 4 )
{
// avg rear chan l/r
for (i = 0; i < count; i++)
{
pbuf3[i].left = AVG( pbuf1[i].left, pbufrear1[i].left) + AVG( pbuf2[i].left, pbufrear2[i].left );
pbuf3[i].right = AVG( pbuf1[i].right, pbufrear1[i].right) + AVG( pbuf2[i].right, pbufrear2[i].right );
}
goto gain2ch;
}
if ( cchan1 == 2 && cchan2 == 5 )
{
// avg rear chan l/r + center split into left/right
for (i = 0; i < count; i++)
{
l = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
pbuf3[i].left = pbuf1[i].left + AVG( l, pbufrear2[i].left );
pbuf3[i].right = pbuf1[i].right + AVG( r, pbufrear2[i].right );
}
goto gain2ch;
}
if ( cchan1 == 4 && cchan2 == 5)
{
for (i = 0; i < count; i++)
{
l = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
pbuf3[i].left = AVG( pbuf1[i].left, pbufrear1[i].left) + AVG( l, pbufrear2[i].left );
pbuf3[i].right = AVG( pbuf1[i].right, pbufrear1[i].right) + AVG( r, pbufrear2[i].right );
}
goto gain2ch;
}
if ( cchan1 == 5 && cchan2 == 5)
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left + ((pbufcenter1[i].left) >> 1);
r = pbuf1[i].right + ((pbufcenter1[i].left) >> 1);
l2 = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r2 = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
pbuf3[i].left = AVG( l, pbufrear1[i].left) + AVG( l2, pbufrear2[i].left );
pbuf3[i].right = AVG( r, pbufrear1[i].right) + AVG( r2, pbufrear2[i].right );
} goto gain2ch;
}
}
// destination buffer quad - duplicate n chans up to quad
if ( cchan3 == 4 )
{
// pb1 4ch + pb2 4ch -> pb3 4ch
// pb1 (2ch->4ch) + pb2 4ch -> pb3 4ch
// pb1 (2ch->4ch) + pb2 (2ch->4ch) -> pb3 4ch
if ( cchan1 == 4 && cchan2 == 4)
{
// mix front -> front, rear -> rear
for (i = 0; i < count; i++)
{
pbuf3[i].left = pbuf1[i].left + pbuf2[i].left;
pbuf3[i].right = pbuf1[i].right + pbuf2[i].right;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
}
goto gain4ch;
}
if ( cchan1 == 2 && cchan2 == 4)
{
for (i = 0; i < count; i++)
{
// split 2 ch left -> front left, rear left
// split 2 ch right -> front right, rear right
xl = pbuf1[i].left;
xr = pbuf1[i].right;
pbuf3[i].left = xl + pbuf2[i].left;
pbuf3[i].right = xr + pbuf2[i].right;
pbufrear3[i].left = xl + pbufrear2[i].left;
pbufrear3[i].right = xr + pbufrear2[i].right;
}
goto gain4ch;
}
if ( cchan1 == 2 && cchan2 == 2)
{
// mix l,r, split into front l, front r
for (i = 0; i < count; i++)
{
xl = pbuf1[i].left + pbuf2[i].left;
xr = pbuf1[i].right + pbuf2[i].right;
pbufrear3[i].left = pbuf3[i].left = xl;
pbufrear3[i].right = pbuf3[i].right = xr;
}
goto gain4ch;
}
if ( cchan1 == 2 && cchan2 == 5 )
{
for (i = 0; i < count; i++)
{
// split center of chan2 into left/right
l2 = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r2 = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
xl = pbuf1[i].left;
xr = pbuf1[i].right;
pbuf3[i].left = xl + l2;
pbuf3[i].right = xr + r2;
pbufrear3[i].left = xl + pbufrear2[i].left;
pbufrear3[i].right = xr + pbufrear2[i].right;
}
goto gain4ch;
}
if ( cchan1 == 4 && cchan2 == 5)
{
for (i = 0; i < count; i++)
{
l2 = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r2 = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
pbuf3[i].left = pbuf1[i].left + l2;
pbuf3[i].right = pbuf1[i].right + r2;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
}
goto gain4ch;
}
if ( cchan1 == 5 && cchan2 == 5 )
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left + ((pbufcenter1[i].left) >> 1);
r = pbuf1[i].right + ((pbufcenter1[i].left) >> 1);
l2 = pbuf2[i].left + ((pbufcenter2[i].left) >> 1);
r2 = pbuf2[i].right + ((pbufcenter2[i].left) >> 1);
pbuf3[i].left = l + l2;
pbuf3[i].right = r + r2;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
}
goto gain4ch;
}
}
// 5 channel destination
if (cchan3 == 5)
{
// up convert from 2 or 4 ch buffer to 5 ch buffer:
// center channel is synthesized from front left, front right
if (cchan1 == 2 && cchan2 == 2)
{
for (i = 0; i < count; i++)
{
// split 2 ch left -> front left, center, rear left
// split 2 ch right -> front right, center, rear right
l = pbuf1[i].left;
r = pbuf1[i].right;
MIX_CenterFromLeftRight(&l, &r, &c);
l2 = pbuf2[i].left;
r2 = pbuf2[i].right;
MIX_CenterFromLeftRight(&l2, &r2, &c2);
pbuf3[i].left = l + l2;
pbuf3[i].right = r + r2;
pbufrear3[i].left = pbuf1[i].left + pbuf2[i].left;
pbufrear3[i].right = pbuf1[i].right + pbuf2[i].right;
pbufcenter3[i].left = c + c2;
}
goto gain5ch;
}
if (cchan1 == 2 && cchan2 == 4)
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left;
r = pbuf1[i].right;
MIX_CenterFromLeftRight(&l, &r, &c);
l2 = pbuf2[i].left;
r2 = pbuf2[i].right;
MIX_CenterFromLeftRight(&l2, &r2, &c2);
pbuf3[i].left = l + l2;
pbuf3[i].right = r + r2;
pbufrear3[i].left = pbuf1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbuf1[i].right + pbufrear2[i].right;
pbufcenter3[i].left = c + c2;
}
goto gain5ch;
}
if (cchan1 == 2 && cchan2 == 5)
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left;
r = pbuf1[i].right;
MIX_CenterFromLeftRight(&l, &r, &c);
pbuf3[i].left = l + pbuf2[i].left;
pbuf3[i].right = r + pbuf2[i].right;
pbufrear3[i].left = pbuf1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbuf1[i].right + pbufrear2[i].right;
pbufcenter3[i].left = c + pbufcenter2[i].left;
}
goto gain5ch;
}
if (cchan1 == 4 && cchan2 == 4)
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left;
r = pbuf1[i].right;
MIX_CenterFromLeftRight(&l, &r, &c);
l2 = pbuf2[i].left;
r2 = pbuf2[i].right;
MIX_CenterFromLeftRight(&l2, &r2, &c2);
pbuf3[i].left = l + l2;
pbuf3[i].right = r + r2;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
pbufcenter3[i].left = c + c2;
}
goto gain5ch;
}
if (cchan1 == 4 && cchan2 == 5)
{
for (i = 0; i < count; i++)
{
l = pbuf1[i].left;
r = pbuf1[i].right;
MIX_CenterFromLeftRight(&l, &r, &c);
pbuf3[i].left = l + pbuf2[i].left;
pbuf3[i].right = r + pbuf2[i].right;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
pbufcenter3[i].left = c + pbufcenter2[i].left;
}
goto gain5ch;
}
if ( cchan2 == 5 && cchan1 == 5 )
{
for (i = 0; i < count; i++)
{
pbuf3[i].left = pbuf1[i].left + pbuf2[i].left;
pbuf3[i].right = pbuf1[i].right + pbuf2[i].right;
pbufrear3[i].left = pbufrear1[i].left + pbufrear2[i].left;
pbufrear3[i].right = pbufrear1[i].right + pbufrear2[i].right;
pbufcenter3[i].left = pbufcenter1[i].left + pbufcenter2[i].left;
}
goto gain5ch;
}
}
gain2ch:
if ( gain_out == 256) // KDB: perf
return;
for (i = 0; i < count; i++)
{
pbuf3[i].left = (pbuf3[i].left * gain_out) >> 8;
pbuf3[i].right = (pbuf3[i].right * gain_out) >> 8;
}
return;
gain4ch:
if ( gain_out == 256) // KDB: perf
return;
for (i = 0; i < count; i++)
{
pbuf3[i].left = (pbuf3[i].left * gain_out) >> 8;
pbuf3[i].right = (pbuf3[i].right * gain_out) >> 8;
pbufrear3[i].left = (pbufrear3[i].left * gain_out) >> 8;
pbufrear3[i].right = (pbufrear3[i].right * gain_out) >> 8;
}
return;
gain5ch:
if ( gain_out == 256) // KDB: perf
return;
for (i = 0; i < count; i++)
{
pbuf3[i].left = (pbuf3[i].left * gain_out) >> 8;
pbuf3[i].right = (pbuf3[i].right * gain_out) >> 8;
pbufrear3[i].left = (pbufrear3[i].left * gain_out) >> 8;
pbufrear3[i].right = (pbufrear3[i].right * gain_out) >> 8;
pbufcenter3[i].left = (pbufcenter3[i].left * gain_out) >> 8;
}
return;
}
// multiply all values in paintbuffer by fgain
void MIX_ScalePaintBuffer( int bufferIndex, int count, float fgain )
{
portable_samplepair_t *pbuf = g_paintBuffers[bufferIndex].pbuf;
portable_samplepair_t *pbufrear = g_paintBuffers[bufferIndex].pbufrear;
portable_samplepair_t *pbufcenter = g_paintBuffers[bufferIndex].pbufcenter;
int gain = 256 * fgain;
int i;
if (gain == 256)
return;
if ( !g_paintBuffers[bufferIndex].fsurround )
{
for (i = 0; i < count; i++)
{
pbuf[i].left = (pbuf[i].left * gain) >> 8;
pbuf[i].right = (pbuf[i].right * gain) >> 8;
}
}
else
{
for (i = 0; i < count; i++)
{
pbuf[i].left = (pbuf[i].left * gain) >> 8;
pbuf[i].right = (pbuf[i].right * gain) >> 8;
pbufrear[i].left = (pbufrear[i].left * gain) >> 8;
pbufrear[i].right = (pbufrear[i].right * gain) >> 8;
}
if (g_paintBuffers[bufferIndex].fsurround_center)
{
for (i = 0; i < count; i++)
{
pbufcenter[i].left = (pbufcenter[i].left * gain) >> 8;
// pbufcenter[i].right = (pbufcenter[i].right * gain) >> 8; mono center channel
}
}
}
}
// DEBUG peak detection values
#define _SDEBUG 1
#ifdef _SDEBUG
float sdebug_avg_in = 0.0;
float sdebug_in_count = 0.0;
float sdebug_avg_out = 0.0;
float sdebug_out_count = 0.0;
#define SDEBUG_TOTAL_COUNT (3*44100)
#endif // DEBUG
// DEBUG code - get and show peak value of specified paintbuffer
// DEBUG code - ibuf is buffer index, count is # samples to test, pppeakprev stores peak
void SDEBUG_GetAvgValue( int ibuf, int count, float *pav )
{
#ifdef _SDEBUG
if (snd_showstart.GetInt() != 4 )
return;
float av = 0.0;
for (int i = 0; i < count; i++)
av += (float)(abs(g_paintBuffers[ibuf].pbuf->left) + abs(g_paintBuffers[ibuf].pbuf->right))/2.0;
*pav = av / count;
#endif // DEBUG
}
void SDEBUG_GetAvgIn( int ibuf, int count)
{
float av = 0.0;
SDEBUG_GetAvgValue( ibuf, count, &av );
sdebug_avg_in = ((av * count ) + (sdebug_avg_in * sdebug_in_count)) / (count + sdebug_in_count);
sdebug_in_count += count;
}
void SDEBUG_GetAvgOut( int ibuf, int count)
{
float av = 0.0;
SDEBUG_GetAvgValue( ibuf, count, &av );
sdebug_avg_out = ((av * count ) + (sdebug_avg_out * sdebug_out_count)) / (count + sdebug_out_count);
sdebug_out_count += count;
}
void SDEBUG_ShowAvgValue()
{
#ifdef _SDEBUG
if (sdebug_in_count > SDEBUG_TOTAL_COUNT)
{
if ((int)sdebug_avg_in > 20.0 && (int)sdebug_avg_out > 20.0)
DevMsg("dsp avg gain:%1.2f in:%1.2f out:%1.2f 1/gain:%1.2f\n", sdebug_avg_out/sdebug_avg_in, sdebug_avg_in, sdebug_avg_out, sdebug_avg_in/sdebug_avg_out);
sdebug_avg_in = 0.0;
sdebug_avg_out = 0.0;
sdebug_in_count = 0.0;
sdebug_out_count = 0.0;
}
#endif // DEBUG
}
// clip all values in paintbuffer to 16bit.
// if fsurround is set for paintbuffer, also process rear buffer samples
void MIX_CompressPaintbuffer(int ipaint, int count)
{
VPROF("CompressPaintbuffer");
int i;
paintbuffer_t *ppaint = MIX_GetPPaintFromIPaint(ipaint);
portable_samplepair_t *pbf;
portable_samplepair_t *pbr;
portable_samplepair_t *pbc;
pbf = ppaint->pbuf;
pbr = ppaint->pbufrear;
pbc = ppaint->pbufcenter;
for (i = 0; i < count; i++)
{
pbf->left = CLIP(pbf->left);
pbf->right = CLIP(pbf->right);
pbf++;
}
if ( ppaint->fsurround )
{
Assert (pbr);
for (i = 0; i < count; i++)
{
pbr->left = CLIP(pbr->left);
pbr->right = CLIP(pbr->right);
pbr++;
}
}
if ( ppaint->fsurround_center )
{
Assert (pbc);
for (i = 0; i < count; i++)
{
pbc->left = CLIP(pbc->left);
//pbc->right = CLIP(pbc->right); mono center channel
pbc++;
}
}
}
// mix and upsample channels to 44khz 'ipaintbuffer'
// mix channels matching 'flags' (SOUND_MIX_DRY, SOUND_MIX_WET, SOUND_MIX_SPEAKER) into specified paintbuffer
// upsamples 11khz, 22khz channels to 44khz.
// NOTE: only call this on channels that will be mixed into only 1 paintbuffer
// and that will not be mixed until the next mix pass! otherwise, MIX_MixChannelsToPaintbuffer
// will advance any internal pointers on mixed channels; subsequent calls will be at
// incorrect offset.
void MIX_MixUpsampleBuffer( CChannelList &list, int ipaintbuffer, int end, int count, int flags )
{
VPROF("MixUpsampleBuffer");
int ipaintcur = MIX_GetCurrentPaintbufferIndex(); // save current paintbuffer
// reset paintbuffer upsampling filter index
MIX_ResetPaintbufferFilterCounter( ipaintbuffer );
// prevent other paintbuffers from being mixed
MIX_DeactivateAllPaintbuffers();
MIX_ActivatePaintbuffer( ipaintbuffer ); // operates on MIX_MixChannelsToPaintbuffer
MIX_SetCurrentPaintbuffer( ipaintbuffer ); // operates on MixUpSample
// mix 11khz channels to buffer
if ( list.m_has11kChannels )
{
MIX_MixChannelsToPaintbuffer( list, end, flags, SOUND_11k, SOUND_11k );
// upsample 11khz buffer by 2x
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_11k), FILTERTYPE_LINEAR );
}
if ( list.m_has22kChannels || list.m_has11kChannels )
{
// mix 22khz channels to buffer
MIX_MixChannelsToPaintbuffer( list, end, flags, SOUND_22k, SOUND_22k );
#if (SOUND_DMA_SPEED > SOUND_22k)
// upsample 22khz buffer by 2x
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_22k), FILTERTYPE_LINEAR );
#endif
}
// mix 44khz channels to buffer
MIX_MixChannelsToPaintbuffer( list, end, flags, SOUND_44k, SOUND_DMA_SPEED);
MIX_DeactivateAllPaintbuffers();
// restore previous paintbuffer
MIX_SetCurrentPaintbuffer( ipaintcur );
}
// upsample and mix sounds into final 44khz versions of the following paintbuffers:
// SOUND_BUFFER_ROOM, SOUND_BUFFER_FACING, IFACINGAWAY, SOUND_BUFFER_DRY, SOUND_BUFFER_SPEAKER, SOUND_BUFFER_SPECIALs
// dsp fx are then applied to these buffers by the caller.
// caller also remixes all into final SOUND_BUFFER_PAINT output.
void MIX_UpsampleAllPaintbuffers( CChannelList &list, int end, int count )
{
VPROF( "MixUpsampleAll" );
// 'dry' and 'speaker' channel sounds mix 100% into their corresponding buffers
// mix and upsample all 'dry' sounds (channels) to 44khz SOUND_BUFFER_DRY paintbuffer
if ( list.m_hasDryChannels )
MIX_MixUpsampleBuffer( list, SOUND_BUFFER_DRY, end, count, SOUND_MIX_DRY );
// mix and upsample all 'speaker' sounds (channels) to 44khz SOUND_BUFFER_SPEAKER paintbuffer
if ( list.m_hasSpeakerChannels )
MIX_MixUpsampleBuffer( list, SOUND_BUFFER_SPEAKER, end, count, SOUND_MIX_SPEAKER );
// mix and upsample all 'special dsp' sounds (channels) to 44khz SOUND_BUFFER_SPECIALs paintbuffer
for ( int iDSP = 0; iDSP < list.m_nSpecialDSPs.Count(); ++iDSP )
{
for ( int i = SOUND_BUFFER_SPECIAL_START; i < g_paintBuffers.Count(); ++i )
{
paintbuffer_t *pSpecialBuffer = MIX_GetPPaintFromIPaint( i );
if ( pSpecialBuffer->nSpecialDSP == list.m_nSpecialDSPs[ iDSP ] && pSpecialBuffer->idsp_specialdsp != -1 )
{
MIX_MixUpsampleBuffer( list, i, end, count, SOUND_MIX_SPECIAL_DSP );
break;
}
}
}
// 'room', 'facing' 'facingaway' sounds are mixed into up to 3 buffers:
// 11khz sounds are mixed into 3 buffers based on distance from listener, and facing direction
// These buffers are room, facing, facingaway
// These 3 mixed buffers are then each upsampled to 22khz.
// 22khz sounds are mixed into the 3 buffers based on distance from listener, and facing direction
// These 3 mixed buffers are then each upsampled to 44khz.
// 44khz sounds are mixed into the 3 buffers based on distance from listener, and facing direction
MIX_DeactivateAllPaintbuffers();
// set paintbuffer upsample filter indices to 0
MIX_ResetPaintbufferFilterCounters();
if ( !g_bDspOff )
{
// only mix to roombuffer if dsp fx are on KDB: perf
MIX_ActivatePaintbuffer(SOUND_BUFFER_ROOM); // operates on MIX_MixChannelsToPaintbuffer
}
MIX_ActivatePaintbuffer(SOUND_BUFFER_FACING);
if ( g_bdirectionalfx )
{
// mix to facing away buffer only if directional presets are set
MIX_ActivatePaintbuffer(SOUND_BUFFER_FACINGAWAY);
}
// mix 11khz sounds:
// pan sounds between 3 busses: facing, facingaway and room buffers
MIX_MixChannelsToPaintbuffer( list, end, SOUND_MIX_WET, SOUND_11k, SOUND_11k);
// upsample all 11khz buffers by 2x
if ( !g_bDspOff )
{
// only upsample roombuffer if dsp fx are on KDB: perf
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_ROOM); // operates on MixUpSample
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_11k), FILTERTYPE_LINEAR );
}
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_FACING);
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_11k), FILTERTYPE_LINEAR );
if ( g_bdirectionalfx )
{
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_FACINGAWAY);
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_11k), FILTERTYPE_LINEAR );
}
// mix 22khz sounds:
// pan sounds between 3 busses: facing, facingaway and room buffers
MIX_MixChannelsToPaintbuffer( list, end, SOUND_MIX_WET, SOUND_22k, SOUND_22k);
// upsample all 22khz buffers by 2x
#if ( SOUND_DMA_SPEED > SOUND_22k )
if ( !g_bDspOff )
{
// only upsample roombuffer if dsp fx are on KDB: perf
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_ROOM);
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_22k), FILTERTYPE_LINEAR );
}
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_FACING);
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_22k), FILTERTYPE_LINEAR );
if ( g_bdirectionalfx )
{
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_FACINGAWAY);
g_AudioDevice->MixUpsample( count / (SOUND_DMA_SPEED / SOUND_22k), FILTERTYPE_LINEAR );
}
#endif
// mix all 44khz sounds to all active paintbuffers
MIX_MixChannelsToPaintbuffer( list, end, SOUND_MIX_WET, SOUND_44k, SOUND_DMA_SPEED);
MIX_DeactivateAllPaintbuffers();
MIX_SetCurrentPaintbuffer(SOUND_BUFFER_PAINT);
}
ConVar snd_cull_duplicates("snd_cull_duplicates","0",FCVAR_ALLOWED_IN_COMPETITIVE,"If nonzero, aggressively cull duplicate sounds during mixing. The number specifies the number of duplicates allowed to be played.");
// Helper class for determining whether a given channel number should be culled from
// mixing, if snd_cull_duplicates is enabled (psychoacoustic quashing).
class CChannelCullList
{
public:
// default constructor
CChannelCullList() : m_numChans(0) {};
// call if you plan on culling channels - and not otherwise, it's a little expensive
// (that's why it's not in the constructor)
void Initialize( CChannelList &list );
// returns true if a given channel number has been marked for culling
inline bool ShouldCull( int channelNum )
{
return (m_numChans > channelNum) ? m_bShouldCull[channelNum] : false;
}
// an array of sound names and their volumes
// TODO: there may be a way to do this faster on 360 (eg, pad to 128bit, use SIMD)
struct sChannelVolData
{
int m_channelNum;
int m_vol; // max volume of sound. -1 means "do not cull, ever, do not even do the math"
uintp m_nameHash; // a unique id for a sound file
};
protected:
sChannelVolData m_channelInfo[MAX_CHANNELS];
bool m_bShouldCull[MAX_CHANNELS]; // in ChannelList order, not sorted order
int m_numChans;
};
// comparator for qsort as used below (eg a lambda)
// returns < 0 if a should come before b, > 0 if a should come after, 0 otherwise
static int __cdecl ChannelVolComparator ( const void * a, const void * b )
{
// greater numbers come first.
return static_cast<const CChannelCullList::sChannelVolData *>(b)->m_vol - static_cast<const CChannelCullList::sChannelVolData *>(a)->m_vol;
}
void CChannelCullList::Initialize( CChannelList &list )
{
VPROF("CChannelCullList::Initialize");
// First, build a sorted list of channels by decreasing volume, and by a hash of their wavname.
m_numChans = list.Count();
for ( int i = m_numChans - 1 ; i >= 0 ; --i )
{
channel_t *ch = list.GetChannel(i);
m_channelInfo[i].m_channelNum = i;
if ( ch && ch->pMixer->IsReadyToMix() )
{
m_channelInfo[i].m_vol = ChannelLoudestCurVolume(ch);
AssertMsg(m_channelInfo[i].m_vol >= 0, "Sound channel has a negative volume?");
m_channelInfo[i].m_nameHash = (uintp) ch->sfx;
}
else
{
m_channelInfo[i].m_vol = -1;
m_channelInfo[i].m_nameHash = NULL; // doesn't matter
}
}
// set the unused channels to invalid data
for ( int i = m_numChans ; i < MAX_CHANNELS ; ++i )
{
m_channelInfo[i].m_channelNum = -1;
m_channelInfo[i].m_vol = -1;
}
// Sort the list.
qsort( m_channelInfo, MAX_CHANNELS, sizeof(sChannelVolData), ChannelVolComparator );
// Then, determine if the given sound is less than the nth loudest of its hash. If so, mark its flag
// for removal.
// TODO: use an actual algorithm rather than this bogus quadratic technique.
// (I'm using it for now because we don't have convenient/fast hash table
// classes, which would be the linear-time way to deal with this).
const int cutoff = snd_cull_duplicates.GetInt();
for ( int i = 0 ; i < m_numChans ; ++i ) // i is index in original channel list
{
channel_t *ch = list.GetChannel(i);
// for each sound, determine where it ranks in loudness
int howManyLouder = 0;
for ( int j = 0 ;
m_channelInfo[j].m_channelNum != i && m_channelInfo[j].m_vol >= 0 && j < MAX_CHANNELS ;
++j )
{
// j steps through the sorted list until we find ourselves:
if (m_channelInfo[j].m_nameHash == (uintp)(ch->sfx))
{
// that's another channel playing this sound but louder than me
++howManyLouder;
}
}
if (howManyLouder >= cutoff)
{
// this sound should be culled
m_bShouldCull[i] = true;
}
else
{
// this sound should not be culled
m_bShouldCull[i] = false;
}
}
}
ConVar snd_mute_losefocus("snd_mute_losefocus", "1", FCVAR_ARCHIVE);
// build a list of channels that will actually do mixing in this update
// remove all active channels that won't mix for some reason
void MIX_BuildChannelList( CChannelList &list )
{
VPROF("MIX_BuildChannelList");
g_ActiveChannels.GetActiveChannels( list );
list.m_nSpecialDSPs.RemoveAll();
list.m_hasDryChannels = false;
list.m_hasSpeakerChannels = false;
list.m_has11kChannels = false;
list.m_has22kChannels = false;
list.m_has44kChannels = false;
bool delayStartServer = false;
bool delayStartClient = false;
bool bPaused = g_pSoundServices->IsGamePaused();
#ifdef POSIX
bool bActive = g_pSoundServices->IsGameActive();
bool bStopOnFocusLoss = !bActive && snd_mute_losefocus.GetBool();
#endif
CChannelCullList cullList;
if (snd_cull_duplicates.GetInt() > 0)
{
cullList.Initialize(list);
}
// int numQuashed = 0;
for ( int i = list.Count(); --i >= 0; )
{
channel_t *ch = list.GetChannel(i);
bool bRemove = false;
// Certain async loaded sounds lazily load into memory in the background, use this to determine
// if the sound is ready for mixing
CAudioSource *pSource = NULL;
if ( ch->pMixer->IsReadyToMix() )
{
pSource = S_LoadSound( ch->sfx, ch );
// Don't mix sound data for sounds with 'zero' volume. If it's a non-looping sound,
// just remove the sound when its volume goes to zero. If it's a 'dry' channel sound (ie: music)
// then assume bZeroVolume is fade in - don't restart
// To be 'zero' volume, all target volume and current volume values must all be less than 5
bool bZeroVolume = BChannelLowVolume( ch, 1 );
if ( !pSource || ( bZeroVolume && !pSource->IsLooped() && !ch->flags.bdry ) )
{
// NOTE: Since we've loaded the sound, check to see if it's a sentence. Play them at zero anyway
// to keep the character's lips moving and the captions happening.
if ( !pSource || pSource->GetSentence() == NULL )
{
S_FreeChannel( ch );
bRemove = true;
}
}
else if ( bZeroVolume )
{
bRemove = true;
}
// If the sound wants to stop when the game pauses, do so
if ( bPaused && SND_ShouldPause(ch) )
{
bRemove = true;
}
#ifdef POSIX
// If we aren't the active app and the option for background audio isn't on, mute the audio
// Windows has it's own system for background muting
if ( !bRemove && bStopOnFocusLoss )
{
bRemove = true;
// Free up the sound channels otherwise they start filling up
if ( pSource && ( !pSource->IsLooped() && !pSource->IsStreaming() ) )
{
S_FreeChannel( ch );
}
}
#endif
// On lowend, aggressively cull duplicate sounds.
if ( !bRemove && snd_cull_duplicates.GetInt() > 0 )
{
// We can't simply remove them, because then sounds will pile up waiting to finish later.
// We need to flag them for not mixing.
list.m_quashed[i] = cullList.ShouldCull(i);
/*
if (list.m_quashed[i])
{
numQuashed++;
// Msg("removed %i\n", i);
}
*/
}
else
{
list.m_quashed[i] = false;
}
}
else
{
bRemove = true;
}
if ( bRemove )
{
list.RemoveChannelFromList(i);
continue;
}
if ( ch->flags.bSpeaker )
{
list.m_hasSpeakerChannels = true;
}
if ( ch->special_dsp != 0 )
{
if ( list.m_nSpecialDSPs.Find( ch->special_dsp ) == -1 )
{
list.m_nSpecialDSPs.AddToTail( ch->special_dsp );
}
}
if ( ch->flags.bdry )
{
list.m_hasDryChannels = true;
}
int rate = pSource->SampleRate();
if ( rate == SOUND_11k )
{
list.m_has11kChannels = true;
}
else if ( rate == SOUND_22k )
{
list.m_has22kChannels = true;
}
else if ( rate == SOUND_44k )
{
list.m_has44kChannels = true;
}
if ( ch->flags.delayed_start && !SND_IsMouth(ch) )
{
if ( ch->flags.fromserver )
{
delayStartServer = true;
}
else
{
delayStartClient = true;
}
}
// get playback pitch
ch->pitch = ch->pMixer->ModifyPitch( ch->basePitch * 0.01f );
}
// DevMsg( "%d channels quashed.\n", numQuashed );
// This code will resync the delay calculation clock really often
// any time there are no scheduled waves or the game is paused
// we go ahead and reset the clock
// That way the clock is only used for short periods of time
// and we need no solution for drift
if ( bPaused || (host_frametime_unbounded > host_frametime) )
{
delayStartClient = false;
delayStartServer = false;
}
if (!delayStartServer)
{
S_SyncClockAdjust(CLOCK_SYNC_SERVER);
}
if (!delayStartClient)
{
S_SyncClockAdjust(CLOCK_SYNC_CLIENT);
}
}
// main mixing rountine - mix up to 'endtime' samples.
// All channels are mixed in a paintbuffer and then sent to
// hardware.
// A mix pass is performed, resulting in mixed sounds in SOUND_BUFFER_ROOM, SOUND_BUFFER_FACING, SOUND_BUFFER_FACINGAWAY, SOUND_BUFFER_DRY, SOUND_BUFFER_SPEAKER, SOUND_BUFFER_SPECIALs
// directional sounds are panned and mixed between SOUND_BUFFER_FACING and SOUND_BUFFER_FACINGAWAY
// omnidirectional sounds are panned 100% into SOUND_BUFFER_FACING
// sound sources far from player (ie: near back of room ) are mixed in proportion to this distance
// into SOUND_BUFFER_ROOM
// sounds with ch->bSpeaker set are mixed in mono into SOUND_BUFFER_SPEAKER
// sounds with ch->bSpecialDSP set are mixed in mono into SOUND_BUFFER_SPECIALs
// dsp_facingaway fx (2 or 4ch filtering) are then applied to the SOUND_BUFFER_FACINGAWAY
// dsp_speaker fx (1ch) are then applied to the SOUND_BUFFER_SPEAKER
// dsp_specialdsp fx (1ch) are then applied to the SOUND_BUFFER_SPECIALs
// dsp_room fx (1ch reverb) are then applied to the SOUND_BUFFER_ROOM
// All buffers are recombined into the SOUND_BUFFER_PAINT
// The dsp_water and dsp_player fx are applied in series to the SOUND_BUFFER_PAINT
// Finally, the SOUND_BUFFER_DRY buffer is mixed into the SOUND_BUFFER_PAINT
extern ConVar dsp_off;
extern ConVar snd_profile;
extern void DEBUG_StartSoundMeasure(int type, int samplecount );
extern void DEBUG_StopSoundMeasure(int type, int samplecount );
extern ConVar dsp_enhance_stereo;
extern ConVar dsp_volume;
extern ConVar dsp_vol_5ch;
extern ConVar dsp_vol_4ch;
extern ConVar dsp_vol_2ch;
extern void MXR_SetCurrentSoundMixer( const char *szsoundmixer );
extern ConVar snd_soundmixer;
void MIX_PaintChannels( int endtime, bool bIsUnderwater )
{
VPROF("MIX_PaintChannels");
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s", __FUNCTION__ );
int end;
int count;
bool b_spatial_delays = dsp_enhance_stereo.GetInt() != 0 ? true : false;
bool room_fsurround_sav;
bool room_fsurround_center_sav;
paintbuffer_t *proom = MIX_GetPPaintFromIPaint(SOUND_BUFFER_ROOM);
CheckNewDspPresets();
MXR_SetCurrentSoundMixer( snd_soundmixer.GetString() );
// dsp performance tuning
g_snd_profile_type = snd_profile.GetInt();
// dsp_off is true if no dsp processing is to run
// directional dsp processing is enabled if dsp_facingaway is non-zero
g_bDspOff = dsp_off.GetInt() ? 1 : 0;
CChannelList list;
MIX_BuildChannelList(list);
// get master dsp volume
g_dsp_volume = dsp_volume.GetFloat();
// attenuate master dsp volume by 2,4 or 5 ch settings
if ( g_AudioDevice->IsSurround() )
{
g_dsp_volume *= ( g_AudioDevice->IsSurroundCenter() ? dsp_vol_5ch.GetFloat() : dsp_vol_4ch.GetFloat() );
}
else
{
g_dsp_volume *= dsp_vol_2ch.GetFloat();
}
if ( !g_bDspOff )
{
g_bdirectionalfx = dsp_facingaway.GetInt() ? 1 : 0;
}
else
{
g_bdirectionalfx = 0;
}
// get dsp preset gain values, update gain crossfaders, used when mixing dsp processed buffers into paintbuffer
SDEBUG_ShowAvgValue();
// the cache needs to hold the audio in memory during mixing, so tell it that mixing is starting
wavedatacache->OnMixBegin();
while ( g_paintedtime < endtime )
{
VPROF("MIX_PaintChannels inner loop");
// mix a full 'paintbuffer' of sound
// clamp at paintbuffer size
end = endtime;
if (endtime - g_paintedtime > PAINTBUFFER_SIZE)
{
end = g_paintedtime + PAINTBUFFER_SIZE;
}
// number of 44khz samples to mix into paintbuffer, up to paintbuffer size
count = end - g_paintedtime;
// clear all mix buffers
g_AudioDevice->MixBegin( count );
// upsample all mix buffers.
// results in 44khz versions of:
// SOUND_BUFFER_ROOM, SOUND_BUFFER_FACING, SOUND_BUFFER_FACINGAWAY, SOUND_BUFFER_DRY, SOUND_BUFFER_SPEAKER, SOUND_BUFFER_SPECIALs
MIX_UpsampleAllPaintbuffers( list, end, count );
// apply appropriate dsp fx to each buffer, remix buffers into single quad output buffer
// apply 2 or 4ch filtering to IFACINGAWAY buffer
if ( g_bdirectionalfx )
{
g_AudioDevice->ApplyDSPEffects( idsp_facingaway, MIX_GetPFrontFromIPaint(SOUND_BUFFER_FACINGAWAY), MIX_GetPRearFromIPaint(SOUND_BUFFER_FACINGAWAY), MIX_GetPCenterFromIPaint(SOUND_BUFFER_FACINGAWAY), count );
}
if ( !g_bDspOff && list.m_hasSpeakerChannels )
{
// apply 1ch filtering to SOUND_BUFFER_SPEAKER
g_AudioDevice->ApplyDSPEffects( idsp_speaker, MIX_GetPFrontFromIPaint(SOUND_BUFFER_SPEAKER), MIX_GetPRearFromIPaint(SOUND_BUFFER_SPEAKER), MIX_GetPCenterFromIPaint(SOUND_BUFFER_SPEAKER), count );
// mix SOUND_BUFFER_SPEAKER with SOUND_BUFFER_ROOM and SOUND_BUFFER_FACING
MIX_ScalePaintBuffer( SOUND_BUFFER_SPEAKER, count, 0.7 );
MIX_MixPaintbuffers( SOUND_BUFFER_SPEAKER, SOUND_BUFFER_FACING, SOUND_BUFFER_FACING, count, 1.0 ); // +70% dry speaker
MIX_ScalePaintBuffer( SOUND_BUFFER_SPEAKER, count, 0.43 );
MIX_MixPaintbuffers( SOUND_BUFFER_SPEAKER, SOUND_BUFFER_ROOM, SOUND_BUFFER_ROOM, count, 1.0 ); // +30% wet speaker
}
if ( !g_bDspOff )
{
// apply 1ch filtering to SOUND_BUFFER_SPECIALs
for ( int iDSP = 0; iDSP < list.m_nSpecialDSPs.Count(); ++iDSP )
{
bool bFoundMixer = false;
for ( int i = SOUND_BUFFER_SPECIAL_START; i < g_paintBuffers.Count(); ++i )
{
paintbuffer_t *pSpecialBuffer = MIX_GetPPaintFromIPaint( i );
if ( pSpecialBuffer->nSpecialDSP == list.m_nSpecialDSPs[ iDSP ] && pSpecialBuffer->idsp_specialdsp != -1 )
{
g_AudioDevice->ApplyDSPEffects( pSpecialBuffer->idsp_specialdsp, MIX_GetPFrontFromIPaint( i ), MIX_GetPRearFromIPaint( i ), MIX_GetPCenterFromIPaint( i ), count );
// mix SOUND_BUFFER_SPECIALs with SOUND_BUFFER_ROOM and SOUND_BUFFER_FACING
MIX_ScalePaintBuffer( i, count, 0.7 );
MIX_MixPaintbuffers( i, SOUND_BUFFER_FACING, SOUND_BUFFER_FACING, count, 1.0 ); // +70% dry speaker
MIX_ScalePaintBuffer( i, count, 0.43 );
MIX_MixPaintbuffers( i, SOUND_BUFFER_ROOM, SOUND_BUFFER_ROOM, count, 1.0 ); // +30% wet speaker
bFoundMixer = true;
break;
}
}
// Couldn't find a mixer with the correct DSP, so make a new one!
if ( !bFoundMixer )
{
bool bSurroundCenter = g_AudioDevice->IsSurroundCenter();
bool bSurround = g_AudioDevice->IsSurround() || bSurroundCenter;
int nIndex = g_paintBuffers.AddToTail();
MIX_InitializePaintbuffer( &(g_paintBuffers[ nIndex ]), bSurround, bSurroundCenter );
g_paintBuffers[ nIndex ].flags = SOUND_BUSS_SPECIAL_DSP;
// special dsp buffer mixes to mono
g_paintBuffers[ nIndex ].fsurround = false;
g_paintBuffers[ nIndex ].fsurround_center = false;
g_paintBuffers[ nIndex ].idsp_specialdsp = -1;
g_paintBuffers[ nIndex ].nSpecialDSP = list.m_nSpecialDSPs[ iDSP ];
g_paintBuffers[ nIndex ].nPrevSpecialDSP = g_paintBuffers[ nIndex ].nSpecialDSP;
g_paintBuffers[ nIndex ].idsp_specialdsp = DSP_Alloc( g_paintBuffers[ nIndex ].nSpecialDSP, 300, 1 );
}
}
}
// apply dsp_room effects to room buffer
g_AudioDevice->ApplyDSPEffects( Get_idsp_room(), MIX_GetPFrontFromIPaint(SOUND_BUFFER_ROOM), MIX_GetPRearFromIPaint(SOUND_BUFFER_ROOM), MIX_GetPCenterFromIPaint(SOUND_BUFFER_ROOM), count );
// save room buffer surround status, in case we upconvert it
room_fsurround_sav = proom->fsurround;
room_fsurround_center_sav = proom->fsurround_center;
// apply left/center/right/lrear/rrear spatial delays to room buffer
if ( b_spatial_delays && !g_bDspOff && !DSP_RoomDSPIsOff() )
{
// upgrade mono room buffer to surround status so we can apply spatial delays to all channels
MIX_ConvertBufferToSurround( SOUND_BUFFER_ROOM );
g_AudioDevice->ApplyDSPEffects( idsp_spatial, MIX_GetPFrontFromIPaint(SOUND_BUFFER_ROOM), MIX_GetPRearFromIPaint(SOUND_BUFFER_ROOM), MIX_GetPCenterFromIPaint(SOUND_BUFFER_ROOM), count );
}
if ( g_bdirectionalfx ) // KDB: perf
{
// Recombine IFACING and IFACINGAWAY buffers into SOUND_BUFFER_PAINT
MIX_MixPaintbuffers( SOUND_BUFFER_FACING, SOUND_BUFFER_FACINGAWAY, SOUND_BUFFER_PAINT, count, DSP_NOROOM_MIX );
// Add in dsp room fx to paintbuffer, mix at 75%
MIX_MixPaintbuffers( SOUND_BUFFER_ROOM, SOUND_BUFFER_PAINT, SOUND_BUFFER_PAINT, count, DSP_ROOM_MIX );
}
else
{
// Mix IFACING buffer with SOUND_BUFFER_ROOM
// (SOUND_BUFFER_FACINGAWAY contains no data, IFACINGBBUFFER has full dry mix based on distance from listener)
// if dsp disabled, mix 100% facingbuffer, otherwise, mix 75% facingbuffer + roombuffer
float mix = g_bDspOff ? 1.0 : DSP_ROOM_MIX;
MIX_MixPaintbuffers( SOUND_BUFFER_ROOM, SOUND_BUFFER_FACING, SOUND_BUFFER_PAINT, count, mix );
}
// restore room buffer surround status, in case we upconverted it
proom->fsurround = room_fsurround_sav;
proom->fsurround_center = room_fsurround_center_sav;
// Apply underwater fx dsp_water (serial in-line)
if ( bIsUnderwater )
{
// BUG: if out of water, previous delays will be heard. must clear dly buffers.
g_AudioDevice->ApplyDSPEffects( idsp_water, MIX_GetPFrontFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPRearFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPCenterFromIPaint(SOUND_BUFFER_PAINT), count );
}
// find dsp gain
SDEBUG_GetAvgIn(SOUND_BUFFER_PAINT, count);
// Apply player fx dsp_player (serial in-line) - does nothing if dsp fx are disabled
g_AudioDevice->ApplyDSPEffects( idsp_player, MIX_GetPFrontFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPRearFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPCenterFromIPaint(SOUND_BUFFER_PAINT), count );
// display dsp gain
SDEBUG_GetAvgOut(SOUND_BUFFER_PAINT, count);
/*
// apply left/center/right/lrear/rrear spatial delays to paint buffer
if ( b_spatial_delays )
g_AudioDevice->ApplyDSPEffects( idsp_spatial, MIX_GetPFrontFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPRearFromIPaint(SOUND_BUFFER_PAINT), MIX_GetPCenterFromIPaint(SOUND_BUFFER_PAINT), count );
*/
// Add dry buffer, set output gain to water * player dsp gain (both 1.0 if not active)
MIX_MixPaintbuffers( SOUND_BUFFER_PAINT, SOUND_BUFFER_DRY, SOUND_BUFFER_PAINT, count, 1.0);
// clip all values > 16 bit down to 16 bit
// NOTE: This is required - the hardware buffer transfer routines no longer perform clipping.
MIX_CompressPaintbuffer( SOUND_BUFFER_PAINT, count );
// transfer SOUND_BUFFER_PAINT paintbuffer out to DMA buffer
MIX_SetCurrentPaintbuffer( SOUND_BUFFER_PAINT );
g_AudioDevice->TransferSamples( end );
g_paintedtime = end;
}
// the cache needs to hold the audio in memory during mixing, so tell it that mixing is complete
wavedatacache->OnMixEnd();
}
// Applies volume scaling (evenly) to all fl,fr,rl,rr volumes
// used for voice ducking and panning between various mix busses
// Ensures if mixing to speaker buffer, only speaker sounds pass through
// Called just before mixing wav data to current paintbuffer.
// a) if another player in a multiplayer game is speaking, scale all volumes down.
// b) if mixing to SOUND_BUFFER_ROOM, scale all volumes by ch.dspmix and dsp_room gain
// c) if mixing to SOUND_BUFFER_FACINGAWAY, scale all volumes by ch.dspface and dsp_facingaway gain
// d) If SURROUND_ON, but buffer is not surround, recombined front/rear volumes
// returns false if channel is to be entirely skipped.
bool MIX_ScaleChannelVolume( paintbuffer_t *ppaint, channel_t *pChannel, int volume[CCHANVOLUMES], int mixchans )
{
int i;
int mixflag = ppaint->flags;
float scale;
char wavtype = pChannel->wavtype;
float dspmix;
// copy current channel volumes into output array
ChannelCopyVolumes( pChannel, volume, 0, CCHANVOLUMES );
dspmix = pChannel->dspmix;
// if dsp is off, or room dsp is off, mix 0% to mono room buffer, 100% to facing buffer
if ( g_bDspOff || DSP_RoomDSPIsOff() )
dspmix = 0.0;
// duck all sound volumes except speaker's voice
#if !defined( NO_VOICE )
int duckScale = min((int)(g_DuckScale * 256), g_SND_VoiceOverdriveInt);
#else
int duckScale = (int)(g_DuckScale * 256);
#endif
if( duckScale < 256 )
{
if( pChannel->pMixer )
{
CAudioSource *pSource = pChannel->pMixer->GetSource();
if( !pSource->IsVoiceSource() )
{
// Apply voice overdrive..
for (i = 0; i < CCHANVOLUMES; i++)
volume[i] = (volume[i] * duckScale) >> 8;
}
}
}
// If mixing to the room buss, adjust volume based on channel's dspmix setting.
// dspmix is DSP_MIX_MAX (~0.78) if sound is far from player, DSP_MIX_MIN (~0.24) if sound is near player
if ( mixflag & SOUND_BUSS_ROOM )
{
// set dsp mix volume, scaled by global dsp_volume
float dspmixvol = fpmin(dspmix * g_dsp_volume, 1.0f);
// if dspmix is 1.0, 100% of sound goes to SOUND_BUFFER_ROOM and 0% to SOUND_BUFFER_FACING
for (i = 0; i < CCHANVOLUMES; i++)
volume[i] = (int)((float)(volume[i]) * dspmixvol);
}
// If global dsp volume is less than 1, reduce dspmix (ie: increase dry volume)
// If gloabl dsp volume is greater than 1, do not reduce dspmix
if (g_dsp_volume < 1.0)
dspmix *= g_dsp_volume;
// If mixing to facing/facingaway buss, adjust volume based on sound entity's facing direction.
// If sound directly faces player, ch->dspface = 1.0. If facing directly away, ch->dspface = -1.0.
// mix to lowpass buffer if facing away, to allpass if facing
// scale 1.0 - facing player, scale 0, facing away
scale = (pChannel->dspface + 1.0) / 2.0;
// UNDONE: get front cone % from channel to set this.
// bias scale such that 1.0 to 'cone' is considered facing. Facing cone narrows as cone -> 1.0
// and 'cone' -> 0.0 becomes 1.0 -> 0.0
float cone = 0.6f;
scale = scale * (1/cone);
scale = clamp( scale, 0.0f, 1.0f );
// pan between facing and facing away buffers
// if ( !g_bdirectionalfx || wavtype == CHAR_DOPPLER || wavtype == CHAR_OMNI || (wavtype == CHAR_DIRECTIONAL && mixchans == 2) )
if ( !g_bdirectionalfx || wavtype != CHAR_DIRECTIONAL )
{
// if no directional fx mix 0% to facingaway buffer
// if wavtype is DOPPLER, mix 0% to facingaway buffer - DOPPLER wavs have a custom mixer
// if wavtype is OMNI, mix 0% to facingaway buffer - OMNI wavs have no directionality
// if wavtype is DIRECTIONAL and stereo encoded, mix 0% to facingaway buffer - DIRECTIONAL STEREO wavs have a custom mixer
scale = 1.0;
}
if ( mixflag & SOUND_BUSS_FACING )
{
// facing player
// if dspface is 1.0, 100% of sound goes to SOUND_BUFFER_FACING
for (i = 0; i < CCHANVOLUMES; i++)
volume[i] = (int)((float)(volume[i]) * scale * (1.0 - dspmix));
}
else if ( mixflag & SOUND_BUSS_FACINGAWAY )
{
// facing away from player
// if dspface is 0.0, 100% of sound goes to SOUND_BUFFER_FACINGAWAY
for (i = 0; i < CCHANVOLUMES; i++)
volume[i] = (int)((float)(volume[i]) * (1.0 - scale) * (1.0 - dspmix));
}
// NOTE: this must occur last in this routine:
if ( g_AudioDevice->IsSurround() && !ppaint->fsurround )
{
// if 4ch or 5ch spatialization on, but current mix buffer is 2ch,
// recombine front + rear volumes (revert to 2ch spatialization)
volume[IFRONT_RIGHT] += volume[IREAR_RIGHT];
volume[IFRONT_LEFT] += volume[IREAR_LEFT];
volume[IFRONT_RIGHTD] += volume[IREAR_RIGHTD];
volume[IFRONT_LEFTD] += volume[IREAR_LEFTD];
// if 5 ch, recombine center channel vol
if ( g_AudioDevice->IsSurroundCenter() )
{
volume[IFRONT_RIGHT] += volume[IFRONT_CENTER] / 2;
volume[IFRONT_LEFT] += volume[IFRONT_CENTER] / 2;
volume[IFRONT_RIGHTD] += volume[IFRONT_CENTERD] / 2;
volume[IFRONT_LEFTD] += volume[IFRONT_CENTERD] / 2;
}
// clear rear & center volumes
volume[IREAR_RIGHT] = 0;
volume[IREAR_LEFT] = 0;
volume[IFRONT_CENTER] = 0;
volume[IREAR_RIGHTD] = 0;
volume[IREAR_LEFTD] = 0;
volume[IFRONT_CENTERD] = 0;
}
bool fzerovolume = true;
for (i = 0; i < CCHANVOLUMES; i++)
{
volume[i] = clamp(volume[i], 0, 255);
if (volume[i])
fzerovolume = false;
}
if ( fzerovolume )
{
// DevMsg ("Skipping mix of 0 volume sound! \n");
return false;
}
return true;
}
//===============================================================================
// Low level mixing routines
//===============================================================================
void Snd_WriteLinearBlastStereo16( void )
{
#if !id386
int i;
int val;
for ( i=0; i<snd_linear_count; i+=2 )
{
// scale and clamp left 16bit signed: [0x8000, 0x7FFF]
val = ( snd_p[i] * snd_vol )>>8;
if ( val > 32767 )
snd_out[i] = 32767;
else if ( val < -32768 )
snd_out[i] = -32768;
else
snd_out[i] = val;
// scale and clamp right 16bit signed: [0x8000, 0x7FFF]
val = ( snd_p[i+1] * snd_vol )>>8;
if ( val > 32767 )
snd_out[i+1] = 32767;
else if ( val < -32768 )
snd_out[i+1] = -32768;
else
snd_out[i+1] = val;
}
#else
__asm
{
// input data
mov ebx,snd_p
// output data
mov edi,snd_out
// iterate from end to beginning
mov ecx,snd_linear_count
// scale table
mov esi,snd_vol
// scale and clamp 16bit signed lsw: [0x8000, 0x7FFF]
WLBS16_LoopTop:
mov eax,[ebx+ecx*4-8]
imul eax,esi
sar eax,0x08
cmp eax,0x7FFF
jg WLBS16_ClampHigh
cmp eax,0xFFFF8000
jnl WLBS16_ClampDone
mov eax,0xFFFF8000
jmp WLBS16_ClampDone
WLBS16_ClampHigh:
mov eax,0x7FFF
WLBS16_ClampDone:
// scale and clamp 16bit signed msw: [0x8000, 0x7FFF]
mov edx,[ebx+ecx*4-4]
imul edx,esi
sar edx,0x08
cmp edx,0x7FFF
jg WLBS16_ClampHigh2
cmp edx,0xFFFF8000
jnl WLBS16_ClampDone2
mov edx,0xFFFF8000
jmp WLBS16_ClampDone2
WLBS16_ClampHigh2:
mov edx,0x7FFF
WLBS16_ClampDone2:
shl edx,0x10
and eax,0xFFFF
or edx,eax
mov [edi+ecx*2-4],edx
// two shorts per iteration
sub ecx,0x02
jnz WLBS16_LoopTop
}
#endif
}
void SND_InitScaletable (void)
{
int i, j;
for (i=0 ; i<SND_SCALE_LEVELS; i++)
for (j=0 ; j<256 ; j++)
snd_scaletable[i][j] = ((signed char)j) * i * (1<<SND_SCALE_SHIFT);
}
void SND_PaintChannelFrom8(portable_samplepair_t *pOutput, int *volume, byte *pData8, int count)
{
#if !id386
int data;
int *lscale, *rscale;
int i;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
for (i=0 ; i<count ; i++)
{
data = pData8[i];
pOutput[i].left += lscale[data];
pOutput[i].right += rscale[data];
}
#else
// portable_samplepair_t structure
#define psp_left 0
#define psp_right 4
#define psp_size 8
static int tempStore;
__asm
{
// prologue
push ebp
// esp = pOutput
mov eax, pOutput
mov tempStore, eax
xchg esp,tempStore
// ebx = volume
mov ebx,volume
// esi = pData8
mov esi,pData8
// ecx = count
mov ecx,count
// These values depend on the setting of SND_SCALE_BITS
// The mask must mask off all the lower bits you aren't using in the multiply
// so for 7 bits, the mask is 0xFE, 6 bits 0xFC, etc.
// The shift must multiply by the table size. There are 256 4-byte values in the table at each level.
// So each index must be shifted left by 10, but since the bits we use are in the MSB rather than LSB
// they must be shifted right by 8 - SND_SCALE_BITS. e.g., for a 7 bit number the left shift is:
// 10 - (8-7) = 9. For a 5 bit number it's 10 - (8-5) = 7.
mov eax,[ebx]
mov edx,[ebx + 4]
and eax,0xFE
and edx,0xFE
// shift up by 10 to index table, down by 1 to make the 7 MSB of the bytes an index
// eax = lscale
// edx = rscale
shl eax,0x09
shl edx,0x09
add eax,OFFSET snd_scaletable
add edx,OFFSET snd_scaletable
// ebx = data byte
sub ebx,ebx
mov bl,[esi+ecx-1]
// odd or even number of L/R samples
test ecx,0x01
jz PCF8_Loop
// process odd L/R sample
mov edi,[eax+ebx*4]
mov ebp,[edx+ebx*4]
add edi,[esp+ecx*psp_size-psp_size+psp_left]
add ebp,[esp+ecx*psp_size-psp_size+psp_right]
mov [esp+ecx*psp_size-psp_size+psp_left],edi
mov [esp+ecx*psp_size-psp_size+psp_right],ebp
mov bl,[esi+ecx-1-1]
dec ecx
jz PCF8_Done
PCF8_Loop:
// process L/R sample N
mov edi,[eax+ebx*4]
mov ebp,[edx+ebx*4]
add edi,[esp+ecx*psp_size-psp_size+psp_left]
add ebp,[esp+ecx*psp_size-psp_size+psp_right]
mov [esp+ecx*psp_size-psp_size+psp_left],edi
mov [esp+ecx*psp_size-psp_size+psp_right],ebp
mov bl,[esi+ecx-1-1]
// process L/R sample N-1
mov edi,[eax+ebx*4]
mov ebp,[edx+ebx*4]
add edi,[esp+ecx*psp_size-psp_size*2+psp_left]
add ebp,[esp+ecx*psp_size-psp_size*2+psp_right]
mov [esp+ecx*psp_size-psp_size*2+psp_left],edi
mov [esp+ecx*psp_size-psp_size*2+psp_right],ebp
mov bl,[esi+ecx-1-2]
// two L/R samples per iteration
sub ecx,0x02
jnz PCF8_Loop
PCF8_Done:
// epilogue
xchg esp,tempStore
pop ebp
}
#endif
}
//===============================================================================
// SOFTWARE MIXING ROUTINES
//===============================================================================
// UNDONE: optimize these
// grab samples from left source channel only and mix as if mono.
// volume array contains appropriate spatialization volumes for doppler left (incoming sound)
void SW_Mix8StereoDopplerLeft( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += lscale[pData[sampleIndex]];
pOutput[i].right += rscale[pData[sampleIndex]];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// grab samples from right source channel only and mix as if mono.
// volume array contains appropriate spatialization volumes for doppler right (outgoing sound)
void SW_Mix8StereoDopplerRight( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += lscale[pData[sampleIndex+1]];
pOutput[i].right += rscale[pData[sampleIndex+1]];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// grab samples from left source channel only and mix as if mono.
// volume array contains appropriate spatialization volumes for doppler left (incoming sound)
void SW_Mix16StereoDopplerLeft( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += (volume[0] * (int)(pData[sampleIndex]))>>8;
pOutput[i].right += (volume[1] * (int)(pData[sampleIndex]))>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// grab samples from right source channel only and mix as if mono.
// volume array contains appropriate spatialization volumes for doppler right (outgoing sound)
void SW_Mix16StereoDopplerRight( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += (volume[0] * (int)(pData[sampleIndex+1]))>>8;
pOutput[i].right += (volume[1] * (int)(pData[sampleIndex+1]))>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// mix left wav (front facing) with right wav (rear facing) based on soundfacing direction
void SW_Mix8StereoDirectional( float soundfacing, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int x;
int l,r;
signed char lb,rb;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
// if soundfacing -1.0, sound source is facing away from player
// if soundfacing 0.0, sound source is perpendicular to player
// if soundfacing 1.0, sound source is facing player
int frontmix = (int)(256.0f * ((1.f + soundfacing) / 2.f)); // 0 -> 256
for ( int i = 0; i < outCount; i++ )
{
lb = (pData[sampleIndex]); // get left byte
rb = (pData[sampleIndex+1]); // get right byte
l = ((int)lb);
r = ((int)rb);
x = ( r + ((( l - r ) * frontmix) >> 8) );
pOutput[i].left += lscale[x & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// mix left wav (front facing) with right wav (rear facing) based on soundfacing direction
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix8StereoDirectional_Interp( float soundfacing, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interpl, interpr;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
int x;
// if soundfacing -1.0, sound source is facing away from player
// if soundfacing 0.0, sound source is perpendicular to player
// if soundfacing 1.0, sound source is facing player
int frontmix = (int)(256.0f * ((1.f + soundfacing) / 2.f)); // 0 -> 256
for ( int i = 0; i < outCount; i++ )
{
// interpolate between first & second sample (the samples bordering sampleFrac12 fraction)
first = (int)((signed char)(pData[sampleIndex])); // left byte
second = (int)((signed char)(pData[sampleIndex+2]));
interpl = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
first = (int)((signed char)(pData[sampleIndex+1])); // right byte
second = (int)((signed char)(pData[sampleIndex+3]));
interpr = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
// crossfade between right/left based on directional mix
x = ( interpr + ((( interpl - interpr ) * frontmix) >> 8) );
pOutput[i].left += lscale[x & 0xFF]; // scale and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
// mix left wav (front facing) with right wav (rear facing) based on soundfacing direction
void SW_Mix16StereoDirectional( float soundfacing, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
fixedint sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int x;
int l, r;
// if soundfacing -1.0, sound source is facing away from player
// if soundfacing 0.0, sound source is perpendicular to player
// if soundfacing 1.0, sound source is facing player
int frontmix = (int)(256.0f * ((1.f + soundfacing) / 2.f)); // 0 -> 256
for ( int i = 0; i < outCount; i++ )
{
// get left, right samples
l = (int)(pData[sampleIndex]);
r = (int)(pData[sampleIndex+1]);
// crossfade between left & right based on front/rear facing
x = ( r + ((( l - r ) * frontmix) >> 8) );
pOutput[i].left += (volume[0] * x) >> 8;
pOutput[i].right += (volume[1] * x) >> 8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// mix left wav (front facing) with right wav (rear facing) based on soundfacing direction
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix16StereoDirectional_Interp( float soundfacing, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int x;
int first, second, interpl, interpr;
// if soundfacing -1.0, sound source is facing away from player
// if soundfacing 0.0, sound source is perpendicular to player
// if soundfacing 1.0, sound source is facing player
int frontmix = (int)(256.0f * ((1.f + soundfacing) / 2.f)); // 0 -> 256
for ( int i = 0; i < outCount; i++ )
{
// get interpolated left, right samples
first = (int)(pData[sampleIndex]);
second = (int)(pData[sampleIndex+2]);
interpl = first + (((second - first) * (int)sampleFrac14) >> 14);
first = (int)(pData[sampleIndex+1]);
second = (int)(pData[sampleIndex+3]);
interpr = first + (((second - first) * (int)sampleFrac14) >> 14);
// crossfade between left & right based on front/rear facing
x = ( interpr + ((( interpl - interpr ) * frontmix) >> 8) );
pOutput[i].left += (volume[0] * x) >> 8;
pOutput[i].right += (volume[1] * x) >> 8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
// distance variant wav (left is close, right is far)
void SW_Mix8StereoDistVar( float distmix, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int x;
int l,r;
signed char lb, rb;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
// distmix 0 - sound is near player (100% wav left)
// distmix 1.0 - sound is far from player (100% wav right)
int nearmix = (int)(256.0f * (1.0f - distmix));
int farmix = (int)(256.0f * distmix);
// if mixing at max or min range, skip crossfade (KDB: perf)
if (!nearmix)
{
for ( int i = 0; i < outCount; i++ )
{
rb = (pData[sampleIndex+1]); // get right byte
x = (int) rb;
pOutput[i].left += lscale[x & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
return;
}
if (!farmix)
{
for ( int i = 0; i < outCount; i++ )
{
lb = (pData[sampleIndex]); // get left byte
x = (int) lb;
pOutput[i].left += lscale[x & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
return;
}
// crossfade left/right
for ( int i = 0; i < outCount; i++ )
{
lb = (pData[sampleIndex]); // get left byte
rb = (pData[sampleIndex+1]); // get right byte
l = (int)lb;
r = (int)rb;
x = ( l + (((r - l) * farmix ) >> 8) );
pOutput[i].left += lscale[x & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// distance variant wav (left is close, right is far)
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix8StereoDistVar_Interp( float distmix, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int x;
// distmix 0 - sound is near player (100% wav left)
// distmix 1.0 - sound is far from player (100% wav right)
int nearmix = (int)(256.0f * (1.0f - distmix));
int farmix = (int)(256.0f * distmix);
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interpl, interpr;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
// if mixing at max or min range, skip crossfade (KDB: perf)
if (!nearmix)
{
for ( int i = 0; i < outCount; i++ )
{
first = (int)((signed char)(pData[sampleIndex+1])); // right sample
second = (int)((signed char)(pData[sampleIndex+3]));
interpr = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
pOutput[i].left += lscale[interpr & 0xFF]; // scale and convert to 16 bit
pOutput[i].right += rscale[interpr & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
return;
}
if (!farmix)
{
for ( int i = 0; i < outCount; i++ )
{
first = (int)((signed char)(pData[sampleIndex])); // left sample
second = (int)((signed char)(pData[sampleIndex+2]));
interpl = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
pOutput[i].left += lscale[interpl & 0xFF]; // scale and convert to 16 bit
pOutput[i].right += rscale[interpl & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
return;
}
// crossfade left/right
for ( int i = 0; i < outCount; i++ )
{
// interpolate between first & second sample (the samples bordering sampleFrac14 fraction)
first = (int)((signed char)(pData[sampleIndex]));
second = (int)((signed char)(pData[sampleIndex+2]));
interpl = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
first = (int)((signed char)(pData[sampleIndex+1]));
second = (int)((signed char)(pData[sampleIndex+3]));
interpr = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
// crossfade between left and right based on distance mix
x = ( interpl + (((interpr - interpl) * farmix ) >> 8) );
pOutput[i].left += lscale[x & 0xFF]; // scale and convert to 16 bit
pOutput[i].right += rscale[x & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
// distance variant wav (left is close, right is far)
void SW_Mix16StereoDistVar( float distmix, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int x;
int l,r;
// distmix 0 - sound is near player (100% wav left)
// distmix 1.0 - sound is far from player (100% wav right)
int nearmix = Float2Int(256.0f * (1.f - distmix));
int farmix = Float2Int(256.0f * distmix);
// if mixing at max or min range, skip crossfade (KDB: perf)
if (!nearmix)
{
for ( int i = 0; i < outCount; i++ )
{
x = pData[sampleIndex+1]; // right sample
pOutput[i].left += (volume[0] * x)>>8;
pOutput[i].right += (volume[1] * x)>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
return;
}
if (!farmix)
{
for ( int i = 0; i < outCount; i++ )
{
x = pData[sampleIndex]; // left sample
pOutput[i].left += (volume[0] * x)>>8;
pOutput[i].right += (volume[1] * x)>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
return;
}
// crossfade left/right
for ( int i = 0; i < outCount; i++ )
{
l = pData[sampleIndex];
r = pData[sampleIndex+1];
x = ( l + (((r - l) * farmix) >> 8) );
pOutput[i].left += (volume[0] * x)>>8;
pOutput[i].right += (volume[1] * x)>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// distance variant wav (left is close, right is far)
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix16StereoDistVar_Interp( float distmix, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int x;
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interpl, interpr;
// distmix 0 - sound is near player (100% wav left)
// distmix 1.0 - sound is far from player (100% wav right)
int nearmix = Float2Int(256.0f * (1.f - distmix));
int farmix = Float2Int(256.0f * distmix);
// if mixing at max or min range, skip crossfade (KDB: perf)
if (!nearmix)
{
for ( int i = 0; i < outCount; i++ )
{
first = (int)(pData[sampleIndex+1]); // right sample
second = (int)(pData[sampleIndex+3]);
interpr = first + (((second - first) * (int)sampleFrac14) >> 14);
pOutput[i].left += (volume[0] * interpr)>>8;
pOutput[i].right += (volume[1] * interpr)>>8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
return;
}
if (!farmix)
{
for ( int i = 0; i < outCount; i++ )
{
first = (int)(pData[sampleIndex]); // left sample
second = (int)(pData[sampleIndex+2]);
interpl = first + (((second - first) * (int)sampleFrac14) >> 14);
pOutput[i].left += (volume[0] * interpl)>>8;
pOutput[i].right += (volume[1] * interpl)>>8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
return;
}
// crossfade left/right
for ( int i = 0; i < outCount; i++ )
{
first = (int)(pData[sampleIndex]);
second = (int)(pData[sampleIndex+2]);
interpl = first + (((second - first) * (int)sampleFrac14) >> 14);
first = (int)(pData[sampleIndex+1]);
second = (int)(pData[sampleIndex+3]);
interpr = first + (((second - first) * (int)sampleFrac14) >> 14);
// crossfade between left & right samples
x = ( interpl + (((interpr - interpl) * farmix) >> 8) );
pOutput[i].left += (volume[0] * x) >> 8;
pOutput[i].right += (volume[1] * x) >> 8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
void SW_Mix8Mono( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
// Not using pitch shift?
if ( rateScaleFix == FIX(1) )
{
// native code
SND_PaintChannelFrom8( pOutput, volume, (byte *)pData, outCount );
return;
}
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += lscale[pData[sampleIndex]];
pOutput[i].right += rscale[pData[sampleIndex]];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac);
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix8Mono_Interp( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount)
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interp;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
// iterate 0th sample to outCount-1 sample
for (int i = 0; i < outCount; i++ )
{
// interpolate between first & second sample (the samples bordering sampleFrac12 fraction)
first = (int)((signed char)(pData[sampleIndex]));
second = (int)((signed char)(pData[sampleIndex+1]));
interp = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
pOutput[i].left += lscale[interp & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[interp & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14);
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
void SW_Mix8Stereo( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += lscale[pData[sampleIndex]];
pOutput[i].right += rscale[pData[sampleIndex+1]];
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix8Stereo_Interp( portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount)
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interpl, interpr;
int *lscale, *rscale;
lscale = snd_scaletable[volume[0] >> SND_SCALE_SHIFT];
rscale = snd_scaletable[volume[1] >> SND_SCALE_SHIFT];
// iterate 0th sample to outCount-1 sample
for (int i = 0; i < outCount; i++ )
{
// interpolate between first & second sample (the samples bordering sampleFrac12 fraction)
first = (int)((signed char)(pData[sampleIndex])); // left
second = (int)((signed char)(pData[sampleIndex+2]));
interpl = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
first = (int)((signed char)(pData[sampleIndex+1])); // right
second = (int)((signed char)(pData[sampleIndex+3]));
interpr = first + ( ((second - first) * (int)sampleFrac14) >> 14 );
pOutput[i].left += lscale[interpl & 0xFF]; // multiply by volume and convert to 16 bit
pOutput[i].right += rscale[interpr & 0xFF];
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
void SW_Mix16Mono_Shift( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int vol0 = volume[0];
int vol1 = volume[1];
#if !id386
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += (vol0 * (int)(pData[sampleIndex]))>>8;
pOutput[i].right += (vol1 * (int)(pData[sampleIndex]))>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac);
sampleFrac = FIX_FRACPART(sampleFrac);
}
#else
// in assembly, you can make this 32.32 instead of 4.28 and use the carry flag instead of masking
int rateScaleInt = FIX_INTPART(rateScaleFix);
unsigned int rateScaleFrac = FIX_FRACPART(rateScaleFix) << (32-FIX_BITS);
__asm
{
mov eax, volume ;
movq mm0, DWORD PTR [eax] ; vol1, vol0 (32-bits each)
packssdw mm0, mm0 ; pack and replicate... vol1, vol0, vol1, vol0 (16-bits each)
//pxor mm7, mm7 ; mm7 is my zero register...
xor esi, esi
mov eax, DWORD PTR [pOutput] ; store initial output ptr
mov edx, DWORD PTR [pData] ; store initial input ptr
mov ebx, inputOffset;
mov ecx, outCount;
BEGINLOAD:
movd mm2, WORD PTR [edx+2*esi] ; load first piece of data from pData
punpcklwd mm2, mm2 ; 0, 0, pData_1st, pData_1st
add ebx, rateScaleFrac ; do the crazy fixed integer math
adc esi, rateScaleInt
movd mm3, WORD PTR [edx+2*esi] ; load second piece of data from pData
punpcklwd mm3, mm3 ; 0, 0, pData_2nd, pData_2nd
punpckldq mm2, mm3 ; pData_2nd, pData_2nd, pData_2nd, pData_2nd
add ebx, rateScaleFrac ; do the crazy fixed integer math
adc esi, rateScaleInt
movq mm3, mm2 ; copy the goods
pmullw mm2, mm0 ; pData_2nd*vol1, pData_2nd*vol0, pData_1st*vol1, pData_1st*vol0 (bits 0-15)
pmulhw mm3, mm0 ; pData_2nd*vol1, pData_2nd*vol0, pData_1st*vol1, pData_1st*vol0 (bits 16-31)
movq mm4, mm2 ; copy
movq mm5, mm3 ; copy
punpcklwd mm2, mm3 ; pData_1st*vol1, pData_1st*vol0 (bits 0-31)
punpckhwd mm4, mm5 ; pData_2nd*vol1, pData_2nd*vol0 (bits 0-31)
psrad mm2, 8 ; shift right by 8
psrad mm4, 8 ; shift right by 8
add ecx, -2 ; decrement i-value
paddd mm2, QWORD PTR [eax] ; add to existing vals
paddd mm4, QWORD PTR [eax+8] ;
movq QWORD PTR [eax], mm2 ; store back
movq QWORD PTR [eax+8], mm4 ;
add eax, 10h ;
cmp ecx, 01h ; see if we can quit
jg BEGINLOAD ; Kipp Owens is a doof...
jl END ; Nick Shaffner is killing me...
movsx edi, WORD PTR [edx+2*esi] ; load first 16 bit val and zero-extend
imul edi, vol0 ; multiply pData[sampleIndex] by volume[0]
sar edi, 08h ; divide by 256
add DWORD PTR [eax], edi ; add to pOutput[i].left
movsx edi, WORD PTR [edx+2*esi] ; load same 16 bit val and zero-extend (cuz I thrashed the reg)
imul edi, vol1 ; multiply pData[sampleIndex] by volume[1]
sar edi, 08h ; divide by 256
add DWORD PTR [eax+04h], edi ; add to pOutput[i].right
END:
emms;
}
#endif
}
void SW_Mix16Mono_NoShift( portable_samplepair_t *pOutput, int *volume, short *pData, int outCount )
{
int vol0 = volume[0];
int vol1 = volume[1];
#if !id386
for ( int i = 0; i < outCount; i++ )
{
int x = *pData++;
pOutput[i].left += (x * vol0) >> 8;
pOutput[i].right += (x * vol1) >> 8;
}
#else
__asm
{
mov eax, volume ;
movq mm0, DWORD PTR [eax] ; vol1, vol0 (32-bits each)
packssdw mm0, mm0 ; pack and replicate... vol1, vol0, vol1, vol0 (16-bits each)
//pxor mm7, mm7 ; mm7 is my zero register...
mov eax, DWORD PTR [pOutput] ; store initial output ptr
mov edx, DWORD PTR [pData] ; store initial input ptr
mov ecx, outCount;
BEGINLOAD:
movd mm2, WORD PTR [edx] ; load first piece o data from pData
punpcklwd mm2, mm2 ; 0, 0, pData_1st, pData_1st
add edx,2 ; move to the next sample
movd mm3, WORD PTR [edx] ; load second piece o data from pData
punpcklwd mm3, mm3 ; 0, 0, pData_2nd, pData_2nd
punpckldq mm2, mm3 ; pData_2nd, pData_2nd, pData_2nd, pData_2nd
add edx,2 ; move to the next sample
movq mm3, mm2 ; copy the goods
pmullw mm2, mm0 ; pData_2nd*vol1, pData_2nd*vol0, pData_1st*vol1, pData_1st*vol0 (bits 0-15)
pmulhw mm3, mm0 ; pData_2nd*vol1, pData_2nd*vol0, pData_1st*vol1, pData_1st*vol0 (bits 16-31)
movq mm4, mm2 ; copy
movq mm5, mm3 ; copy
punpcklwd mm2, mm3 ; pData_1st*vol1, pData_1st*vol0 (bits 0-31)
punpckhwd mm4, mm5 ; pData_2nd*vol1, pData_2nd*vol0 (bits 0-31)
psrad mm2, 8 ; shift right by 8
psrad mm4, 8 ; shift right by 8
add ecx, -2 ; decrement i-value
paddd mm2, QWORD PTR [eax] ; add to existing vals
paddd mm4, QWORD PTR [eax+8] ;
movq QWORD PTR [eax], mm2 ; store back
movq QWORD PTR [eax+8], mm4 ;
add eax, 10h ;
cmp ecx, 01h ; see if we can quit
jg BEGINLOAD ; I can cut and paste code!
jl END ;
movsx edi, WORD PTR [edx] ; load first 16 bit val and zero-extend
mov esi,edi ; save a copy for the other channel
imul edi, vol0 ; multiply pData[sampleIndex] by volume[0]
sar edi, 08h ; divide by 256
add DWORD PTR [eax], edi ; add to pOutput[i].left
; esi has a copy, use it now
imul esi, vol1 ; multiply pData[sampleIndex] by volume[1]
sar esi, 08h ; divide by 256
add DWORD PTR [eax+04h], esi ; add to pOutput[i].right
END:
emms;
}
#endif
}
void SW_Mix16Mono( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
if ( rateScaleFix == FIX(1) )
{
SW_Mix16Mono_NoShift( pOutput, volume, pData, outCount );
}
else
{
SW_Mix16Mono_Shift( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
}
}
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix16Mono_Interp( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interp;
for ( int i = 0; i < outCount; i++ )
{
first = (int)(pData[sampleIndex]);
second = (int)(pData[sampleIndex+1]);
interp = first + (((second - first) * (int)sampleFrac14) >> 14);
pOutput[i].left += (volume[0] * interp) >> 8;
pOutput[i].right += (volume[1] * interp) >> 8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14);
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
void SW_Mix16Stereo( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
int sampleIndex = 0;
fixedint sampleFrac = inputOffset;
for ( int i = 0; i < outCount; i++ )
{
pOutput[i].left += (volume[0] * (int)(pData[sampleIndex]))>>8;
pOutput[i].right += (volume[1] * (int)(pData[sampleIndex+1]))>>8;
sampleFrac += rateScaleFix;
sampleIndex += FIX_INTPART(sampleFrac)<<1;
sampleFrac = FIX_FRACPART(sampleFrac);
}
}
// interpolating pitch shifter - sample(s) from preceding buffer are preloaded in
// pData buffer, ensuring we can always provide 'outCount' samples.
void SW_Mix16Stereo_Interp( portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
fixedint sampleIndex = 0;
fixedint rateScaleFix14 = FIX_28TO14(rateScaleFix); // convert 28 bit fixed point to 14 bit fixed point
fixedint sampleFrac14 = FIX_28TO14(inputOffset);
int first, second, interpl, interpr;
for ( int i = 0; i < outCount; i++ )
{
first = (int)(pData[sampleIndex]);
second = (int)(pData[sampleIndex+2]);
interpl = first + (((second - first) * (int)sampleFrac14) >> 14);
first = (int)(pData[sampleIndex+1]);
second = (int)(pData[sampleIndex+3]);
interpr = first + (((second - first) * (int)sampleFrac14) >> 14);
pOutput[i].left += (volume[0] * interpl) >> 8;
pOutput[i].right += (volume[1] * interpr) >> 8;
sampleFrac14 += rateScaleFix14;
sampleIndex += FIX_INTPART14(sampleFrac14)<<1;
sampleFrac14 = FIX_FRACPART14(sampleFrac14);
}
}
// return true if mixer should use high quality pitch interpolation for this sound
bool FUseHighQualityPitch( channel_t *pChannel )
{
// do not use interpolating pitch shifter if:
// low quality flag set on sound (ie: wave name is prepended with CHAR_FAST_PITCH)
// or pitch has no fractional part
// or snd_pitchquality is 0
if ( !snd_pitchquality.GetInt() || pChannel->flags.bfast_pitch )
return false;
return ( (pChannel->pitch != floor(pChannel->pitch)) );
}
//===============================================================================
// DISPATCHERS FOR MIXING ROUTINES
//===============================================================================
void Mix8MonoWavtype( channel_t *pChannel, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix8Mono_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix8Mono( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
}
void Mix16MonoWavtype( channel_t *pChannel, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix16Mono_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
// fast native coded mixers with lower quality pitch shift
SW_Mix16Mono( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
}
void Mix8StereoWavtype( channel_t *pChannel, portable_samplepair_t *pOutput, int *volume, byte *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
switch ( pChannel->wavtype )
{
case CHAR_DOPPLER:
SW_Mix8StereoDopplerLeft( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
SW_Mix8StereoDopplerRight( pOutput, &volume[IFRONT_LEFTD], pData, inputOffset, rateScaleFix, outCount );
break;
case CHAR_DIRECTIONAL:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix8StereoDirectional_Interp( pChannel->dspface, pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix8StereoDirectional( pChannel->dspface, pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
case CHAR_DISTVARIANT:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix8StereoDistVar_Interp( pChannel->distmix, pOutput, volume, pData, inputOffset, rateScaleFix, outCount);
else
SW_Mix8StereoDistVar( pChannel->distmix, pOutput, volume, pData, inputOffset, rateScaleFix, outCount);
break;
case CHAR_OMNI:
// non directional stereo - all channel volumes are the same
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix8Stereo_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix8Stereo( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
default:
case CHAR_SPATIALSTEREO:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix8Stereo_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix8Stereo( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
}
}
void Mix16StereoWavtype( channel_t *pChannel, portable_samplepair_t *pOutput, int *volume, short *pData, int inputOffset, fixedint rateScaleFix, int outCount )
{
switch ( pChannel->wavtype )
{
case CHAR_DOPPLER:
SW_Mix16StereoDopplerLeft( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
SW_Mix16StereoDopplerRight( pOutput, &volume[IFRONT_LEFTD], pData, inputOffset, rateScaleFix, outCount );
break;
case CHAR_DIRECTIONAL:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix16StereoDirectional_Interp( pChannel->dspface, pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix16StereoDirectional( pChannel->dspface, pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
case CHAR_DISTVARIANT:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix16StereoDistVar_Interp( pChannel->distmix, pOutput, volume, pData, inputOffset, rateScaleFix, outCount);
else
SW_Mix16StereoDistVar( pChannel->distmix, pOutput, volume, pData, inputOffset, rateScaleFix, outCount);
break;
case CHAR_OMNI:
// non directional stereo - all channel volumes are same
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix16Stereo_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix16Stereo( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
default:
case CHAR_SPATIALSTEREO:
if ( FUseHighQualityPitch( pChannel ) )
SW_Mix16Stereo_Interp( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
else
SW_Mix16Stereo( pOutput, volume, pData, inputOffset, rateScaleFix, outCount );
break;
}
}
//===============================================================================
// Client entity mouth movement code. Set entity mouthopen variable, based
// on the sound envelope of the voice channel playing.
// KellyB 10/22/97
//===============================================================================
extern IBaseClientDLL *g_ClientDLL;
// called when voice channel is first opened on this entity
static CMouthInfo *GetMouthInfoForChannel( channel_t *pChannel )
{
#ifndef DEDICATED
// If it's a sound inside the client UI, ask the client for the mouthinfo
if ( pChannel->soundsource == SOUND_FROM_UI_PANEL )
return g_ClientDLL ? g_ClientDLL->GetClientUIMouthInfo() : NULL;
#endif
int mouthentity = pChannel->speakerentity == -1 ? pChannel->soundsource : pChannel->speakerentity;
IClientEntity *pClientEntity = entitylist->GetClientEntity( mouthentity );
if( !pClientEntity )
return NULL;
return pClientEntity->GetMouth();
}
void SND_InitMouth( channel_t *pChannel )
{
if ( SND_IsMouth( pChannel ) )
{
CMouthInfo *pMouth = GetMouthInfoForChannel(pChannel);
// init mouth movement vars
if ( pMouth )
{
pMouth->mouthopen = 0;
pMouth->sndavg = 0;
pMouth->sndcount = 0;
if ( pChannel->sfx->pSource && pChannel->sfx->pSource->GetSentence() )
{
pMouth->AddSource( pChannel->sfx->pSource, pChannel->flags.m_bIgnorePhonemes );
}
}
}
}
// called when channel stops
void SND_CloseMouth(channel_t *pChannel)
{
if ( SND_IsMouth( pChannel ) )
{
CMouthInfo *pMouth = GetMouthInfoForChannel(pChannel);
if ( pMouth )
{
// shut mouth
int idx = pMouth->GetIndexForSource( pChannel->sfx->pSource );
if ( idx != UNKNOWN_VOICE_SOURCE )
{
pMouth->RemoveSourceByIndex(idx);
}
else
{
pMouth->ClearVoiceSources();
}
pMouth->mouthopen = 0;
}
}
}
#define CAVGSAMPLES 10
// need this to make the debug code below work.
//#include "snd_wave_source.h"
void SND_MoveMouth8( channel_t *ch, CAudioSource *pSource, int count )
{
int data;
char *pdata = NULL;
int i;
int savg;
int scount;
CMouthInfo *pMouth = GetMouthInfoForChannel( ch );
if ( !pMouth )
return;
if ( pSource->GetSentence() )
{
int idx = pMouth->GetIndexForSource( pSource );
if ( idx == UNKNOWN_VOICE_SOURCE )
{
if ( pMouth->AddSource( pSource, ch->flags.m_bIgnorePhonemes ) == NULL )
{
DevMsg( 1, "out of voice sources, won't lipsync %s\n", ch->sfx->getname() );
#if 0
for ( int i = 0; i < pMouth->GetNumVoiceSources(); i++ )
{
CVoiceData *pVoice = pMouth->GetVoiceSource(i);
CAudioSourceWave *pWave = dynamic_cast<CAudioSourceWave *>(pVoice->GetSource());
const char *pName = "unknown";
if ( pWave && pWave->GetName() )
pName = pWave->GetName();
Msg("Playing %s...\n", pName );
}
#endif
}
}
else
{
// Update elapsed time from mixer
CVoiceData *vd = pMouth->GetVoiceSource( idx );
Assert( vd );
if ( vd )
{
Assert( pSource->SampleRate() > 0 );
float elapsed = ( float )ch->pMixer->GetSamplePosition() / ( float )pSource->SampleRate();
vd->SetElapsedTime( elapsed );
}
}
}
if ( IsX360() )
{
// not supporting because data is assumed to be 8 bit and bypasses mixer (decoding)
return;
}
if ( pMouth->NeedsEnvelope() )
{
int availableSamples = pSource->GetOutputData((void**)&pdata, ch->pMixer->GetSamplePosition(), count, NULL );
if( pdata == NULL )
return;
i = 0;
scount = pMouth->sndcount;
savg = 0;
while ( i < availableSamples && scount < CAVGSAMPLES )
{
data = pdata[i];
savg += abs(data);
i += 80 + ((byte)data & 0x1F);
scount++;
}
pMouth->sndavg += savg;
pMouth->sndcount = (byte) scount;
if ( pMouth->sndcount >= CAVGSAMPLES )
{
pMouth->mouthopen = pMouth->sndavg / CAVGSAMPLES;
pMouth->sndavg = 0;
pMouth->sndcount = 0;
}
}
else
{
pMouth->mouthopen = 0;
}
}
void SND_UpdateMouth( channel_t *pChannel )
{
CMouthInfo *m = GetMouthInfoForChannel( pChannel );
if ( !m )
return;
if ( pChannel->sfx )
{
m->AddSource( pChannel->sfx->pSource, pChannel->flags.m_bIgnorePhonemes );
}
}
void SND_ClearMouth( channel_t *pChannel )
{
CMouthInfo *m = GetMouthInfoForChannel( pChannel );
if ( !m )
return;
if ( pChannel->sfx )
{
m->RemoveSource( pChannel->sfx->pSource );
}
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : *pChannel -
// Output : Returns true on success, false on failure.
//-----------------------------------------------------------------------------
bool SND_IsMouth( channel_t *pChannel )
{
#ifndef DEDICATED
if ( pChannel->soundsource == SOUND_FROM_UI_PANEL )
return true;
#endif
if ( !entitylist )
{
return false;
}
if ( pChannel->entchannel == CHAN_VOICE || pChannel->entchannel == CHAN_VOICE2 )
{
return true;
}
if ( pChannel->sfx &&
pChannel->sfx->pSource &&
pChannel->sfx->pSource->GetSentence() )
{
return true;
}
return false;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : *pChannel -
// Output : Returns true on success, false on failure.
//-----------------------------------------------------------------------------
bool SND_ShouldPause( channel_t *pChannel )
{
return pChannel->flags.m_bShouldPause;
}
//===============================================================================
// Movie recording support
//===============================================================================
void SND_RecordInit()
{
g_paintedtime = 0;
g_soundtime = 0;
// TMP Wave file supports stereo only, so force stereo
if ( snd_surround.GetInt() != 2 )
{
snd_surround.SetValue( 2 );
}
}
void SND_MovieStart( void )
{
if ( IsX360() )
return;
if ( !cl_movieinfo.IsRecording() )
return;
SND_RecordInit();
// 44k: engine playback rate is now 44100...changed from 22050
if ( cl_movieinfo.DoWav() )
{
WaveCreateTmpFile( cl_movieinfo.moviename, SOUND_DMA_SPEED, 16, 2 );
}
}
void SND_MovieEnd( void )
{
if ( IsX360() )
return;
if ( !cl_movieinfo.IsRecording() )
{
return;
}
if ( cl_movieinfo.DoWav() )
{
WaveFixupTmpFile( cl_movieinfo.moviename );
}
}
bool SND_IsRecording()
{
return ( ( IsReplayRendering() || cl_movieinfo.IsRecording() ) && !Con_IsVisible() );
}
extern IVideoRecorder *g_pVideoRecorder;
void SND_RecordBuffer( void )
{
if ( IsX360() )
return;
if ( !SND_IsRecording() )
return;
int i;
int val;
int bufferSize = snd_linear_count * sizeof(short);
short *tmp = (short *)_alloca( bufferSize );
for (i=0 ; i<snd_linear_count ; i+=2)
{
val = (snd_p[i]*snd_vol)>>8;
tmp[i] = CLIP(val);
val = (snd_p[i+1]*snd_vol)>>8;
tmp[i+1] = CLIP(val);
}
if ( IsReplayRendering() )
{
#if defined( REPLAY_ENABLED )
extern IClientReplayContext *g_pClientReplayContext;
IReplayMovieRenderer *pMovieRenderer = g_pClientReplayContext->GetMovieRenderer();
if ( IsReplayRendering() && pMovieRenderer && pMovieRenderer->IsAudioSyncFrame() )
{
pMovieRenderer->RenderAudio( (unsigned char *)tmp, bufferSize, snd_linear_count );
}
#endif
}
else
{
if ( cl_movieinfo.DoWav() )
{
WaveAppendTmpFile( cl_movieinfo.moviename, tmp, 16, snd_linear_count );
}
if ( cl_movieinfo.DoVideoSound() )
{
g_pVideoRecorder->AppendAudioSamples( tmp, bufferSize );
}
}
}
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