Modified source engine (2017) developed by valve and leaked in 2020. Not for commercial purporses
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//========= Copyright Valve Corporation, All rights reserved. ============//
//
// Purpose: Main control for any streaming sound output device.
//
//===========================================================================//
#include "audio_pch.h"
#include "const.h"
#include "cdll_int.h"
#include "client_class.h"
#include "icliententitylist.h"
#include "tier0/vcrmode.h"
#include "con_nprint.h"
#include "tier0/icommandline.h"
#include "vox_private.h"
#include "traceinit.h"
#include "cmd.h"
#include "toolframework/itoolframework.h"
#include "vstdlib/random.h"
#include "vstdlib/jobthread.h"
#include "vaudio/ivaudio.h"
#include "client.h"
#include "cl_main.h"
#include "utldict.h"
#include "mempool.h"
#include "enginetrace.h" // for traceline
#include "bspflags.h" // for traceline
#include "gametrace.h" // for traceline
#include "vphysics_interface.h" // for surface props
#include "ispatialpartitioninternal.h" // for entity enumerator
#include "debugoverlay.h"
#include "icliententity.h"
#include "cmodel_engine.h"
#include "staticpropmgr.h"
#include "server.h"
#include "edict.h"
#include "pure_server.h"
#include "filesystem/IQueuedLoader.h"
#include "voice.h"
#if defined( _X360 )
#include "xbox/xbox_console.h"
#include "xmp.h"
#endif
#include "replay/iclientreplaycontext.h"
#include "replay/ireplaymovierenderer.h"
#include "video/ivideoservices.h"
extern IVideoServices *g_pVideo;
/*
#include "gl_model_private.h"
#include "world.h"
#include "vphysics_interface.h"
#include "client_class.h"
#include "server_class.h"
*/
// memdbgon must be the last include file in a .cpp file!!!
#include "tier0/memdbgon.h"
///////////////////////////////////
// DEBUGGING
//
// Turn this on to print channel output msgs.
//
//#define DEBUG_CHANNELS
#define SNDLVL_TO_DIST_MULT( sndlvl ) ( sndlvl ? ((pow( 10.0f, snd_refdb.GetFloat() / 20 ) / pow( 10.0f, (float)sndlvl / 20 )) / snd_refdist.GetFloat()) : 0 )
#define DIST_MULT_TO_SNDLVL( dist_mult ) (soundlevel_t)(int)( dist_mult ? ( 20 * log10( pow( 10.0f, snd_refdb.GetFloat() / 20 ) / (dist_mult * snd_refdist.GetFloat()) ) ) : 0 )
extern ConVar dsp_spatial;
extern IPhysicsSurfaceProps *physprop;
extern bool IsReplayRendering();
static void S_Play( const CCommand &args );
static void S_PlayVol( const CCommand &args );
void S_SoundList(void);
static void S_Say ( const CCommand &args );
void S_Update_(float);
void S_StopAllSounds(bool clear);
void S_StopAllSoundsC(void);
void S_ShutdownMixThread();
const char *GetClientClassname( SoundSource soundsource );
float SND_GetGainObscured( channel_t *ch, bool fplayersound, bool flooping, bool bAttenuated );
void DSP_ChangePresetValue( int idsp, int channel, int iproc, float value );
bool DSP_CheckDspAutoEnabled( void );
void DSP_SetDspAuto( int dsp_preset );
float dB_To_Radius ( float db );
int dsp_room_GetInt ( void );
bool MXR_LoadAllSoundMixers( void );
void MXR_ReleaseMemory( void );
int MXR_GetMixGroupListFromDirName( const char *pDirname, byte *pList, int listMax );
void MXR_GetMixGroupFromSoundsource( channel_t *pchan, SoundSource soundsource, soundlevel_t soundlevel);
float MXR_GetVolFromMixGroup( int rgmixgroupid[8], int *plast_mixgroupid );
char *MXR_GetGroupnameFromId( int mixgroupid );
int MXR_GetMixgroupFromName( const char *pszgroupname );
void MXR_DebugShowMixVolumes( void );
#ifdef _DEBUG
static void MXR_DebugSetMixGroupVolume( const CCommand &args );
#endif //_DEBUG
void MXR_UpdateAllDuckerVolumes( void );
void ChannelSetVolTargets( channel_t *pch, int *pvolumes, int ivol_offset, int cvol );
void ChannelUpdateVolXfade( channel_t *pch );
void ChannelClearVolumes( channel_t *pch );
float VOX_GetChanVol(channel_t *ch);
void ConvertListenerVectorTo2D( Vector *pvforward, Vector *pvright );
int ChannelGetMaxVol( channel_t *pch );
// Forceably ends voice tweak mode (only occurs during snd_restart
void VoiceTweak_EndVoiceTweakMode();
bool VoiceTweak_IsStillTweaking();
// Only does anything for voice tweak channel so if view entity changes it doesn't fade out to zero volume
void Voice_Spatialize( channel_t *channel );
// =======================================================================
// Internal sound data & structures
// =======================================================================
channel_t channels[MAX_CHANNELS];
int total_channels;
CActiveChannels g_ActiveChannels;
static double g_LastSoundFrame = 0.0f; // last full frame of sound
static double g_LastMixTime = 0.0f; // last time we did mixing
static float g_EstFrameTime = 0.1f; // estimated frame time running average
// x360 override to fade out game music when the user is playing music through the dashboard
static float g_DashboardMusicMixValue = 1.0f;
static float g_DashboardMusicMixTarget = 1.0f;
const float g_DashboardMusicFadeRate = 0.5f; // Fades one half full-scale volume per second (two seconds for complete fadeout)
// sound mixers
int g_csoundmixers = 0; // total number of soundmixers found
int g_cgrouprules = 0; // total number of group rules found
int g_cgroupclass = 0;
// this is used to enable/disable music playback on x360 when the user selects his own soundtrack to play
void S_EnableMusic( bool bEnable )
{
if ( bEnable )
{
g_DashboardMusicMixTarget = 1.0f;
}
else
{
g_DashboardMusicMixTarget = 0.0f;
}
}
bool IsSoundSourceLocalPlayer( int soundsource )
{
if ( soundsource == SOUND_FROM_UI_PANEL )
return true;
return ( soundsource == g_pSoundServices->GetViewEntity() );
}
CThreadMutex g_SndMutex;
#define THREAD_LOCK_SOUND() AUTO_LOCK( g_SndMutex )
const int MASK_BLOCK_AUDIO = CONTENTS_SOLID|CONTENTS_MOVEABLE|CONTENTS_WINDOW;
void CActiveChannels::Add( channel_t *pChannel )
{
Assert( pChannel->activeIndex == 0 );
m_list[m_count] = pChannel - channels;
m_count++;
pChannel->activeIndex = m_count;
}
void CActiveChannels::Remove( channel_t *pChannel )
{
if ( pChannel->activeIndex == 0 )
return;
int activeIndex = pChannel->activeIndex - 1;
Assert( activeIndex >= 0 && activeIndex < m_count );
Assert( pChannel == &channels[m_list[activeIndex]] );
m_count--;
// Not the last one? Swap the last one with this one and fix its index
if ( activeIndex < m_count )
{
m_list[activeIndex] = m_list[m_count];
channels[m_list[activeIndex]].activeIndex = activeIndex+1;
}
pChannel->activeIndex = 0;
}
void CActiveChannels::GetActiveChannels( CChannelList &list )
{
list.m_count = m_count;
if ( m_count )
{
Q_memcpy( list.m_list, m_list, sizeof(m_list[0])*m_count );
}
for ( int i = SOUND_BUFFER_SPECIAL_START; i < g_paintBuffers.Count(); ++i )
{
paintbuffer_t *pSpecialBuffer = MIX_GetPPaintFromIPaint( i );
if ( pSpecialBuffer->nSpecialDSP != 0 )
{
list.m_nSpecialDSPs.AddToTail( pSpecialBuffer->nSpecialDSP );
}
}
list.m_hasSpeakerChannels = true;
list.m_has11kChannels = true;
list.m_has22kChannels = true;
list.m_has44kChannels = true;
list.m_hasDryChannels = true;
}
void CActiveChannels::Init()
{
m_count = 0;
}
bool snd_initialized = false;
Vector listener_origin;
static Vector listener_forward;
Vector listener_right;
static Vector listener_up;
static bool s_bIsListenerUnderwater;
static vec_t sound_nominal_clip_dist=SOUND_NORMAL_CLIP_DIST;
// @TODO (toml 05-08-02): put this somewhere more reasonable
vec_t S_GetNominalClipDist()
{
return sound_nominal_clip_dist;
}
int g_soundtime = 0; // sample PAIRS output since start
int g_paintedtime = 0; // sample PAIRS mixed since start
float g_ReplaySoundTimeFracAccumulator = 0.0f; // Used by replay
float g_ClockSyncArray[NUM_CLOCK_SYNCS] = {0};
int g_SoundClockPaintTime[NUM_CLOCK_SYNCS] = {0};
// default 10ms
ConVar snd_delay_sound_shift("snd_delay_sound_shift","0.01");
// this forces the clock to resync on the next delayed/sync sound
void S_SyncClockAdjust( clocksync_index_t syncIndex )
{
g_ClockSyncArray[syncIndex] = 0;
g_SoundClockPaintTime[syncIndex] = 0;
}
float S_ComputeDelayForSoundtime( float soundtime, clocksync_index_t syncIndex )
{
// reset clock and return 0
if ( g_ClockSyncArray[syncIndex] == 0 )
{
// Put the current time marker one tick back to impose a minimum delay on the first sample
// this shifts the drift over so the sounds are more likely to delay (rather than skip)
// over the burst
// NOTE: The first sound after a sync MUST have a non-zero delay for the delay channel
// detection logic to work (otherwise we keep resetting the clock)
g_ClockSyncArray[syncIndex] = soundtime - host_state.interval_per_tick;
g_SoundClockPaintTime[syncIndex] = g_paintedtime;
}
// how much time has passed in the game since we did a clock sync?
float gameDeltaTime = soundtime - g_ClockSyncArray[syncIndex];
// how many samples have been mixed since we did a clock sync?
int paintedSamples = g_paintedtime - g_SoundClockPaintTime[syncIndex];
int dmaSpeed = g_AudioDevice->DeviceDmaSpeed();
int gameSamples = (gameDeltaTime * dmaSpeed);
int delaySamples = gameSamples - paintedSamples;
float delay = delaySamples / float(dmaSpeed);
if ( gameDeltaTime < 0 || fabs(delay) > 0.500f )
{
// Note that the equations assume a correlation between game time and real time
// some kind of clock error. This can happen with large host_timescale or when the
// framerate hitches drastically (game time is a smaller clamped value wrt real time).
// The current sync estimate has probably drifted due to this or some other problem, recompute.
//Msg("Clock ERROR!: %.2f %.2f\n", gameDeltaTime, delay);
S_SyncClockAdjust(syncIndex);
return 0;
}
return delay + snd_delay_sound_shift.GetFloat();
}
static int s_buffers = 0;
static int s_oldsampleOutCount = 0;
static float s_lastsoundtime = 0.0f;
bool s_bOnLoadScreen = false;
static CClassMemoryPool< CSfxTable > s_SoundPool( MAX_SFX );
struct SfxDictEntry
{
CSfxTable *pSfx;
};
static CUtlMap< FileNameHandle_t, SfxDictEntry > s_Sounds( 0, 0, DefLessFunc( FileNameHandle_t ) );
class CDummySfx : public CSfxTable
{
public:
virtual const char *getname()
{
return name;
}
void setname( const char *pName )
{
Q_strncpy( name, pName, sizeof( name ) );
OnNameChanged(name);
}
private:
char name[MAX_PATH];
};
static CDummySfx dummySfx;
// returns true if ok to procede with TraceRay calls
bool SND_IsInGame( void )
{
return cl.IsActive();
}
CSfxTable::CSfxTable()
{
m_namePoolIndex = s_Sounds.InvalidIndex();
pSource = NULL;
m_bUseErrorFilename = false;
m_bIsUISound = false;
m_bIsLateLoad = false;
m_bMixGroupsCached = false;
m_pDebugName = NULL;
}
void CSfxTable::SetNamePoolIndex( int index )
{
m_namePoolIndex = index;
if ( m_namePoolIndex != s_Sounds.InvalidIndex() )
{
OnNameChanged(getname());
}
#ifdef _DEBUG
m_pDebugName = strdup( getname() );
#endif
}
void CSfxTable::OnNameChanged( const char *pName )
{
if ( pName && g_cgrouprules )
{
char szString[MAX_PATH];
Q_strncpy( szString, pName, sizeof(szString) );
Q_FixSlashes( szString, '/' );
m_mixGroupCount = MXR_GetMixGroupListFromDirName( szString, m_mixGroupList, ARRAYSIZE(m_mixGroupList) );
m_bMixGroupsCached = true;
}
}
//-----------------------------------------------------------------------------
// Purpose: Wrapper for sfxtable->getname()
// Output : char const
//-----------------------------------------------------------------------------
const char *CSfxTable::getname()
{
if ( s_Sounds.InvalidIndex() != m_namePoolIndex )
{
char* pString = tmpstr512();
if ( g_pFileSystem )
g_pFileSystem->String( s_Sounds.Key( m_namePoolIndex ), pString, 512 );
else
{
pString[0] = 0;
}
return pString;
}
return NULL;
}
FileNameHandle_t CSfxTable::GetFileNameHandle()
{
if ( s_Sounds.InvalidIndex() != m_namePoolIndex )
{
return s_Sounds.Key( m_namePoolIndex );
}
return NULL;
}
const char *CSfxTable::GetFileName()
{
if ( IsX360() && m_bUseErrorFilename )
{
// Redirecting error sounds to a valid empty wave, prevents a bad loading retry pattern during gameplay
// which may event sounds skipped by preload, because they don't exist.
return "common/null.wav";
}
const char *pName = getname();
return pName ? PSkipSoundChars( pName ) : NULL;
}
bool CSfxTable::IsPrecachedSound()
{
const char *pName = getname();
if ( sv.IsActive() )
{
// Server uses zero to mark invalid sounds
return sv.LookupSoundIndex( pName ) != 0 ? true : false;
}
// Client uses -1
// WE SHOULD FIX THIS!!!
return ( cl.LookupSoundIndex( pName ) != -1 ) ? true : false;
}
float g_DuckScale = 1.0f;
// Structure used for fading in and out client sound volume.
typedef struct
{
float initial_percent;
// How far to adjust client's volume down by.
float percent;
// GetHostTime() when we started adjusting volume
float starttime;
// # of seconds to get to faded out state
float fadeouttime;
// # of seconds to hold
float holdtime;
// # of seconds to restore
float fadeintime;
} soundfade_t;
static soundfade_t soundfade; // Client sound fading singleton object
// 0)headphones 2)stereo speakers 4)quad 5)5point1
// autodetected from windows settings
ConVar snd_surround( "snd_surround_speakers", "-1", FCVAR_INTERNAL_USE );
ConVar snd_legacy_surround( "snd_legacy_surround", "0", FCVAR_ARCHIVE );
ConVar snd_noextraupdate( "snd_noextraupdate", "0" );
ConVar snd_show( "snd_show", "0", FCVAR_CHEAT, "Show sounds info" );
ConVar snd_visualize ("snd_visualize", "0", FCVAR_CHEAT, "Show sounds location in world" );
ConVar snd_pitchquality( "snd_pitchquality", "1", FCVAR_ARCHIVE ); // 1) use high quality pitch shifters
// master volume
static ConVar volume( "volume", "1.0", FCVAR_ARCHIVE | FCVAR_ARCHIVE_XBOX, "Sound volume", true, 0.0f, true, 1.0f );
// user configurable music volume
ConVar snd_musicvolume( "snd_musicvolume", "1.0", FCVAR_ARCHIVE | FCVAR_ARCHIVE_XBOX, "Music volume", true, 0.0f, true, 1.0f );
ConVar snd_mixahead( "snd_mixahead", "0.1", FCVAR_ARCHIVE );
ConVar snd_mix_async( "snd_mix_async", "0" );
#ifdef _DEBUG
static ConCommand snd_mixvol("snd_mixvol", MXR_DebugSetMixGroupVolume, "Set named Mixgroup to mix volume.");
#endif
// vaudio DLL
IVAudio *vaudio = NULL;
CSysModule *g_pVAudioModule = NULL;
//-----------------------------------------------------------------------------
// Resource loading for sound
//-----------------------------------------------------------------------------
class CResourcePreloadSound : public CResourcePreload
{
public:
CResourcePreloadSound() : m_SoundNames( 0, 0, true )
{
}
virtual bool CreateResource( const char *pName )
{
CSfxTable *pSfx = S_PrecacheSound( pName );
if ( !pSfx )
{
return false;
}
m_SoundNames.AddString( pSfx->GetFileName() );
return true;
}
virtual void PurgeUnreferencedResources()
{
bool bSpew = ( g_pQueuedLoader->GetSpewDetail() & LOADER_DETAIL_PURGES ) != 0;
for ( int i = s_Sounds.FirstInorder(); i != s_Sounds.InvalidIndex(); i = s_Sounds.NextInorder( i ) )
{
// the master sound table grows forever
// remove sound sources from the master sound table that were not in the preload list
CSfxTable *pSfx = s_Sounds[i].pSfx;
if ( pSfx && pSfx->pSource )
{
if ( pSfx->m_bIsUISound )
{
// never purge ui
continue;
}
UtlSymId_t symbol = m_SoundNames.Find( pSfx->GetFileName() );
if ( symbol == UTL_INVAL_SYMBOL )
{
// sound was not part of preload, purge it
if ( bSpew )
{
Msg( "CResourcePreloadSound: Purging: %s\n", pSfx->GetFileName() );
}
pSfx->pSource->CacheUnload();
delete pSfx->pSource;
pSfx->pSource = NULL;
}
}
}
m_SoundNames.RemoveAll();
if ( !g_pQueuedLoader->IsSameMapLoading() )
{
wavedatacache->Flush();
}
}
virtual void PurgeAll()
{
bool bSpew = ( g_pQueuedLoader->GetSpewDetail() & LOADER_DETAIL_PURGES ) != 0;
for ( int i = s_Sounds.FirstInorder(); i != s_Sounds.InvalidIndex(); i = s_Sounds.NextInorder( i ) )
{
// the master sound table grows forever
// remove sound sources from the master sound table that were not in the preload list
CSfxTable *pSfx = s_Sounds[i].pSfx;
if ( pSfx && pSfx->pSource )
{
if ( pSfx->m_bIsUISound )
{
// never purge ui
if ( bSpew )
{
Msg( "CResourcePreloadSound: Skipping: %s\n", pSfx->GetFileName() );
}
continue;
}
// sound was not part of preload, purge it
if ( bSpew )
{
Msg( "CResourcePreloadSound: Purging: %s\n", pSfx->GetFileName() );
}
pSfx->pSource->CacheUnload();
delete pSfx->pSource;
pSfx->pSource = NULL;
}
}
m_SoundNames.RemoveAll();
wavedatacache->Flush();
}
private:
CUtlSymbolTable m_SoundNames;
};
static CResourcePreloadSound s_ResourcePreloadSound;
//-----------------------------------------------------------------------------
// Purpose:
// Output : float
//-----------------------------------------------------------------------------
float S_GetMasterVolume( void )
{
float scale = 1.0f;
if ( soundfade.percent != 0 )
{
scale = clamp( (float)soundfade.percent / 100.0f, 0.0f, 1.0f );
scale = 1.0f - scale;
}
return volume.GetFloat() * scale;
}
void S_SoundInfo_f(void)
{
if ( !g_AudioDevice->IsActive() )
{
Msg( "Sound system not started\n" );
return;
}
Msg( "Sound Device: %s\n", g_AudioDevice->DeviceName() );
Msg( " Channels: %d\n", g_AudioDevice->DeviceChannels() );
Msg( " Samples: %d\n", g_AudioDevice->DeviceSampleCount() );
Msg( " Bits/Sample: %d\n", g_AudioDevice->DeviceSampleBits() );
Msg( " Rate: %d\n", g_AudioDevice->DeviceDmaSpeed() );
Msg( "total_channels: %d\n", total_channels);
if ( IsX360() )
{
// dump a glimpse of the mixing state
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
Msg( "\nActive Channels: (%d)\n", list.Count() );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = list.GetChannel(i);
Msg( "%s (Mixer: 0x%p)\n", pChannel->sfx->GetFileName(), pChannel->pMixer );
}
}
}
/*
================
S_Startup
================
*/
void S_Startup( void )
{
if ( !snd_initialized )
return;
if ( !g_AudioDevice || g_AudioDevice == Audio_GetNullDevice() )
{
g_AudioDevice = IAudioDevice::AutoDetectInit( CommandLine()->CheckParm( "-wavonly" ) != 0 );
if ( !g_AudioDevice )
{
Error( "Unable to init audio" );
}
}
}
static ConCommand play("play", S_Play, "Play a sound.", FCVAR_SERVER_CAN_EXECUTE );
static ConCommand playflush( "playflush", S_Play, "Play a sound, reloading from disk in case of changes." );
static ConCommand playvol( "playvol", S_PlayVol, "Play a sound at a specified volume." );
static ConCommand speak( "speak", S_Say, "Play a constructed sentence." );
static ConCommand stopsound( "stopsound", S_StopAllSoundsC, 0, FCVAR_CHEAT); // Marked cheat because it gives an advantage to players minimising ambient noise.
static ConCommand soundlist( "soundlist", S_SoundList, "List all known sounds." );
static ConCommand soundinfo( "soundinfo", S_SoundInfo_f, "Describe the current sound device." );
bool IsValidSampleRate( int rate )
{
return rate == SOUND_11k || rate == SOUND_22k || rate == SOUND_44k;
}
void VAudioInit()
{
if ( IsPC() )
{
if ( !IsPosix() )
{
// vaudio_miles.dll will load this...
g_pFileSystem->GetLocalCopy( "mss32.dll" );
}
g_pVAudioModule = FileSystem_LoadModule( "vaudio_minimp3" );
if ( g_pVAudioModule )
{
CreateInterfaceFn vaudioFactory = Sys_GetFactory( g_pVAudioModule );
vaudio = (IVAudio *)vaudioFactory( VAUDIO_INTERFACE_VERSION, NULL );
}
}
}
/*
================
S_Init
================
*/
void S_Init( void )
{
if ( sv.IsDedicated() && !CommandLine()->CheckParm( "-forcesound" ) )
return;
DevMsg( "Sound Initialization: Start\n" );
// KDB: init sentence array
TRACEINIT( VOX_Init(), VOX_Shutdown() );
VAudioInit();
if ( CommandLine()->CheckParm( "-nosound" ) )
{
g_AudioDevice = Audio_GetNullDevice();
TRACEINIT( audiosourcecache->Init( host_parms.memsize >> 2 ), audiosourcecache->Shutdown() );
return;
}
snd_initialized = true;
g_ActiveChannels.Init();
S_Startup();
MIX_InitAllPaintbuffers();
SND_InitScaletable();
MXR_LoadAllSoundMixers();
S_StopAllSounds( true );
TRACEINIT( audiosourcecache->Init( host_parms.memsize >> 2 ), audiosourcecache->Shutdown() );
AllocDsps( true );
if ( IsX360() )
{
g_pQueuedLoader->InstallLoader( RESOURCEPRELOAD_SOUND, &s_ResourcePreloadSound );
}
DevMsg( "Sound Initialization: Finish, Sampling Rate: %i\n", g_AudioDevice->DeviceDmaSpeed() );
#ifdef _X360
BOOL bPlaybackControl;
// get initial state of the x360 media player
if ( XMPTitleHasPlaybackControl( &bPlaybackControl ) == ERROR_SUCCESS )
{
S_EnableMusic(bPlaybackControl!=0);
}
Assert( g_pVideo != NULL );
if ( g_pVideo != NULL )
{
if ( g_pVideo->SoundDeviceCommand( VideoSoundDeviceOperation::HOOK_X_AUDIO, NULL ) != VideoResult::SUCCESS )
{
Assert( 0 );
}
}
#endif
}
// =======================================================================
// Shutdown sound engine
// =======================================================================
void S_Shutdown(void)
{
#if !defined( _X360 )
if ( VoiceTweak_IsStillTweaking() )
{
VoiceTweak_EndVoiceTweakMode();
}
#endif
S_StopAllSounds( true );
S_ShutdownMixThread();
TRACESHUTDOWN( audiosourcecache->Shutdown() );
SNDDMA_Shutdown();
for ( int i = s_Sounds.FirstInorder(); i != s_Sounds.InvalidIndex(); i = s_Sounds.NextInorder( i ) )
{
if ( s_Sounds[i].pSfx )
{
delete s_Sounds[i].pSfx->pSource;
s_Sounds[i].pSfx->pSource = NULL;
}
}
s_Sounds.RemoveAll();
s_SoundPool.Clear();
// release DSP resources
FreeDsps( true );
MXR_ReleaseMemory();
// release sentences resources
TRACESHUTDOWN( VOX_Shutdown() );
if ( IsPC() )
{
// shutdown vaudio
if ( vaudio )
delete vaudio;
FileSystem_UnloadModule( g_pVAudioModule );
g_pVAudioModule = NULL;
vaudio = NULL;
}
MIX_FreeAllPaintbuffers();
snd_initialized = false;
g_paintedtime = 0;
g_soundtime = 0;
g_ReplaySoundTimeFracAccumulator = 0.0f;
s_buffers = 0;
s_oldsampleOutCount = 0;
s_lastsoundtime = 0.0f;
#if !defined( _X360 )
Voice_Deinit();
#endif
}
bool S_IsInitted()
{
return snd_initialized;
}
// =======================================================================
// Load a sound
// =======================================================================
//-----------------------------------------------------------------------------
// Return sfx and set pfInCache to 1 if
// name is in name cache. Otherwise, alloc
// a new spot in name cache and return 0
// in pfInCache.
//-----------------------------------------------------------------------------
CSfxTable *S_FindName( const char *szName, int *pfInCache )
{
int i;
CSfxTable *sfx = NULL;
char szBuff[MAX_PATH];
const char *pName;
if ( !szName )
{
Error( "S_FindName: NULL\n" );
}
pName = szName;
if ( IsX360() )
{
Q_strncpy( szBuff, pName, sizeof( szBuff ) );
int len = Q_strlen( szBuff )-4;
if ( len > 0 && !Q_strnicmp( szBuff+len, ".mp3", 4 ) )
{
// convert unsupported .mp3 to .wav
Q_strcpy( szBuff+len, ".wav" );
}
pName = szBuff;
if ( pName[0] == CHAR_STREAM )
{
// streaming (or not) is hardcoded to alternate criteria
// prevent the same sound from creating disparate instances
pName++;
}
}
// see if already loaded
FileNameHandle_t fnHandle = g_pFileSystem->FindOrAddFileName( pName );
i = s_Sounds.Find( fnHandle );
if ( i != s_Sounds.InvalidIndex() )
{
sfx = s_Sounds[i].pSfx;
Assert( sfx );
if ( pfInCache )
{
// indicate whether or not sound is currently in the cache.
*pfInCache = ( sfx->pSource && sfx->pSource->IsCached() ) ? 1 : 0;
}
return sfx;
}
else
{
SfxDictEntry entry;
entry.pSfx = ( CSfxTable * )s_SoundPool.Alloc();
Assert( entry.pSfx );
i = s_Sounds.Insert( fnHandle, entry );
sfx = s_Sounds[i].pSfx;
sfx->SetNamePoolIndex( i );
sfx->pSource = NULL;
if ( pfInCache )
{
*pfInCache = 0;
}
}
return sfx;
}
//-----------------------------------------------------------------------------
// S_LoadSound
//
// Check to see if wave data is in the cache. If so, return pointer to data.
// If not, allocate cache space for wave data, load wave file into temporary heap
// space, and dump/convert file data into cache.
//-----------------------------------------------------------------------------
double g_flAccumulatedSoundLoadTime = 0.0f;
CAudioSource *S_LoadSound( CSfxTable *pSfx, channel_t *ch )
{
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s", __FUNCTION__ );
VPROF("S_LoadSound");
if ( !pSfx->pSource )
{
if ( IsX360() )
{
if ( SND_IsInGame() && !g_pQueuedLoader->IsMapLoading() )
{
// sound should be present (due to reslists), but NOT allowing a load hitch during gameplay
// loading a sound during gameplay is a bad experience, causes a very expensive sync i/o to fetch the header
// and in the case of a memory wave, the actual audio data
bool bFound = false;
if ( !pSfx->m_bIsLateLoad )
{
if ( pSfx->getname() != PSkipSoundChars( pSfx->getname() ) )
{
// the sound might already exist as an undecorated audio source
FileNameHandle_t fnHandle = g_pFileSystem->FindOrAddFileName( pSfx->GetFileName() );
int i = s_Sounds.Find( fnHandle );
if ( i != s_Sounds.InvalidIndex() )
{
CSfxTable *pOtherSfx = s_Sounds[i].pSfx;
Assert( pOtherSfx );
CAudioSource *pOtherSource = pOtherSfx->pSource;
if ( pOtherSource && pOtherSource->IsCached() )
{
// Can safely let the "load" continue because the headers are expected to be in the preload
// that are now persisted and the wave data cache will find an existing audio buffer match,
// so no sync i/o should occur from either.
bFound = true;
}
}
}
if ( !bFound )
{
// warn once
DevWarning( "S_LoadSound: Late load '%s', skipping.\n", pSfx->getname() );
pSfx->m_bIsLateLoad = true;
}
}
if ( !bFound )
{
return NULL;
}
}
else if ( pSfx->m_bIsLateLoad )
{
// outside of gameplay, let the load happen
pSfx->m_bIsLateLoad = false;
}
}
double st = Plat_FloatTime();
bool bStream = false;
bool bUserVox = false;
// sound chars can explicitly categorize usage
bStream = TestSoundChar( pSfx->getname(), CHAR_STREAM );
if ( !bStream )
{
bUserVox = TestSoundChar( pSfx->getname(), CHAR_USERVOX );
}
// override streaming
if ( IsX360() )
{
const char *s_CriticalSounds[] =
{
"common",
"items",
"physics",
"player",
"ui",
"weapons",
};
// stream everything but critical sounds
bStream = true;
const char *pFileName = pSfx->GetFileName();
for ( int i = 0; i<ARRAYSIZE( s_CriticalSounds ); i++ )
{
size_t len = strlen( s_CriticalSounds[i] );
if ( !Q_strnicmp( pFileName, s_CriticalSounds[i], len ) && ( pFileName[len] == '\\' || pFileName[len] == '/' ) )
{
// never stream these, regardless of sound chars
bStream = false;
break;
}
}
}
if ( bStream )
{
// setup as a streaming resource
pSfx->pSource = Audio_CreateStreamedWave( pSfx );
}
else
{
if ( bUserVox )
{
if ( !IsX360() )
{
pSfx->pSource = Voice_SetupAudioSource( ch->soundsource, ch->entchannel );
}
else
{
// not supporting
Assert( 0 );
}
}
else
{
// load all into memory directly
pSfx->pSource = Audio_CreateMemoryWave( pSfx );
}
}
double ed = Plat_FloatTime();
g_flAccumulatedSoundLoadTime += ( ed - st );
}
else
{
pSfx->pSource->CheckAudioSourceCache();
}
if ( !pSfx->pSource )
{
return NULL;
}
// first time to load? Create the mixer
if ( ch && !ch->pMixer )
{
ch->pMixer = pSfx->pSource->CreateMixer( ch->initialStreamPosition );
if ( !ch->pMixer )
{
return NULL;
}
}
return pSfx->pSource;
}
//-----------------------------------------------------------------------------
// S_PrecacheSound
//
// Reserve space for the name of the sound in a global array.
// Load the data for the non-streaming sound. Streaming sounds
// defer loading of data until just before playback.
//-----------------------------------------------------------------------------
CSfxTable *S_PrecacheSound( const char *name )
{
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s", __FUNCTION__ );
if ( !g_AudioDevice )
return NULL;
if ( !g_AudioDevice->IsActive() )
return NULL;
CSfxTable *sfx = S_FindName( name, NULL );
if ( sfx )
{
// cache sound
S_LoadSound( sfx, NULL );
}
else
{
Assert( !"S_PrecacheSound: Failed to create sfx" );
}
return sfx;
}
void S_InternalReloadSound( CSfxTable *sfx )
{
if ( !sfx || !sfx->pSource )
return;
sfx->pSource->CacheUnload();
delete sfx->pSource;
sfx->pSource = NULL;
char pExt[10];
Q_ExtractFileExtension( sfx->getname(), pExt, sizeof(pExt) );
int nSource = !Q_stricmp( pExt, "mp3" ) ? CAudioSource::AUDIO_SOURCE_MP3 : CAudioSource::AUDIO_SOURCE_WAV;
// audiosourcecache->RebuildCacheEntry( nSource, sfx->IsPrecachedSound(), sfx );
audiosourcecache->GetInfo( nSource, sfx->IsPrecachedSound(), sfx ); // Do a size/date check and rebuild the cache entry if necessary.
}
//-----------------------------------------------------------------------------
// Refresh a sound in the cache
//-----------------------------------------------------------------------------
void S_ReloadSound( const char *name )
{
if ( IsX360() )
{
// not supporting
Assert( 0 );
return;
}
if ( !g_AudioDevice )
return;
if ( !g_AudioDevice->IsActive() )
return;
CSfxTable *sfx = S_FindName( name, NULL );
#ifdef _DEBUG
if ( sfx )
{
Assert( Q_stricmp( sfx->getname(), name ) == 0 );
}
#endif
S_InternalReloadSound( sfx );
}
// See comments on CL_HandlePureServerWhitelist for details of what we're doing here.
void S_ReloadFilesInList( IFileList *pFilesToReload )
{
if ( !IsPC() )
return;
S_StopAllSounds( true );
wavedatacache->Flush();
audiosourcecache->ForceRecheckDiskInfo(); // Force all cached audio data to recheck size/date info next time it's accessed.
CUtlVector< CSfxTable * > processed;
int iLast = s_Sounds.LastInorder();
for ( int i = s_Sounds.FirstInorder(); i != iLast; i = s_Sounds.NextInorder( i ) )
{
FileNameHandle_t fnHandle = s_Sounds.Key( i );
char filename[MAX_PATH * 3];
if ( !g_pFileSystem->String( fnHandle, filename, sizeof( filename ) ) )
{
Assert( !"S_HandlePureServerWhitelist - can't get a filename." );
continue;
}
// If the file isn't cached in yet, then the filesystem hasn't touched its file, so don't bother.
CSfxTable *sfx = s_Sounds[i].pSfx;
if ( sfx && ( processed.Find( sfx ) == processed.InvalidIndex() ) )
{
char fullFilename[MAX_PATH*2];
if ( IsSoundChar( filename[0] ) )
Q_snprintf( fullFilename, sizeof( fullFilename ), "sound/%s", &filename[1] );
else
Q_snprintf( fullFilename, sizeof( fullFilename ), "sound/%s", filename );
if ( !pFilesToReload->IsFileInList( fullFilename ) )
continue;
processed.AddToTail( sfx );
S_InternalReloadSound( sfx );
}
}
}
//-----------------------------------------------------------------------------
// Unfortunate confusing terminology.
// Here prefetching means hinting to the audio source (which may be a stream)
// to get its async data in flight.
//-----------------------------------------------------------------------------
void S_PrefetchSound( char const *name, bool bPlayOnce )
{
CSfxTable *sfx;
if ( !g_AudioDevice )
return;
if ( !g_AudioDevice->IsActive() )
return;
sfx = S_FindName( name, NULL );
if ( sfx )
{
// cache sound
S_LoadSound( sfx, NULL );
}
if ( !sfx || !sfx->pSource )
{
return;
}
// hint the sound to start loading
sfx->pSource->Prefetch();
if ( bPlayOnce )
{
sfx->pSource->SetPlayOnce( true );
}
}
void S_MarkUISound( CSfxTable *pSfx )
{
pSfx->m_bIsUISound = true;
}
unsigned int RemainingSamples( channel_t *pChannel )
{
if ( !pChannel || !pChannel->sfx || !pChannel->sfx->pSource )
return 0;
unsigned int timeleft = pChannel->sfx->pSource->SampleCount();
if ( pChannel->sfx->pSource->IsLooped() )
{
return pChannel->sfx->pSource->SampleRate();
}
if ( pChannel->pMixer )
{
timeleft -= pChannel->pMixer->GetSamplePosition();
}
return timeleft;
}
// chooses the voice stealing algorithm
ConVar voice_steal("voice_steal", "2");
/*
=================
SND_StealDynamicChannel
Select a channel from the dynamic channel allocation area. For the given entity,
override any other sound playing on the same channel (see code comments below for
exceptions).
=================
*/
channel_t *SND_StealDynamicChannel(SoundSource soundsource, int entchannel, const Vector &origin, CSfxTable *sfx, float flDelay, bool bDoNotOverwriteExisting)
{
int canSteal[MAX_DYNAMIC_CHANNELS];
int canStealCount = 0;
int sameSoundCount = 0;
unsigned int sameSoundRemaining = 0xFFFFFFFF;
int sameSoundIndex = -1;
int sameVol = 0xFFFF;
int availableChannel = -1;
bool bDelaySame = false;
int nExactMatch[MAX_DYNAMIC_CHANNELS];
int nExactCount = 0;
// first pass to replace sounds on same ent/channel, and search for free or stealable channels otherwise
for ( int ch_idx = 0; ch_idx < MAX_DYNAMIC_CHANNELS ; ch_idx++)
{
channel_t *ch = &channels[ch_idx];
if ( ch->activeIndex )
{
// channel CHAN_AUTO never overrides sounds on same channel
if ( entchannel != CHAN_AUTO )
{
int checkChannel = entchannel;
if ( checkChannel == -1 )
{
if ( ch->entchannel != CHAN_STREAM && ch->entchannel != CHAN_VOICE && ch->entchannel != CHAN_VOICE2 )
{
checkChannel = ch->entchannel;
}
}
if ( ch->soundsource == soundsource && (soundsource != -1) && ch->entchannel == checkChannel )
{
// we found an exact match for this entity and this channel, but the sound we want to play is considered
// low priority so instead of stomping this entry pretend we couldn't find a free slot to play and let
// the existing sound keep going
if ( bDoNotOverwriteExisting )
return NULL;
if ( ch->flags.delayed_start )
{
nExactMatch[nExactCount] = ch_idx;
nExactCount++;
continue;
}
return ch; // always override sound from same entity
}
}
// Never steal the channel of a streaming sound that is currently playing or
// voice over IP data that is playing or any sound on CHAN_VOICE( acting )
if ( ch->entchannel == CHAN_STREAM || ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE2 )
continue;
// don't let monster sounds override player sounds
if ( g_pSoundServices->IsPlayer( ch->soundsource ) && !g_pSoundServices->IsPlayer(soundsource) )
continue;
if ( ch->sfx == sfx )
{
bDelaySame = ch->flags.delayed_start ? true : bDelaySame;
sameSoundCount++;
int maxVolume = ChannelGetMaxVol( ch );
unsigned int remaining = RemainingSamples(ch);
if ( maxVolume < sameVol || (maxVolume == sameVol && remaining < sameSoundRemaining) )
{
sameSoundIndex = ch_idx;
sameVol = maxVolume;
sameSoundRemaining = remaining;
}
}
canSteal[canStealCount++] = ch_idx;
}
else
{
if ( availableChannel < 0 )
{
availableChannel = ch_idx;
}
}
}
// coalesce the timeline for this channel
if ( nExactCount > 0 )
{
uint nFreeSampleTime = g_paintedtime + (flDelay * SOUND_DMA_SPEED);
channel_t *pReturn = &channels[nExactMatch[0]];
uint nMinRemaining = RemainingSamples( pReturn );
if ( pReturn->nFreeChannelAtSampleTime == 0 || pReturn->nFreeChannelAtSampleTime > nFreeSampleTime )
{
pReturn->nFreeChannelAtSampleTime = nFreeSampleTime;
}
for ( int i = 1; i < nExactCount; i++ )
{
channel_t *pChannel = &channels[nExactMatch[i]];
if ( pChannel->nFreeChannelAtSampleTime == 0 || pChannel->nFreeChannelAtSampleTime > nFreeSampleTime )
{
pChannel->nFreeChannelAtSampleTime = nFreeSampleTime;
}
uint nRemain = RemainingSamples( pChannel );
if ( nRemain < nMinRemaining )
{
pReturn = pChannel;
nMinRemaining = nRemain;
}
}
// if there's only one, mark it to be freed but don't reuse it.
// otherwise mark all others to be freed and use the closest one to being done
if ( nExactCount > 1 )
{
return pReturn;
}
}
// Limit the number of times a given sfx/wave can play simultaneously
if ( voice_steal.GetInt() > 1 && sameSoundIndex >= 0 )
{
// if sounds of this type are normally delayed, then add an extra slot for stealing
// NOTE: In HL2 these are usually NPC gunshot sounds - and stealing too soon will cut
// them off early. This is a safe heuristic to avoid that problem. There's probably a better
// long-term solution involving only counting channels that are actually going to play (delay included)
// at the same time as this one.
int maxSameSounds = bDelaySame ? 5 : 4;
float distSqr = 0.0f;
if ( sfx->pSource )
{
distSqr = origin.DistToSqr(listener_origin);
if ( sfx->pSource->IsLooped() )
{
maxSameSounds = 3;
}
}
// don't play more than N copies of the same sound, steal the quietest & closest one otherwise
if ( sameSoundCount >= maxSameSounds )
{
channel_t *ch = &channels[sameSoundIndex];
// you're already playing a closer version of this sound, don't steal
if ( distSqr > 0.0f && ch->origin.DistToSqr(listener_origin) < distSqr && entchannel != CHAN_WEAPON )
return NULL;
//Msg("Sound playing %d copies, stole %s (%d)\n", sameSoundCount, ch->sfx->getname(), sameVol );
return ch;
}
}
// if there's a free channel, just take that one - don't steal
if ( availableChannel >= 0 )
return &channels[availableChannel];
// Still haven't found a suitable channel, so choose the one with the least amount of time left to play
float life_left = FLT_MAX;
int first_to_die = -1;
bool bAllowVoiceSteal = voice_steal.GetBool();
for ( int i = 0; i < canStealCount; i++ )
{
int ch_idx = canSteal[i];
channel_t *ch = &channels[ch_idx];
float timeleft = 0;
if ( bAllowVoiceSteal )
{
int maxVolume = ChannelGetMaxVol( ch );
if ( maxVolume < 5 )
{
//Msg("Sound quiet, stole %s for %s\n", ch->sfx->getname(), sfx->getname() );
return ch;
}
if ( ch->sfx && ch->sfx->pSource )
{
unsigned int sampleCount = RemainingSamples( ch );
timeleft = (float)sampleCount / (float)ch->sfx->pSource->SampleRate();
}
}
else
{
// UNDONE: Kill this when voice_steal 0,1,2 has been tested
// UNDONE: This is the old buggy code that we're trying to replace
if ( ch->sfx )
{
// basically steals the first one you come to
timeleft = 1; //ch->end - paintedtime
}
}
if ( timeleft < life_left )
{
life_left = timeleft;
first_to_die = ch_idx;
}
}
if ( first_to_die >= 0 )
{
//Msg("Stole %s, timeleft %d\n", channels[first_to_die].sfx->getname(), life_left );
return &channels[first_to_die];
}
return NULL;
}
channel_t *SND_PickDynamicChannel(SoundSource soundsource, int entchannel, const Vector &origin, CSfxTable *sfx, float flDelay, bool bDoNotOverwriteExisting)
{
channel_t *pChannel = SND_StealDynamicChannel( soundsource, entchannel, origin, sfx, flDelay, bDoNotOverwriteExisting );
if ( !pChannel )
return NULL;
if (pChannel->sfx)
{
// Don't restart looping sounds for the same entity
CAudioSource *pSource = pChannel->sfx->pSource;
if ( pSource )
{
if ( pSource->IsLooped() )
{
if ( pChannel->soundsource == soundsource && pChannel->entchannel == entchannel && pChannel->sfx == sfx )
{
// same looping sound, same ent, same channel, don't restart the sound
return NULL;
}
}
}
// be sure and release previous channel
// if sentence.
// ("Stealing channel from %s\n", channels[first_to_die].sfx->getname() );
S_FreeChannel(pChannel);
}
return pChannel;
}
/*
=====================
SND_PickStaticChannel
=====================
Pick an empty channel from the static sound area, or allocate a new
channel. Only fails if we're at max_channels (128!!!) or if
we're trying to allocate a channel for a stream sound that is
already playing.
*/
channel_t *SND_PickStaticChannel(int soundsource, CSfxTable *pSfx)
{
int i;
channel_t *ch = NULL;
// Check for replacement sound, or find the best one to replace
for (i = MAX_DYNAMIC_CHANNELS; i<total_channels; i++)
if (channels[i].sfx == NULL)
break;
if (i < total_channels)
{
// reuse an empty static sound channel
ch = &channels[i];
}
else
{
// no empty slots, alloc a new static sound channel
if (total_channels == MAX_CHANNELS)
{
DevMsg ("total_channels == MAX_CHANNELS\n");
return NULL;
}
// get a channel for the static sound
ch = &channels[total_channels];
total_channels++;
}
return ch;
}
void S_SpatializeChannel( int pVolume[CCHANVOLUMES/2], int master_vol, const Vector *psourceDir, float gain, float mono )
{
float lscale, rscale, scale;
vec_t dotRight;
Vector sourceDir = *psourceDir;
dotRight = DotProduct(listener_right, sourceDir);
// clear volumes
for (int i = 0; i < CCHANVOLUMES/2; i++)
pVolume[i] = 0;
if (mono > 0.0)
{
// sound has radius, within which spatialization becomes mono:
// mono is 0.0 -> 1.0, from radius 100% to radius 50%
// at radius * 0.5, dotRight is 0 (ie: sound centered left/right)
// at radius * 1.0, dotRight == dotRight
dotRight *= (1.0 - mono);
}
rscale = 1.0 + dotRight;
lscale = 1.0 - dotRight;
// add in distance effect
scale = gain * rscale / 2;
pVolume[IFRONT_RIGHT] = (int) (master_vol * scale);
scale = gain * lscale / 2;
pVolume[IFRONT_LEFT] = (int) (master_vol * scale);
pVolume[IFRONT_RIGHT] = clamp( pVolume[IFRONT_RIGHT], 0, 255 );
pVolume[IFRONT_LEFT] = clamp( pVolume[IFRONT_LEFT], 0, 255 );
}
bool S_IsMusic( channel_t *pChannel )
{
if ( !pChannel->flags.bdry )
return false;
CSfxTable *sfx = pChannel->sfx;
if ( !sfx )
return false;
CAudioSource *source = sfx->pSource;
if ( !source )
return false;
// Don't save restore looping sounds as you can end up with an entity restarting them again and have
// them accumulate, etc.
if ( source->IsLooped() )
return false;
CAudioMixer *pMixer = pChannel->pMixer;
if ( !pMixer )
return false;
for ( int i = 0; i < 8; i++ )
{
if ( pChannel->mixgroups[i] != -1 )
{
char *pGroupName = MXR_GetGroupnameFromId( pChannel->mixgroups[i] );
if ( !Q_strcmp( pGroupName, "Music" ) )
{
return true;
}
}
}
return false;
}
//-----------------------------------------------------------------------------
// Purpose: For save/restore of currently playing music
// Input : list -
//-----------------------------------------------------------------------------
void S_GetCurrentlyPlayingMusic( CUtlVector< musicsave_t >& musiclist )
{
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = &channels[list.GetChannelIndex(i)];
if ( !S_IsMusic( pChannel ) )
continue;
musicsave_t song;
Q_strncpy( song.songname, pChannel->sfx->getname(), sizeof( song.songname ) );
song.sampleposition = pChannel->pMixer->GetPositionForSave();
song.master_volume = pChannel->master_vol;
musiclist.AddToTail( song );
}
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : *song -
//-----------------------------------------------------------------------------
void S_RestartSong( const musicsave_t *song )
{
Assert( song );
// Start the song
CSfxTable *pSound = S_PrecacheSound( song->songname );
if ( pSound )
{
StartSoundParams_t params;
params.staticsound = true;
params.soundsource = SOUND_FROM_WORLD;
params.entchannel = CHAN_STATIC;
params.pSfx = pSound;
params.origin = vec3_origin;
params.fvol = ( (float)song->master_volume / 255.0f );
params.soundlevel = SNDLVL_NONE;
params.flags = SND_NOFLAGS;
params.pitch = PITCH_NORM;
params.initialStreamPosition = song->sampleposition;
S_StartSound( params );
if ( IsPC() )
{
// Now find the channel this went on and skip ahead in the mixer
for (int i = 0; i < total_channels; i++)
{
channel_t *ch = &channels[i];
if ( !ch->pMixer ||
!ch->pMixer->GetSource() )
{
continue;
}
if ( ch->pMixer->GetSource() != pSound->pSource )
{
continue;
}
ch->pMixer->SetPositionFromSaved( song->sampleposition );
break;
}
}
}
}
soundlevel_t SND_GetSndlvl ( channel_t *pchannel );
// calculate ammount of sound to be mixed to dsp, based on distance from listener
ConVar dsp_dist_min("dsp_dist_min", "0.0", FCVAR_DEMO|FCVAR_CHEAT); // range at which sounds are mixed at dsp_mix_min
ConVar dsp_dist_max("dsp_dist_max", "1440.0", FCVAR_DEMO|FCVAR_CHEAT); // range at which sounds are mixed at dsp_mix_max
ConVar dsp_mix_min("dsp_mix_min", "0.2", FCVAR_DEMO ); // dsp mix at dsp_dist_min distance "near"
ConVar dsp_mix_max("dsp_mix_max", "0.8", FCVAR_DEMO ); // dsp mix at dsp_dist_max distance "far"
ConVar dsp_db_min("dsp_db_min", "80", FCVAR_DEMO ); // sounds with sndlvl below this get dsp_db_mixdrop % less dsp mix
ConVar dsp_db_mixdrop("dsp_db_mixdrop", "0.5", FCVAR_DEMO ); // sounds with sndlvl below dsp_db_min get dsp_db_mixdrop % less mix
float DSP_ROOM_MIX = 1.0; // mix volume of dsp_room sounds when added back to 'dry' sounds
float DSP_NOROOM_MIX = 1.0; // mix volume of facing + facing away sounds. added to dsp_room_mix sounds
extern ConVar dsp_off;
// returns 0-1.0 dsp mix value. If sound source is at a range >= DSP_DIST_MAX, return a mix value of
// DSP_MIX_MAX. This mix value is used later to determine wet/dry mix ratio of sounds.
// This ramp changes with db level of sound source, and is set in the dsp room presets by room size
// empirical data: 0.78 is nominal mix for sound 100% at far end of room, 0.24 is mix for sound 25% into room
float SND_GetDspMix( channel_t *pchannel, int idist)
{
float mix;
float dist = (float)idist;
float dist_min = dsp_dist_min.GetFloat();
float dist_max = dsp_dist_max.GetFloat();
float mix_min;
float mix_max;
// only set dsp mix_min & mix_max when sound is first started
if ( pchannel->dsp_mix_min < 0 && pchannel->dsp_mix_max < 0 )
{
mix_min = dsp_mix_min.GetFloat(); // set via dsp_room preset
mix_max = dsp_mix_max.GetFloat(); // set via dsp_room preset
// set mix_min & mix_max based on db level of sound:
// sounds below dsp_db_min decrease dsp_mix_min & dsp_mix_max by N%
// ie: quiet sounds get less dsp mix than loud sounds
soundlevel_t sndlvl = SND_GetSndlvl( pchannel );
soundlevel_t sndlvl_min = (soundlevel_t)(dsp_db_min.GetInt());
if (sndlvl <= sndlvl_min)
{
mix_min *= dsp_db_mixdrop.GetFloat();
mix_max *= dsp_db_mixdrop.GetFloat();
}
pchannel->dsp_mix_min = mix_min;
pchannel->dsp_mix_max = mix_max;
}
else
{
mix_min = pchannel->dsp_mix_min;
mix_max = pchannel->dsp_mix_max;
}
// dspmix is 0 (100% mix to facing buffer) if dsp_off
if ( dsp_off.GetInt() )
return 0.0;
// doppler wavs are mixed dry
if ( pchannel->wavtype == CHAR_DOPPLER )
return 0.0;
// linear ramp - get dry mix %
// dist: 0->(max - min)
dist = clamp( dist, dist_min, dist_max ) - dist_min;
// dist: 0->1.0
dist = dist / (dist_max - dist_min);
// mix: min->max
mix = ((mix_max - mix_min) * dist) + mix_min;
return mix;
}
// calculate crossfade between wav left (close sound) and wav right (far sound) based on
// distance fron listener
#define DVAR_DIST_MIN (20.0 * 12.0) // play full 'near' sound at 20' or less
#define DVAR_DIST_MAX (110.0 * 12.0) // play full 'far' sound at 110' or more
#define DVAR_MIX_MIN 0.0
#define DVAR_MIX_MAX 1.0
// calculate mixing parameter for CHAR_DISTVAR wavs
// returns 0 - 1.0, 1.0 is 100% far sound (wav right)
float SND_GetDistanceMix( channel_t *pchannel, int idist)
{
float mix;
float dist = (float)idist;
// doppler wavs are 100% near - their spatialization is calculated later.
if ( pchannel->wavtype == CHAR_DOPPLER )
return 0.0;
// linear ramp - get dry mix %
// dist 0->(max - min)
dist = clamp( dist, (float) DVAR_DIST_MIN, (float) DVAR_DIST_MAX ) - (float) DVAR_DIST_MIN;
// dist 0->1.0
dist = dist / (DVAR_DIST_MAX - DVAR_DIST_MIN);
// mix min->max
mix = ((DVAR_MIX_MAX - DVAR_MIX_MIN) * dist) + DVAR_MIX_MIN;
return mix;
}
// given facing direction of source, and channel,
// return -1.0 - 1.0, where -1.0 is source facing away from listener
// and 1.0 is source facing listener
float SND_GetFacingDirection( channel_t *pChannel, const QAngle &source_angles )
{
Vector SF; // sound source forward direction unit vector
Vector SL; // sound -> listener unit vector
float dotSFSL;
// no facing direction unless wavtyp CHAR_DIRECTIONAL
if ( pChannel->wavtype != CHAR_DIRECTIONAL )
return 1.0;
VectorSubtract(listener_origin, pChannel->origin, SL);
VectorNormalize(SL);
// compute forward vector for sound entity
AngleVectors( source_angles, &SF, NULL, NULL );
// dot source forward unit vector with source to listener unit vector to get -1.0 - 1.0 facing.
// ie: projection of SF onto SL
dotSFSL = DotProduct( SF, SL );
return dotSFSL;
}
// calculate point of closest approach - caller must ensure that the
// forward facing vector of the entity playing this sound points in exactly the direction of
// travel of the sound. ie: for bullets or tracers, forward vector must point in traceline direction.
// return true if sound is to be played, false if sound cannot be heard (shot away from player)
bool SND_GetClosestPoint( channel_t *pChannel, QAngle &source_angles, Vector &vnearpoint )
{
// S - sound source origin
// L - listener origin
Vector SF; // sound source forward direction unit vector
Vector SL; // sound -> listener vector
Vector SD; // sound->closest point vector
vec_t dSLSF; // magnitude of project of SL onto SF
// P = SF (SF . SL) + S
// only perform this calculation for doppler wavs
if ( pChannel->wavtype != CHAR_DOPPLER )
return false;
// get vector 'SL' from sound source to listener
VectorSubtract(listener_origin, pChannel->origin, SL);
// compute sound->forward vector 'SF' for sound entity
AngleVectors( source_angles, &SF );
VectorNormalize( SF );
dSLSF = DotProduct( SL, SF );
if ( dSLSF <= 0 && !toolframework->IsToolRecording() )
{
// source is pointing away from listener, don't play anything
// unless we're recording in the tool, since we may play back from in front of the source
return false;
}
// project dSLSF along forward unit vector from sound source
VectorMultiply( SF, dSLSF, SD );
// output vector - add SD to sound source origin
VectorAdd( SD, pChannel->origin, vnearpoint );
return true;
}
// given point of nearest approach and sound source facing angles,
// return vector pointing into quadrant in which to play
// doppler left wav (incomming) and doppler right wav (outgoing).
// doppler left is point in space to play left doppler wav
// doppler right is point in space to play right doppler wav
// Also modifies channel pitch based on distance to nearest approach point
#define DOPPLER_DIST_LEFT_TO_RIGHT (4*12) // separate left/right sounds by 4'
#define DOPPLER_DIST_MAX (20*12) // max distance - causes min pitch
#define DOPPLER_DIST_MIN (1*12) // min distance - causes max pitch
#define DOPPLER_PITCH_MAX 1.5 // max pitch change due to distance
#define DOPPLER_PITCH_MIN 0.25 // min pitch change due to distance
#define DOPPLER_RANGE_MAX (10*12) // don't play doppler wav unless within this range
// UNDONE: should be set by caller!
void SND_GetDopplerPoints( channel_t *pChannel, QAngle &source_angles, Vector &vnearpoint, Vector &source_doppler_left, Vector &source_doppler_right)
{
Vector SF; // direction sound source is facing (forward)
Vector LN; // vector from listener to closest approach point
Vector DL;
Vector DR;
// nearpoint is closest point of approach, when playing CHAR_DOPPLER sounds
// SF is normalized vector in direction sound source is facing
AngleVectors( source_angles, &SF );
VectorNormalize( SF );
// source_doppler_left - location in space to play doppler left wav (incomming)
// source_doppler_right - location in space to play doppler right wav (outgoing)
VectorMultiply( SF, -1*DOPPLER_DIST_LEFT_TO_RIGHT, DL );
VectorMultiply( SF, DOPPLER_DIST_LEFT_TO_RIGHT, DR );
VectorAdd( vnearpoint, DL, source_doppler_left );
VectorAdd( vnearpoint, DR, source_doppler_right );
// set pitch of channel based on nearest distance to listener
// LN is vector from listener to closest approach point
VectorSubtract(vnearpoint, listener_origin, LN);
float pitch;
float dist = VectorLength( LN );
// dist varies 0->1
dist = clamp(dist, (float)DOPPLER_DIST_MIN, (float)DOPPLER_DIST_MAX);
dist = (dist - DOPPLER_DIST_MIN) / (DOPPLER_DIST_MAX - DOPPLER_DIST_MIN);
// pitch varies from max to min
pitch = DOPPLER_PITCH_MAX - dist * (DOPPLER_PITCH_MAX - DOPPLER_PITCH_MIN);
pChannel->basePitch = (int)(pitch * 100.0);
}
// console variables used to construct gain curve - don't change these!
extern ConVar snd_foliage_db_loss;
extern ConVar snd_gain;
extern ConVar snd_refdb;
extern ConVar snd_refdist;
extern ConVar snd_gain_max;
extern ConVar snd_gain_min;
ConVar snd_showstart( "snd_showstart", "0", FCVAR_CHEAT ); // showstart always skips info on player footsteps!
// 1 - show sound name, channel, volume, time
// 2 - show dspmix, distmix, dspface, l/r/f/r vols
// 3 - show sound origin coords
// 4 - show gain of dsp_room
// 5 - show dB loss due to obscured sound
// 6 - reserved
// 7 - show 2 and total gain & dist in ft. to sound source
#define SND_DB_MAX 140.0 // max db of any sound source
#define SND_DB_MED 90.0 // db at which compression curve changes
#define SND_DB_MIN 60.0 // min db of any sound source
#define SND_GAIN_PLAYER_WEAPON_DB 2.0 // increase player weapon gain by N dB
// dB = 20 log (amplitude/32768) 0 to -90.3dB
// amplitude = 32768 * 10 ^ (dB/20) 0 to +/- 32768
// gain = amplitude/32768 0 to 1.0
float Gain_To_dB ( float gain )
{
float dB = 20 * log ( gain );
return dB;
}
float dB_To_Gain ( float dB )
{
float gain = powf (10, dB / 20.0);
return gain;
}
float Gain_To_Amplitude ( float gain )
{
return gain * 32768;
}
float Amplitude_To_Gain ( float amplitude )
{
return amplitude / 32768;
}
soundlevel_t SND_GetSndlvl ( channel_t *pchannel )
{
return DIST_MULT_TO_SNDLVL( pchannel->dist_mult );
}
// The complete gain calculation, with SNDLVL given in dB is:
//
// GAIN = 1/dist * snd_refdist * 10 ^ ( ( SNDLVL - snd_refdb - (dist * snd_foliage_db_loss / 1200)) / 20 )
//
// for gain > SND_GAIN_THRESH, start curve smoothing with
//
// GAIN = 1 - 1 / (Y * GAIN ^ SND_GAIN_POWER)
//
// where Y = -1 / ( (SND_GAIN_THRESH ^ SND_GAIN_POWER) * (SND_GAIN_THRESH - 1) )
//
float SND_GetGainFromMult( float gain, float dist_mult, vec_t dist );
// gain curve construction
float SND_GetGain( channel_t *ch, bool fplayersound, bool fmusicsound, bool flooping, vec_t dist, bool bAttenuated )
{
VPROF_("SND_GetGain",2,VPROF_BUDGETGROUP_OTHER_SOUND,false,BUDGETFLAG_OTHER);
if ( ch->flags.m_bCompatibilityAttenuation )
{
// Convert to the original attenuation value.
soundlevel_t soundlevel = DIST_MULT_TO_SNDLVL( ch->dist_mult );
float flAttenuation = SNDLVL_TO_ATTN( soundlevel );
// Now get the goldsrc dist_mult and use the same calculation it uses in SND_Spatialize.
// Straight outta Goldsrc!!!
vec_t nominal_clip_dist = 1000.0;
float flGoldsrcDistMult = flAttenuation / nominal_clip_dist;
dist *= flGoldsrcDistMult;
float flReturnValue = 1.0f - dist;
flReturnValue = clamp( flReturnValue, 0.f, 1.f );
return flReturnValue;
}
else
{
float gain = snd_gain.GetFloat();
if ( fmusicsound )
{
gain = gain * snd_musicvolume.GetFloat();
gain = gain * g_DashboardMusicMixValue;
}
if ( ch->dist_mult )
{
gain = SND_GetGainFromMult( gain, ch->dist_mult, dist );
}
if ( fplayersound )
{
// player weapon sounds get extra gain - this compensates
// for npc distance effect weapons which mix louder as L+R into L,R
// Hack.
if ( ch->entchannel == CHAN_WEAPON )
gain = gain * dB_To_Gain( SND_GAIN_PLAYER_WEAPON_DB );
}
// modify gain if sound source not visible to player
gain = gain * SND_GetGainObscured( ch, fplayersound, flooping, bAttenuated );
if (snd_showstart.GetInt() == 6)
{
DevMsg( "(gain %1.3f : dist ft %1.1f) ", gain, (float)dist/12.0 );
snd_showstart.SetValue(5); // display once
}
return gain;
}
}
// always ramp channel gain changes over time
// returns ramped gain, given new target gain
#define SND_GAIN_FADE_TIME 0.25 // xfade seconds between obscuring gain changes
float SND_FadeToNewGain( channel_t *ch, float gain_new )
{
if ( gain_new == -1.0 )
{
// if -1 passed in, just keep fading to existing target
gain_new = ch->ob_gain_target;
}
// if first time updating, store new gain into gain & target, return
// if gain_new is close to existing gain, store new gain into gain & target, return
if ( ch->flags.bfirstpass || (fabs (gain_new - ch->ob_gain) < 0.01))
{
ch->ob_gain = gain_new;
ch->ob_gain_target = gain_new;
ch->ob_gain_inc = 0.0;
return gain_new;
}
// set up new increment to new target
float frametime = g_pSoundServices->GetHostFrametime();
float speed;
speed = ( frametime / SND_GAIN_FADE_TIME ) * (gain_new - ch->ob_gain);
ch->ob_gain_inc = fabs(speed);
// ch->ob_gain_inc = fabs(gain_new - ch->ob_gain) / 10.0;
ch->ob_gain_target = gain_new;
// if not hit target, keep approaching
if ( fabs( ch->ob_gain - ch->ob_gain_target ) > 0.01 )
{
ch->ob_gain = Approach( ch->ob_gain_target, ch->ob_gain, ch->ob_gain_inc );
}
else
{
// close enough, set gain = target
ch->ob_gain = ch->ob_gain_target;
}
return ch->ob_gain;
}
#define SND_TRACE_UPDATE_MAX 2 // max of N channels may be checked for obscured source per frame
static int g_snd_trace_count = 0; // total tracelines for gain obscuring made this frame
// All new sounds must traceline once,
// but cap the max number of tracelines performed per frame
// for longer or looping sounds to SND_TRACE_UPDATE_MAX.
bool SND_ChannelOkToTrace( channel_t *ch )
{
// always trace first time sound is spatialized (doesn't update counter)
if ( ch->flags.bfirstpass )
{
ch->flags.bTraced = true;
return true;
}
// if already traced max channels this frame, return
if ( g_snd_trace_count >= SND_TRACE_UPDATE_MAX )
return false;
// ok to trace if this sound hasn't yet been traced in this round
if ( ch->flags.bTraced )
return false;
// set flag - don't traceline this sound again until all others have
// been traced
ch->flags.bTraced = true;
g_snd_trace_count++; // total traces this frame
return true;
}
// determine if we need to reset all flags for traceline limiting -
// this happens if we hit a frame whein no tracelines occur ie: all currently
// playing sounds are blocked.
void SND_ChannelTraceReset( void )
{
if ( g_snd_trace_count )
return;
// if no tracelines performed this frame, then reset all
// trace flags
for (int i = 0; i < total_channels; i++)
channels[i].flags.bTraced = false;
}
bool SND_IsLongWave( channel_t *pChannel )
{
CAudioSource *pSource = pChannel->sfx ? pChannel->sfx->pSource : NULL;
if ( pSource )
{
if ( pSource->IsStreaming() )
return true;
// UNDONE: Do this on long wave files too?
#if 0
float length = (float)pSource->SampleCount() / (float)pSource->SampleRate();
if ( length > 0.75f )
return true;
#endif
}
return false;
}
ConVar snd_obscured_gain_db( "snd_obscured_gain_dB", "-2.70", FCVAR_CHEAT ); // dB loss due to obscured sound source
// drop gain on channel if sound emitter obscured by
// world, unbroken windows, closed doors, large solid entities etc.
float SND_GetGainObscured( channel_t *ch, bool fplayersound, bool flooping, bool bAttenuated )
{
float gain = 1.0;
int count = 1;
float snd_gain_db; // dB loss due to obscured sound source
// Unattenuated sounds don't get obscured.
if ( !bAttenuated )
return 1.0f;
if ( fplayersound )
return gain;
// During signon just apply regular state machine since world hasn't been
// created or settled yet...
if ( !SND_IsInGame() )
{
if ( !toolframework->InToolMode() )
{
gain = SND_FadeToNewGain( ch, -1.0 );
}
return gain;
}
// don't do gain obscuring more than once on short one-shot sounds
if ( !ch->flags.bfirstpass && !ch->flags.isSentence && !flooping && !SND_IsLongWave(ch) )
{
gain = SND_FadeToNewGain( ch, -1.0 );
return gain;
}
snd_gain_db = snd_obscured_gain_db.GetFloat();
// if long or looping sound, process N channels per frame - set 'processed' flag, clear by
// cycling through all channels - this maintains a cap on traces per frame
if ( !SND_ChannelOkToTrace( ch ) )
{
// just keep updating fade to existing target gain - no new trace checking
gain = SND_FadeToNewGain( ch, -1.0 );
return gain;
}
// set up traceline from player eyes to sound emitting entity origin
Vector endpoint = ch->origin;
trace_t tr;
CTraceFilterWorldOnly filter; // UNDONE: also test for static props?
Ray_t ray;
ray.Init( MainViewOrigin(), endpoint );
g_pEngineTraceClient->TraceRay( ray, MASK_BLOCK_AUDIO, &filter, &tr );
if (tr.DidHit() && tr.fraction < 0.99)
{
// can't see center of sound source:
// build extents based on dB sndlvl of source,
// test to see how many extents are visible,
// drop gain by snd_gain_db per extent hidden
Vector endpoints[4];
soundlevel_t sndlvl = DIST_MULT_TO_SNDLVL( ch->dist_mult );
float radius;
Vector vsrc_forward;
Vector vsrc_right;
Vector vsrc_up;
Vector vecl;
Vector vecr;
Vector vecl2;
Vector vecr2;
int i;
// get radius
if ( ch->radius > 0 )
radius = ch->radius;
else
radius = dB_To_Radius( sndlvl); // approximate radius from soundlevel
// set up extent endpoints - on upward or downward diagonals, facing player
for (i = 0; i < 4; i++)
endpoints[i] = endpoint;
// vsrc_forward is normalized vector from sound source to listener
VectorSubtract( listener_origin, endpoint, vsrc_forward );
VectorNormalize( vsrc_forward );
VectorVectors( vsrc_forward, vsrc_right, vsrc_up );
VectorAdd( vsrc_up, vsrc_right, vecl );
// if src above listener, force 'up' vector to point down - create diagonals up & down
if ( endpoint.z > listener_origin.z + (10 * 12) )
vsrc_up.z = -vsrc_up.z;
VectorSubtract( vsrc_up, vsrc_right, vecr );
VectorNormalize( vecl );
VectorNormalize( vecr );
// get diagonal vectors from sound source
vecl2 = radius * vecl;
vecr2 = radius * vecr;
vecl = (radius / 2.0) * vecl;
vecr = (radius / 2.0) * vecr;
// endpoints from diagonal vectors
endpoints[0] += vecl;
endpoints[1] += vecr;
endpoints[2] += vecl2;
endpoints[3] += vecr2;
// drop gain for each point on radius diagonal that is obscured
for (count = 0, i = 0; i < 4; i++)
{
// UNDONE: some endpoints are in walls - in this case, trace from the wall hit location
ray.Init( MainViewOrigin(), endpoints[i] );
g_pEngineTraceClient->TraceRay( ray, MASK_BLOCK_AUDIO, &filter, &tr );
if (tr.DidHit() && tr.fraction < 0.99 && !tr.startsolid )
{
count++; // skip first obscured point: at least 2 points + center should be obscured to hear db loss
if (count > 1)
gain = gain * dB_To_Gain( snd_gain_db );
}
}
}
if ( flooping && snd_showstart.GetInt() == 7)
{
static float g_drop_prev = 0;
float drop = (count-1) * snd_gain_db;
if (drop != g_drop_prev)
{
DevMsg( "dB drop: %1.4f \n", drop);
g_drop_prev = drop;
}
}
// crossfade to new gain
gain = SND_FadeToNewGain( ch, gain );
return gain;
}
// convert sound db level to approximate sound source radius,
// used only for determining how much of sound is obscured by world
#define SND_RADIUS_MAX (20.0 * 12.0) // max sound source radius
#define SND_RADIUS_MIN (2.0 * 12.0) // min sound source radius
inline float dB_To_Radius ( float db )
{
float radius = SND_RADIUS_MIN + (SND_RADIUS_MAX - SND_RADIUS_MIN) * (db - SND_DB_MIN) / (SND_DB_MAX - SND_DB_MIN);
return radius;
}
struct snd_spatial_t
{
int chan; // 0..4 cycles through up to 5 channels
int cycle; // 0..2 cycles through 3 vectors per channel
int dist[5][3]; // stores last 3 channel distance values [channel][cycle]
float value_prev[5]; // previous value per channel
double last_change;
};
bool g_ssp_init = false;
snd_spatial_t g_ssp;
// return 0..1 percent difference between a & b
float PercentDifference( float a, float b )
{
float vp;
if (!(int)a && !(int)b)
return 0.0;
if (!(int)a || !(int)b)
return 1.0;
if (a > b)
vp = b / a;
else
vp = a / b;
return (1.0 - vp);
}
// NOTE: Do not change SND_WALL_TRACE_LEN without also changing PRC_MDY6 delay value in snd_dsp.cpp!
#define SND_WALL_TRACE_LEN (100.0*12.0) // trace max of 100' = max of 100 milliseconds of linear delay
#define SND_SPATIAL_WAIT (0.25) // seconds to wait between traces
// change mod delay value on chan 0..3 to v (inches)
void DSP_SetSpatialDelay( int chan, float v )
{
// remap delay value 0..1200 to 1.0 to -1.0 for modulation
float value = ( v / SND_WALL_TRACE_LEN) - 1.0; // -1.0...0
value = value * 2.0; // -2.0...0
value += 1.0; // -1.0...1.0 (0...1200)
value *= -1.0; // 1.0...-1.0 (0...1200)
// assume first processor in dsp_spatial is the modulating delay unit for DSP_ChangePresetValue
int iproc = 0;
DSP_ChangePresetValue( idsp_spatial, chan, iproc, value );
/*
if (chan & 0x01)
DevMsg("RDly: %3.0f \n", v/12 );
else
DevMsg("LDly: %3.0f \n", v/12 );
*/
}
// use non-feedback delay to stereoize (or make quad, or quad + center) the mono dsp_room fx,
// This simulates the average sum of delays caused by reflections
// from the left and right walls relative to the player. The average delay
// difference between left & right wall is (l + r)/2. This becomes the average
// delay difference between left & right ear.
// call at most once per frame to update player->wall spatial delays
void SND_SetSpatialDelays()
{
VPROF("SoundSpatialDelays");
float dist, v, vp;
Vector v_dir, v_dir2;
int chan_max = (g_AudioDevice->IsSurround() ? 4 : 2) + (g_AudioDevice->IsSurroundCenter() ? 1 : 0); // 2, 4, 5 channels
// use listener_forward2d, which doesn't change when player looks up/down.
Vector listener_forward2d;
ConvertListenerVectorTo2D( &listener_forward2d, &listener_right );
// init struct if 1st time through
if ( !g_ssp_init )
{
Q_memset(&g_ssp, 0, sizeof(snd_spatial_t));
g_ssp_init = true;
}
// return if dsp_spatial is 0
if ( !dsp_spatial.GetInt() )
return;
// if listener has not been updated, do nothing
if ((listener_origin == vec3_origin) &&
(listener_forward == vec3_origin) &&
(listener_right == vec3_origin) &&
(listener_up == vec3_origin) )
return;
if ( !SND_IsInGame() )
return;
// get time
double dtime = g_pSoundServices->GetHostTime();
// compare to previous time - if starting new check - don't check for new room until timer expires
if (!g_ssp.chan && !g_ssp.cycle)
{
if (fabs(dtime - g_ssp.last_change) < SND_SPATIAL_WAIT)
return;
}
// cycle through forward, left, rearward vectors, averaging to get left/right delay
// count[chan][cycle] 0,1 0,2 0,3 1,1 1,2 1,3 2,1 2,2 2,3 ...
g_ssp.cycle++;
if (g_ssp.cycle == 3)
{
g_ssp.cycle = 0;
// cycle through front left, front right, rear left, rear right, front center delays
g_ssp.chan++;
if (g_ssp.chan >= chan_max )
g_ssp.chan = 0;
}
// set up traceline from player eyes to surrounding walls
switch( g_ssp.chan )
{
default:
case 0: // front left: trace max 100' 'cone' to player's left
if ( g_AudioDevice->IsSurround() )
{
// 4-5 speaker case - front left
v_dir = (-listener_right + listener_forward2d) / 2.0;
v_dir = g_ssp.cycle ? (g_ssp.cycle == 1 ? -listener_right * 0.5: listener_forward2d * 0.5) : v_dir;
}
else
{
// 2 speaker case - left
v_dir = listener_right * -1.0;
v_dir2 = g_ssp.cycle ? (g_ssp.cycle == 1 ? listener_forward2d * 0.5 : -listener_forward2d * 0.5) : v_dir;
v_dir = (v_dir + v_dir2) / 2.0;
}
break;
case 1: // front right: trace max 100' 'cone' to player's right
if ( g_AudioDevice->IsSurround() )
{
// 4-5 speaker case - front right
v_dir = (listener_right + listener_forward2d) / 2.0;
v_dir = g_ssp.cycle ? (g_ssp.cycle == 1 ? listener_right * 0.5: listener_forward2d * 0.5) : v_dir;
}
else
{
// 2 speaker case - right
v_dir = listener_right;
v_dir2 = g_ssp.cycle ? (g_ssp.cycle == 1 ? listener_forward2d * 0.5 : -listener_forward2d * 0.5) : v_dir;
v_dir = (v_dir + v_dir2) / 2.0;
}
break;
case 2: // rear left: trace max 100' 'cone' to player's rear left
v_dir = (listener_right + listener_forward2d) / -2.0;
v_dir = g_ssp.cycle ? (g_ssp.cycle == 1 ? -listener_right * 0.5 : -listener_forward2d * 0.5) : v_dir;
break;
case 3: // rear right: trace max 100' 'cone' to player's rear right
v_dir = (listener_right - listener_forward2d) / 2.0;
v_dir = g_ssp.cycle ? (g_ssp.cycle == 1 ? listener_right * 0.5: -listener_forward2d * 0.5) : v_dir;
break;
case 4: // front center: trace max 100' 'cone' to player's front
v_dir = listener_forward2d;
v_dir2 = g_ssp.cycle ? (g_ssp.cycle == 1 ? listener_right * 0.15 : -listener_right * 0.15) : v_dir;
v_dir = (v_dir + v_dir2);
break;
}
Vector endpoint;
trace_t tr;
CTraceFilterWorldOnly filter;
endpoint = MainViewOrigin() + v_dir * SND_WALL_TRACE_LEN;
Ray_t ray;
ray.Init( MainViewOrigin(), endpoint );
g_pEngineTraceClient->TraceRay( ray, MASK_BLOCK_AUDIO, &filter, &tr );
dist = SND_WALL_TRACE_LEN;
if ( tr.DidHit() )
{
dist = VectorLength( tr.endpos - MainViewOrigin() );
}
g_ssp.dist[g_ssp.chan][g_ssp.cycle] = dist;
// set new result in dsp_spatial delay params when all delay values have been filled in
if (!g_ssp.cycle && !g_ssp.chan)
{
// update delay for each channel
for (int chan = 0; chan < chan_max; chan++)
{
// compute average of 3 traces per channel
v = (g_ssp.dist[chan][0] + g_ssp.dist[chan][1] + g_ssp.dist[chan][2]) / 3.0;
vp = g_ssp.value_prev[chan];
// only change if 10% difference from previous
if ((vp != v) && int(v) && (PercentDifference( v, vp ) >= 0.1))
{
// update when we have data for all L/R && RL/RR channels...
if (chan & 0x1)
{
float vr = fpmin( v, (50*12.0f) );
float vl = fpmin(g_ssp.value_prev[chan-1], (50*12.0f));
/* UNDONE: not needed, now that this applies only to dsp 'room' buffer
// ensure minimum separation = average distance to walls
float dmin = (vl + vr) / 2.0; // average distance to walls
float d = vl - vr; // l/r separation
// if separation is less than average, increase min
if (abs(d) < dmin/2)
{
if (vl > vr)
vl += dmin/2 - d;
else
vr += dmin/2 - d;
}
*/
DSP_SetSpatialDelay(chan-1, vl);
DSP_SetSpatialDelay(chan, vr);
}
// update center chan
if (chan == 4)
{
float vl = fpmin( v, (50*12.0f) );
DSP_SetSpatialDelay(chan, vl);
}
}
g_ssp.value_prev[chan] = v;
}
// update wait timer now that all values have been checked
g_ssp.last_change = dtime;
}
}
// Dsp Automatic Selection:
// a) enabled by setting dsp_room to DSP_AUTOMATIC. Subsequently, dsp_automatic is the actual dsp value for dsp_room.
// b) disabled by setting dsp_room to anything else
// c) while enabled, detection nodes are placed as player moves into a new space
// i. at each node, a new dsp setting is calculated and dsp_automatic is set to an appropriate preset
// ii. new nodes are set when player moves out of sight of previous node
// iii. moving into line of sight of a detection node causes closest node to player to set dsp_automatic
// see void DAS_CheckNewRoomDSP( ) for main entrypoint
ConVar das_debug( "adsp_debug", "0", FCVAR_ARCHIVE );
// >0: draw blue dsp detection node location
// >1: draw green room trace height detection bars
// 3: draw yellow horizontal trace bars for room width/depth detection
// 4: draw yellow upward traces for height detection
// 5: draw teal box around all props around player
// 6: draw teal box around room as detected
#define DAS_CWALLS 20 // # of wall traces to save for calculating room dimensions
#define DAS_ROOM_TRACE_LEN (400.0*12.0) // max size of trace to check for room dimensions
#define DAS_AUTO_WAIT 0.25 // wait min of n seconds between dsp_room changes and update checks
#define DAS_WIDTH_MIN 0.4 // min % change in avg width of any wall pair to cause new dsp
#define DAS_REFL_MIN 0.5 // min % change in avg refl of any wall to cause new dsp
#define DAS_SKYHIT_MIN 0.8 // min % change in # of sky hits per wall
#define DAS_DIST_MIN (4.0 * 12.0) // min distance between room dsp changes
#define DAS_DIST_MAX (40.0 * 12.0) // max distance to preserve room dsp changes
#define DAS_DIST_MIN_OUTSIDE (6.0 * 12.0) // min distance between room dsp changes outside
#define DAS_DIST_MAX_OUTSIDE (100.0 * 12.0) // max distance to preserve room dsp changes outside
#define IVEC_DIAG_UP 8 // start of diagonal up vectors
#define IVEC_UP 18 // up vector
#define IVEC_DOWN 19 // down vector
#define DAS_REFLECTIVITY_NORM 0.5
#define DAS_REFLECTIVITY_SKY 0.0
// auto dsp room struct
struct das_room_t
{
int dist[DAS_CWALLS]; // distance in units from player to axis aligned and diagonal walls
float reflect[DAS_CWALLS]; // acoustic reflectivity per wall
float skyhits[DAS_CWALLS]; // every sky hit adds 0.1
Vector hit[DAS_CWALLS]; // location of trace hit on wall - used for calculating average centers
Vector norm[DAS_CWALLS]; // wall normal at hit location
Vector vplayer; // 'frozen' location above player's head
Vector vplayer_eyes; // 'frozen' location player's eyes
int width_max; // max width
int length_max; // max length
int height_max; // max height
float refl_avg; // running average of reflectivity of all walls
float refl_walls[6]; // left,right,front,back,ceiling,floor reflectivities
float sky_pct; // percent of sky hits
Vector room_mins; // room bounds
Vector room_maxs;
double last_dsp_change; // time since last dsp change
float diffusion; // 0..1.0 check radius (avg of width_avg) for # of props - scale diffusion based on # found
short iwall; // cycles through walls 0..5, ensuring only one trace per frame
short ent_count; // count of entities found in radius
bool bskyabove; // true if sky found above player (ie: outside)
bool broomready; // true if all distances are filled in and room is ready to check
short lowceiling; // if non-zero, ceiling directly above player if < 112 units
};
// dsp detection node
struct das_node_t
{
Vector vplayer; // position
bool fused; // true if valid node
bool fseesplayer; // true if node sees player on last check
short dsp_preset; // preset
int range_min; // min,max detection ranges
int range_max;
int dist; // last distance to player
// room parameters when node was created:
das_room_t room;
};
#define DAS_CNODES 40 // keep around last n nodes - must be same as DSP_CAUTO_PRESETS!!!
das_node_t g_das_nodes[DAS_CNODES]; // all dsp detection nodes
das_node_t *g_pdas_last_node = NULL; // last node that saw player
int g_das_check_next; // next node to check
int g_das_store_next; // next place to store node
bool g_das_all_checked; // true if all nodes checked
int g_das_checked_count; // count of nodes checked in latest pass
das_room_t g_das_room; // room detector
bool g_bdas_room_init = 0;
bool g_bdas_init_nodes = 0;
bool g_bdas_create_new_node = 0;
bool DAS_TraceNodeToPlayer( das_room_t *proom, das_node_t *pnode );
void DAS_InitAutoRoom( das_room_t *proom);
void DAS_DebugDrawTrace ( trace_t *ptr, int r, int g, int b, float duration, int imax );
Vector g_das_vec3[DAS_CWALLS]; // trace vectors to walls, ceiling, floor
void DAS_InitNodes( void )
{
Q_memset(g_das_nodes, 0, sizeof(das_node_t) * DAS_CNODES);
g_das_check_next = 0;
g_das_store_next = 0;
g_das_all_checked = 0;
g_das_checked_count = 0;
// init all rooms
for (int i = 0; i < DAS_CNODES; i++)
DAS_InitAutoRoom( &(g_das_nodes[i].room) );
// init trace vectors
// set up trace vectors for max, min width
float vl = DAS_ROOM_TRACE_LEN;
float vlu = DAS_ROOM_TRACE_LEN * 0.52;
float vlu2 = DAS_ROOM_TRACE_LEN * 0.48; // don't use 'perfect' diagonals
g_das_vec3[0].Init(vl, 0.0, 0.0); // x left
g_das_vec3[1].Init(-vl, 0.0, 0.0); // x right
g_das_vec3[2].Init(0.0, vl, 0.0); // y front
g_das_vec3[3].Init(0.0, -vl, 0.0); // y back
g_das_vec3[4].Init(-vlu, vlu2, 0.0); // diagonal front left
g_das_vec3[5].Init(vlu, -vlu2, 0.0); // diagonal rear right
g_das_vec3[6].Init(vlu, vlu2, 0.0); // diagonal front right
g_das_vec3[7].Init(-vlu, -vlu2, 0.0); // diagonal rear left
// set up trace vectors for max height - on x=y diagonal
g_das_vec3[8].Init(vlu, vlu2, vlu/2.0); // front right up A x,y,z/2 (IVEC_DIAG_UP)
g_das_vec3[9].Init(vlu, vlu2, vlu); // front right up B x,y,z
g_das_vec3[10].Init(vlu/2.0, vlu2/2.0, vlu); // front right up C x/2,y/2,z
g_das_vec3[11].Init(-vlu, -vlu2, vlu/2.0); // rear left up A -x,-y,z/2
g_das_vec3[12].Init(-vlu, -vlu2, vlu); // rear left up B -x,-y,z
g_das_vec3[13].Init(-vlu/2.0, -vlu2/2.0, vlu); // rear left up C -x/2,-y/2,z
// set up trace vectors for max height - on x axis & y axis
g_das_vec3[14].Init(-vlu, 0, vlu); // left up B -x,0,z
g_das_vec3[15].Init(0, vlu/2.0, vlu); // front up C -x/2,0,z
g_das_vec3[16].Init(0, -vlu, vlu); // rear up B x,0,z
g_das_vec3[17].Init(vlu/2.0, 0, vlu); // right up C x/2,0,z
g_das_vec3[18].Init(0.0, 0.0, vl); // up (IVEC_UP)
g_das_vec3[19].Init(0.0, 0.0, -vl); // down (IVEC_DOWN)
}
void DAS_InitAutoRoom( das_room_t *proom)
{
Q_memset(proom, 0, sizeof (das_room_t));
}
// reset all nodes for next round of visibility checks between player & nodes
void DAS_ResetNodes( void )
{
for (int i = 0; i < DAS_CNODES; i++)
{
g_das_nodes[i].fseesplayer = false;
g_das_nodes[i].dist = 0;
}
g_das_all_checked = false;
g_das_checked_count = 0;
g_bdas_create_new_node = false;
}
// utility function - return next index, wrap at max
int DAS_GetNextIndex( int *pindex, int max )
{
int i = *pindex;
int j;
j = i+1;
if ( j >= max )
j = 0;
*pindex = j;
return i;
}
// returns true if dsp node is within range of player
bool DAS_NodeInRange( das_room_t *proom, das_node_t *pnode )
{
float dist;
dist = VectorLength( proom->vplayer - pnode->vplayer );
// player can still see previous room selection point, and it's less than n feet away,
// then flag this node as visible
pnode->dist = dist;
return ( dist <= pnode->range_max );
}
// update next valid node - set up internal node state if it can see player
// called once per frame
// returns true if all nodes have been checked
bool DAS_CheckNextNode( das_room_t *proom )
{
int i, j;
if ( g_das_all_checked )
return true;
// find next valid node
for (j = 0; j < DAS_CNODES; j++)
{
// track number of nodes checked
g_das_checked_count++;
// get next node in range to check
i = DAS_GetNextIndex( &g_das_check_next, DAS_CNODES );
if ( g_das_nodes[i].fused && DAS_NodeInRange( proom, &(g_das_nodes[i]) ) )
{
// trace to see if player can still see node,
// if so stop checking
if ( DAS_TraceNodeToPlayer( proom, &(g_das_nodes[i]) ))
goto checknode_exit;
}
}
checknode_exit:
// flag that all nodes have been checked
if ( g_das_checked_count >= DAS_CNODES )
g_das_all_checked = true;
return g_das_all_checked;
}
int DAS_GetNextNodeIndex()
{
return g_das_store_next;
}
// store new node for room
void DAS_StoreNode( das_room_t *proom, int dsp_preset)
{
// overwrite node in cyclic list
int i = DAS_GetNextIndex( &g_das_store_next, DAS_CNODES );
g_das_nodes[i].dsp_preset = dsp_preset;
g_das_nodes[i].fused = true;
g_das_nodes[i].vplayer = proom->vplayer;
// calculate node scanning range_max based on room size
if ( !proom->bskyabove )
{
// inside range - halls & tunnels have nodes every 5*width
g_das_nodes[i].range_max = fpmin((int)DAS_DIST_MAX, min(proom->width_max * 5, proom->length_max) );
g_das_nodes[i].range_min = DAS_DIST_MIN;
}
else
{
// outside range
g_das_nodes[i].range_max = DAS_DIST_MAX_OUTSIDE;
g_das_nodes[i].range_min = DAS_DIST_MIN_OUTSIDE;
}
g_das_nodes[i].fseesplayer = false;
g_das_nodes[i].dist = 0;
g_das_nodes[i].room = *proom;
// update last node visible as this node
g_pdas_last_node = &(g_das_nodes[i]);
}
// check all updated nodes,
// return dsp_preset of largest node (by area) that can see player
// return -1 if no preset found
// NOTE: outside nodes can't see player if player is inside and vice versa
// foutside is true if player is outside
int DAS_GetDspPreset( bool foutside )
{
int dsp_preset = -1;
int i;
// int dist_min = 100000;
int area_max = 0;
int area;
// find node that represents room with greatest floor area, return its preset.
for (i = 0; i < DAS_CNODES; i++)
{
if (g_das_nodes[i].fused && g_das_nodes[i].fseesplayer)
{
area = (g_das_nodes[i].room.width_max * g_das_nodes[i].room.length_max);
if ( g_das_nodes[i].room.bskyabove == foutside )
{
if (area > area_max)
{
area_max = area;
dsp_preset = g_das_nodes[i].dsp_preset;
// save pointer to last node that saw player
g_pdas_last_node = &(g_das_nodes[i]);
}
}
/*
// find nearest node, return its preset
if (g_das_nodes[i].dist < dist_min)
{
if ( g_das_nodes[i].room.bskyabove == foutside )
{
dist_min = g_das_nodes[i].dist;
dsp_preset = g_das_nodes[i].dsp_preset;
// save pointer to last node that saw player
g_pdas_last_node = &(g_das_nodes[i]);
}
}
*/
}
}
return dsp_preset;
}
// custom trace filter:
// a) never hit player or monsters or entities
// b) always hit world, or moveables or static props
class CTraceFilterDAS : public ITraceFilter
{
public:
bool ShouldHitEntity( IHandleEntity *pHandleEntity, int contentsMask )
{
IClientUnknown *pUnk = static_cast<IClientUnknown*>(pHandleEntity);
IClientEntity *pEntity;
if ( !pUnk )
return false;
// don't hit non-collideable props
if ( StaticPropMgr()->IsStaticProp( pHandleEntity ) )
{
ICollideable *pCollide = StaticPropMgr()->GetStaticProp( pHandleEntity);
if (!pCollide)
return false;
}
// don't hit any ents
pEntity = pUnk->GetIClientEntity();
if ( pEntity )
return false;
return true;
}
virtual TraceType_t GetTraceType() const
{
return TRACE_EVERYTHING_FILTER_PROPS;
}
};
#define DAS_TRACE_MASK (CONTENTS_SOLID|CONTENTS_MOVEABLE|CONTENTS_WINDOW)
// returns true if clear line exists between node and player
// if node can see player, sets up node distance and flag fseesplayer
bool DAS_TraceNodeToPlayer( das_room_t *proom, das_node_t *pnode )
{
trace_t trP;
CTraceFilterDAS filterP;
bool fseesplayer = false;
float dist;
Ray_t ray;
ray.Init( proom->vplayer, pnode->vplayer );
g_pEngineTraceClient->TraceRay( ray, DAS_TRACE_MASK, &filterP, &trP );
dist = VectorLength( proom->vplayer - pnode->vplayer );
// player can still see previous room selection point, and it's less than n feet away,
// then flag this node as visible
if ( !trP.DidHit() && (dist <= DAS_DIST_MAX) )
{
fseesplayer = true;
pnode->dist = dist;
}
pnode->fseesplayer = fseesplayer;
return fseesplayer;
}
// update room boundary maxs, mins
void DAS_SetRoomBounds( das_room_t *proom, Vector &hit, bool bheight )
{
Vector maxs, mins;
maxs = proom->room_maxs;
mins = proom->room_mins;
if (!bheight)
{
if (hit.x > maxs.x)
maxs.x = hit.x;
if (hit.x < mins.x)
mins.x = hit.x;
if (hit.z > maxs.z)
maxs.z = hit.z;
if (hit.z < mins.z)
mins.z = hit.z;
}
if (bheight)
{
if (hit.y > maxs.y)
maxs.y = hit.y;
if (hit.y < mins.y)
mins.y = hit.y;
}
proom->room_maxs = maxs;
proom->room_mins = mins;
}
// when all walls are updated, calculate max length, width, height, reflectivity, sky hit%, room center
// returns true if room parameters are in good location to place a node
// returns false if room parameters are not in good location to place a node
// note: false occurs if up vector doesn't hit sky, but one or more up diagonal vectors do hit sky
bool DAS_CalcRoomProps( das_room_t *proom )
{
int length_max = 0;
int width_max = 0;
int height_max = 0;
int dist[4];
float area1, area2;
int height;
int i;
int j;
int k;
bool b_diaghitsky = false;
// reject this location if up vector doesn't hit sky, but
// one or more up diagonals do hit sky -
// in this case, player is under a slight overhang, narrow bridge, or
// standing just inside a window or doorway. keep looking for better node location
for (i = IVEC_DIAG_UP; i < IVEC_UP; i++)
{
if (proom->skyhits[i] > 0.0)
b_diaghitsky = true;
}
if (b_diaghitsky && !(proom->skyhits[IVEC_UP] > 0.0))
return false;
// get all distance pairs
for (i = 0; i < IVEC_DIAG_UP; i+=2)
dist[i/2] = proom->dist[i] + proom->dist[i+1]; // 1st pair is width
// if areas differ by more than 25%
// select the pair with the greater area
// if areas do not differ by more than 25%, select the pair with the
// longer measured distance. Filters incorrect selection due to diagonals.
area1 = (float)(dist[0] * dist[1]);
area2 = (float)(dist[2] * dist[3]);
area1 = (int)area1 == 0 ? 1.0 : area1;
area2 = (int)area2 == 0 ? 1.0 : area2;
if ( PercentDifference(area1, area2) > 0.25 )
{
// areas are more than 25% different - select pair with greater area
j = area1 > area2 ? 0 : 2;
}
else
{
// select pair with longer measured distance
int iMaxDist = 0; // index to max dist
int dmax = 0;
for (i = 0; i < 4; i++)
{
if (dist[i] > dmax)
{
dmax = dist[i];
iMaxDist = i;
}
}
j = iMaxDist > 1 ? 2 : 0;
}
// width is always the smaller of the dimensions
width_max = min (dist[j], dist[j+1]);
length_max = max (dist[j], dist[j+1]);
// get max height
for (i = IVEC_DIAG_UP; i < IVEC_DOWN; i++)
{
height = proom->dist[i];
if (height > height_max)
height_max = height;
}
proom->length_max = length_max;
proom->width_max = width_max;
proom->height_max = height_max;
// get room max,min from chosen width, depth
// 0..3 or 4..7
for ( i = j*2; i < 4+(j*2); i++)
DAS_SetRoomBounds( proom, proom->hit[i], false );
// get room height min from down trace
proom->room_mins.z = proom->hit[IVEC_DOWN].z;
// reset room height max to player trace height
proom->room_maxs.z = proom->vplayer.z;
// draw box around room max,min
if (das_debug.GetInt() == 6)
{
// draw box around all objects detected
Vector maxs = proom->room_maxs;
Vector mins = proom->room_mins;
Vector orig = (maxs + mins) / 2.0;
Vector absMax = maxs - orig;
Vector absMin = mins - orig;
CDebugOverlay::AddBoxOverlay( orig, absMax, absMin, vec3_angle, 255, 0, 255, 0, 60.0f );
}
// calculate average reflectivity
float refl = 0.0;
// average reflectivity for walls
// 0..3 or 4..7
for ( k = 0, i = j*2; i < 4+(j*2); i++, k++)
{
refl += proom->reflect[i];
proom->refl_walls[k] = proom->reflect[i];
}
// assume ceiling is open
proom->refl_walls[4] = 0.0;
// get ceiling reflectivity, if any non zero
for ( i = IVEC_DIAG_UP; i < IVEC_DOWN; i++)
{
if (proom->reflect[i] == 0.0)
{
// if any upward trace hit sky, exit;
// ceiling reflectivity is 0.0
proom->refl_walls[4] = 0.0;
i = IVEC_DOWN; // exit loop
}
else
{
// upward trace didn't hit sky, keep checking
proom->refl_walls[4] = proom->reflect[i];
}
}
// add in ceiling reflectivity, if any
refl += proom->refl_walls[4];
// get floor reflectivity
refl += proom->reflect[IVEC_DOWN];
proom->refl_walls[5] = proom->reflect[IVEC_DOWN];
proom->refl_avg = refl / 6.0;
// calculate sky hit percent for this wall
float sky_pct = 0.0;
// 0..3 or 4..7
for ( i = j*2; i < 4+(j*2); i++)
sky_pct += proom->skyhits[i];
for ( i = IVEC_DIAG_UP; i < IVEC_DOWN; i++)
{
if (proom->skyhits[i] > 0.0)
{
// if any upward trace hit sky, exit loop
sky_pct += proom->skyhits[i];
i = IVEC_DOWN;
}
}
// get floor skyhit
sky_pct += proom->skyhits[IVEC_DOWN];
proom->sky_pct = sky_pct;
// check for sky above
proom->bskyabove = false;
for (i = IVEC_DIAG_UP; i < IVEC_DOWN; i++)
{
if (proom->skyhits[i] > 0.0)
proom->bskyabove = true;
}
return true;
}
// return true if trace hit solid
// return false if trace hit sky or didn't hit anything
bool DAS_HitSolid( trace_t *ptr )
{
// if hit nothing return false
if (!ptr->DidHit())
return false;
// if hit sky, return false (not solid)
if (ptr->surface.flags & SURF_SKY)
return false;
return true;
}
// returns true if trace hit sky
bool DAS_HitSky( trace_t *ptr )
{
if (ptr->DidHit() && (ptr->surface.flags & SURF_SKY))
return true;
if (!ptr->DidHit() )
{
float dz = ptr->endpos.z - ptr->startpos.z;
if ( dz > 200*12.0f )
return true;
}
return false;
}
bool DAS_ScanningForHeight( das_room_t *proom )
{
return (proom->iwall >= IVEC_DIAG_UP);
}
bool DAS_ScanningForWidth( das_room_t *proom )
{
return (proom->iwall < IVEC_DIAG_UP);
}
bool DAS_ScanningForFloor( das_room_t *proom )
{
return (proom->iwall == IVEC_DOWN);
}
ConVar das_door_height("adsp_door_height", "112"); // standard door height hl2
ConVar das_wall_height("adsp_wall_height", "128"); // standard wall height hl2
ConVar das_low_ceiling("adsp_low_ceiling", "108"); // low ceiling height hl2
// set origin for tracing out to walls to point above player's head
// allows calculations over walls and floor obstacles, and above door openings
// WARNING: the current settings are optimal for skipping floor and ceiling clutter,
// and for detecting rooms without 'looking' through doors or windows. Don't change these cvars for hl2!
void DAS_SetTraceHeight( das_room_t *proom, trace_t *ptrU, trace_t *ptrD )
{
// NOTE: when tracing down through player's box, endpos and startpos are reversed and
// startsolid and allsolid are true.
int zup = abs(ptrU->endpos.z - ptrU->startpos.z); // height above player's head
int zdown = abs(ptrD->endpos.z - ptrD->startpos.z); // distance to floor from player's head
int h;
h = zup + zdown;
int door_height = das_door_height.GetInt();
int wall_height = das_wall_height.GetInt();
int low_ceiling = das_low_ceiling.GetInt();
if (h > low_ceiling && h <= wall_height)
{
// low ceiling - trace out just above standard door height @ 112
if (h > door_height)
proom->vplayer.z = fpmin(ptrD->endpos.z, ptrD->startpos.z) + door_height + 1;
else
proom->vplayer.z = fpmin(ptrD->endpos.z, ptrD->startpos.z) + h - 1;
}
else if ( h > wall_height )
{
// tall ceiling - trace out over standard walls @ 128
proom->vplayer.z = fpmin(ptrD->endpos.z, ptrD->startpos.z) + wall_height + 1;
}
else
{
// very low ceiling, trace out from just below ceiling
proom->vplayer.z = fpmin(ptrD->endpos.z, ptrD->startpos.z) + h - 1;
proom->lowceiling = h;
}
Assert (proom->vplayer.z <= ptrU->endpos.z);
if (das_debug.GetInt() > 1)
{
// draw line to height, and between floor and ceiling
CDebugOverlay::AddLineOverlay( ptrD->endpos, ptrU->endpos, 0, 255, 0, 255, false, 20 );
Vector mins;
Vector maxs;
mins.Init(-1,-1,-2.0);
maxs.Init(1,1,0);
CDebugOverlay::AddBoxOverlay( proom->vplayer, mins, maxs, vec3_angle, 255, 0, 0, 0, 20 );
CDebugOverlay::AddBoxOverlay( ptrU->endpos, mins, maxs, vec3_angle, 0, 255, 0, 0, 20 );
CDebugOverlay::AddBoxOverlay( ptrD->endpos, mins, maxs, vec3_angle, 0, 255, 0, 0, 20 );
}
}
// prepare room struct for new round of checks:
// clear out struct,
// init trace height origin by finding space above player's head
// returns true if player is in valid position to begin checks from
bool DAS_StartTraceChecks( das_room_t *proom )
{
// starting new check: store player position, init maxs, mins
proom->vplayer_eyes = MainViewOrigin();
proom->vplayer = MainViewOrigin();
proom->height_max = 0;
proom->width_max = 0;
proom->length_max = 0;
proom->room_maxs.Init (0.0, 0.0, 0.0);
proom->room_mins.Init (10000.0, 10000.0, 10000.0);
proom->lowceiling = 0;
// find point between player's head and ceiling - trace out to walls from here
trace_t trU, trD;
CTraceFilterDAS filterU, filterD;
Vector v_dir = g_das_vec3[IVEC_DOWN]; // down - find floor
Vector endpoint = proom->vplayer + v_dir;
Ray_t ray;
ray.Init( proom->vplayer, endpoint );
g_pEngineTraceClient->TraceRay( ray, DAS_TRACE_MASK, &filterD, &trD );
// if player jumping or in air, don't continue
if (trD.DidHit() && abs(trD.endpos.z - trD.startpos.z) > 72)
return false;
v_dir = g_das_vec3[IVEC_UP]; // up - find ceiling
endpoint = proom->vplayer + v_dir;
ray.Init( proom->vplayer, endpoint );
g_pEngineTraceClient->TraceRay( ray, DAS_TRACE_MASK, &filterU, &trU );
// if down trace hits floor, set trace height, otherwise default is player eye location
if ( DAS_HitSolid( &trD) )
DAS_SetTraceHeight( proom, &trU, &trD );
return true;
}
void DAS_DebugDrawTrace ( trace_t *ptr, int r, int g, int b, float duration, int imax)
{
// das_debug == 3: draw horizontal trace bars for room width/depth detection
// das_debug == 4: draw upward traces for height detection
if (das_debug.GetInt() != imax)
return;
CDebugOverlay::AddLineOverlay( ptr->startpos, ptr->endpos, r, g, b, 255, false, duration );
Vector mins;
Vector maxs;
mins.Init(-1,-1,-2.0);
maxs.Init(1,1,0);
CDebugOverlay::AddBoxOverlay( ptr->endpos, mins, maxs, vec3_angle, r, g, b, 0, duration );
}
// wall surface data
struct das_surfdata_t
{
float dist; // distance to player
float reflectivity; // acoustic reflectivity of material on surface
Vector hit; // trace hit location
Vector norm; // wall normal at hit location
};
// trace hit wall surface, get info about surface and store in surfdata struct
// if scanning for height, bounce a second trace off of ceiling and get dist to floor
void DAS_GetSurfaceData( das_room_t *proom, trace_t *ptr, das_surfdata_t *psurfdata )
{
float dist; // distance to player
float reflectivity; // acoustic reflectivity of material on surface
Vector hit; // trace hit location
Vector norm; // wall normal at hit location
surfacedata_t *psurf;
psurf = physprop->GetSurfaceData( ptr->surface.surfaceProps );
reflectivity = psurf ? psurf->audio.reflectivity : DAS_REFLECTIVITY_NORM;
// keep wall hit location and normal, to calc room bounds and center
norm = ptr->plane.normal;
// get length to hit location
dist = VectorLength(ptr->endpos - ptr->startpos);
// if started tracing from within player box, startpos & endpos may be flipped
if (ptr->endpos.z >= ptr->startpos.z)
hit = ptr->endpos;
else
hit = ptr->startpos;
// if checking for max height by bouncing several vectors off of ceiling:
// ignore returned normal from 1st bounce, just search straight down from trace hit location
if ( DAS_ScanningForHeight( proom ) && !DAS_ScanningForFloor( proom ) )
{
trace_t tr2;
CTraceFilterDAS filter2;
norm.Init(0.0, 0.0, -1.0);
Vector endpoint = hit + ( norm * DAS_ROOM_TRACE_LEN );
Ray_t ray;
ray.Init( hit, endpoint );
g_pEngineTraceClient->TraceRay( ray, DAS_TRACE_MASK, &filter2, &tr2 );
//DAS_DebugDrawTrace( &tr2, 255, 255, 0, 10, 1);
if (tr2.DidHit())
{
// get distance between surfaces
dist = VectorLength(tr2.endpos - tr2.startpos);
}
}
// set up surface struct and return
psurfdata->dist = dist;
psurfdata->hit = hit;
psurfdata->norm = norm;
psurfdata->reflectivity = reflectivity;
}
// algorithm for detecting approximate size of space around player. Handles player in corner & non-axis aligned rooms.
// also handles player on catwalk or player under small bridge/overhang.
// The goal is to only change the dsp room description if the the player moves into
// a space which is SIGNIFICANTLY different from the previously set dsp space.
// save player position. find a point above player's head and trace out from here.
// from player position, get max width and max length:
// from player position,
// a) trace x,-x, y,-y axes
// b) trace xy, -xy, x-y, -x-y diagonals
// c) select largest room size detected from max width, max length
// from player position, get height
// a) trace out along front-up (or left-up, back-up, right-up), save hit locations
// b) trace down -z from hit locations
// c) save max height
// when max width, max length, max height all updated, get new player position
// get average room size & wall materials:
// update averages with one traceline per frame only
// returns true if room is fully updated and ready to check
bool DAS_UpdateRoomSize( das_room_t *proom )
{
Vector endpoint;
Vector startpoint;
Vector v_dir;
int iwall;
bool bskyhit = false;
das_surfdata_t surfdata;
// do nothing if room already fully checked
if ( proom->broomready )
return true;
// cycle through all walls, floor, ceiling
// get wall index
iwall = proom->iwall;
// get height above player and init proom for new round of checks
if (iwall == 0)
{
if (!DAS_StartTraceChecks( proom ))
return false; // bad location to check room - player is jumping etc.
}
// get trace vector
v_dir = g_das_vec3[iwall];
// trace out from trace origin, in axis-aligned direction or along diagonals
// if looking for max height, trace from top of player's eyes
if ( DAS_ScanningForHeight( proom ) )
{
startpoint = proom->vplayer_eyes;
endpoint = proom->vplayer_eyes + v_dir;
}
else
{
startpoint = proom->vplayer;
endpoint = proom->vplayer + v_dir;
}
// try less expensive world-only trace first (no props, no ents - just try to hit walls)
trace_t tr;
CTraceFilterWorldOnly filter;
Ray_t ray;
ray.Init( startpoint, endpoint );
g_pEngineTraceClient->TraceRay( ray, CONTENTS_SOLID, &filter, &tr );
// if didn't hit world, or we hit sky when looking horizontally,
// retrace, this time including props
if ( !DAS_HitSolid( &tr ) && DAS_ScanningForWidth( proom ) )
{
CTraceFilterDAS filterDas;
ray.Init( startpoint, endpoint );
g_pEngineTraceClient->TraceRay( ray, DAS_TRACE_MASK, &filterDas, &tr );
}
if (das_debug.GetInt() > 2)
{
// draw trace lines
if ( DAS_HitSolid( &tr ) )
DAS_DebugDrawTrace( &tr, 0, 255, 255, 10, DAS_ScanningForHeight( proom ) + 3);
else
DAS_DebugDrawTrace( &tr, 255, 0, 0, 10, DAS_ScanningForHeight( proom ) + 3); // red lines if sky hit or no hit
}
// init surface data with defaults, in case we didn't hit world
surfdata.dist = DAS_ROOM_TRACE_LEN;
surfdata.reflectivity = DAS_REFLECTIVITY_SKY; // assume sky or open area
surfdata.hit = endpoint; // trace hit location
surfdata.norm = -v_dir;
// check for sky hits
if ( DAS_HitSky( &tr ) )
{
bskyhit = true;
if ( DAS_ScanningForWidth( proom ) )
// ignore horizontal sky hits for distance calculations
surfdata.dist = 1.0;
else
surfdata.dist = surfdata.dist; // debug
}
// get length of trace if it hit world
// if hit solid and not sky (tr.DidHit() && !bskyhit)
// get surface information
if ( DAS_HitSolid( &tr) )
DAS_GetSurfaceData( proom, &tr, &surfdata );
// store surface data
proom->dist[iwall] = surfdata.dist;
proom->reflect[iwall] = clamp(surfdata.reflectivity, 0.0f, 1.0f);
proom->skyhits[iwall] = bskyhit ? 0.1 : 0.0;
proom->hit[iwall] = surfdata.hit;
proom->norm[iwall] = surfdata.norm;
// update wall counter
proom->iwall++;
if (proom->iwall == DAS_CWALLS)
{
bool b_good_node_location;
// calculate room mins, maxs, reflectivity etc
b_good_node_location = DAS_CalcRoomProps( proom );
// reset wall counter
proom->iwall = 0;
proom->broomready = b_good_node_location; // room ready to check if good node location
return b_good_node_location;
}
return false; // room not yet fully updated
}
// create entity enumerator for counting ents & summing volume of ents in room
class CDasEntEnum : public IPartitionEnumerator
{
public:
int m_count; // # of ents in space
float m_volume; // space occupied by ents
public:
void Reset()
{
m_count = 0;
m_volume = 0.0;
}
// called with each handle...
IterationRetval_t EnumElement( IHandleEntity *pHandleEntity )
{
float vol;
// get bounding box of entity
// Generate a collideable
ICollideable *pCollideable = g_pEngineTraceClient->GetCollideable( pHandleEntity );
if ( !pCollideable )
return ITERATION_CONTINUE;
// Check for solid
if ( !IsSolid( pCollideable->GetSolid(), pCollideable->GetSolidFlags() ) )
return ITERATION_CONTINUE;
m_count++;
// compute volume of space occupied by entity
Vector mins = pCollideable->OBBMins();
Vector maxs = pCollideable->OBBMaxs();
vol = fabs((maxs.x - mins.x) * (maxs.y - mins.y) * (maxs.z - mins.z));
m_volume += vol; // add to total vol
if (das_debug.GetInt() == 5)
{
// draw box around all objects detected
Vector orig = pCollideable->GetCollisionOrigin();
CDebugOverlay::AddBoxOverlay( orig, mins, maxs, pCollideable->GetCollisionAngles(), 255, 0, 255, 0, 60.0f );
}
return ITERATION_CONTINUE;
}
};
// determine # of solid ents/props within detected room boundaries
// and set diffusion based on count of ents and spatial volume of ents
void DAS_SetDiffusion( das_room_t *proom )
{
// BRJ 7/12/05
// This was commented out because the y component of proom->room_mins, proom->room_maxs was never
// being computed, causing a bogus box to be sent to the partition system. The results of
// this computation (namely the diffusion + ent_count fields of das_room_t) were never being used.
// Therefore, we'll avoid the enumeration altogether
proom->diffusion = 0.0f;
proom->ent_count = 0;
/*
CDasEntEnum enumerator;
SpatialPartitionListMask_t mask = PARTITION_CLIENT_SOLID_EDICTS; // count only solid ents in room
int count;
float vol;
float volroom;
float dfn;
enumerator.Reset();
SpatialPartition()->EnumerateElementsInBox(mask, proom->room_mins, proom->room_maxs, true, &enumerator );
count = enumerator.m_count;
vol = enumerator.m_volume;
// compute diffusion from volume
// how much space around player is filled with props?
volroom = (proom->room_maxs.x - proom->room_mins.x) * (proom->room_maxs.y - proom->room_mins.y) * (proom->room_maxs.z - proom->room_mins.z);
volroom = fabs(volroom);
if ( !(int)volroom )
volroom = 1.0;
dfn = vol / volroom; // % of total volume occupied by props
dfn = clamp (dfn, 0.0, 1.0);
proom->diffusion = dfn;
proom->ent_count = count;
*/
}
// debug routine to display current room params
void DAS_DisplayRoomDEBUG( das_room_t *proom, bool fnew, float preset )
{
float dx,dy,dz;
Vector ctr;
float count;
if (das_debug.GetInt() == 0)
return;
dx = proom->length_max / 12.0;
dy = proom->width_max / 12.0;
dz = proom->height_max / 12.0;
float refl = proom->refl_avg;
count = (float)(proom->ent_count);
float fsky = (proom->bskyabove ? 1.0 : 0.0);
if (fnew)
DevMsg( "NEW DSP NODE: size:(%.0f,%.0f) height:(%.0f) dif %.4f : refl %.4f : cobj: %.0f : sky %.0f \n", dx, dy, dz, proom->diffusion, refl, count, fsky);
if (!fnew && preset < 0.0)
return;
if (preset >= 0.0)
{
if (proom == NULL)
return;
DevMsg( "DSP PRESET: %.0f size:(%.0f,%.0f) height:(%.0f) dif %.4f : refl %.4f : cobj: %.0f : sky %.0f \n", preset, dx, dy, dz, proom->diffusion, refl, count, fsky);
return;
}
// draw box around new node location
Vector mins;
Vector maxs;
mins.Init(-8,-8,-16);
maxs.Init(8,8,0);
CDebugOverlay::AddBoxOverlay( proom->vplayer, mins, maxs, vec3_angle, 0, 0, 255, 0, 1000.0f );
// draw red box around node origin
mins.Init(-0.5,-0.5,-1.0);
maxs.Init(0.5,0.5,0);
CDebugOverlay::AddBoxOverlay( proom->vplayer, mins, maxs, vec3_angle, 255, 0, 0, 0, 1000.0f );
CDebugOverlay::AddTextOverlay( proom->vplayer, 0, 10, 1.0, "DSP NODE" );
}
// check newly calculated room parameters against current stored params.
// if different, return true.
// NOTE: only call when all proom params have been calculated.
// return false if this is not a good location for creating a new node
bool DAS_CheckNewRoom( das_room_t *proom )
{
bool bnewroom;
float dw,dw2,dr,ds,dh;
int cchanged = 0;
das_room_t *proom_prev = NULL;
Vector2D v2d;
Vector v3d;
float dist;
// player can't see previous node, determine if this is a good place to lay down
// a new node. Get room at last seen node for comparison
if (g_pdas_last_node)
proom_prev = &(g_pdas_last_node->room);
// no previous room node saw player, go create new room node
if (!proom_prev)
{
bnewroom = true;
goto check_ret;
}
// if player not at least n feet from last node, return false
v3d = proom->vplayer - proom_prev->vplayer;
v2d.Init(v3d.x, v3d.y);
dist = Vector2DLength(v2d);
if (dist <= DAS_DIST_MIN)
return false;
// see if room size has changed significantly since last node
bnewroom = true;
dw = 0.0;
dw2 = 0.0;
dh = 0.0;
dr = 0.0;
if ( proom_prev->width_max != 0 )
dw = (float)proom->width_max / (float)proom_prev->width_max; // max width delta
if ( proom_prev->length_max != 0 )
dw2 = (float)proom->length_max / (float)proom_prev->length_max; // max length delta
if ( proom_prev->height_max != 0 )
dh = (float)proom->height_max / (float)proom_prev->height_max; // max height delta
if ( proom_prev->refl_avg != 0.0 )
dr = proom->refl_avg / proom_prev->refl_avg; // reflectivity delta
ds = fabs( proom->sky_pct - proom_prev->sky_pct); // sky hits delta
if (dw > 1.0) dw = 1.0 / dw;
if (dw2 > 1.0) dw = 1.0 / dw2;
if (dh > 1.0) dh = 1.0 / dh;
if (dr > 1.0) dr = 1.0 / dr;
if ( (1.0 - dw) >= DAS_WIDTH_MIN )
cchanged++;
if ( (1.0 - dw2) >= DAS_WIDTH_MIN )
cchanged++;
// if ( (1.0 - dh) >= DAS_WIDTH_MIN ) // don't change room based on height change
// cchanged++;
// new room only if at least 1 changed
if (cchanged >= 1)
goto check_ret;
// if ( (1.0 - dr) >= DAS_REFL_MIN ) // don't change room based on reflectivity change
// goto check_ret;
// if (ds >= DAS_SKYHIT_MIN )
// goto check_ret;
// new room if sky above changes state
if (proom->bskyabove != proom_prev->bskyabove)
goto check_ret;
// room didn't change significantly, return false
bnewroom = false;
check_ret:
if ( bnewroom )
{
// if low ceiling detected < 112 units, and max height is > low ceiling height by 20%, discard - no change
// this detects player in doorway, under pipe or narrow bridge
if ( proom->lowceiling && (proom->lowceiling < proom->height_max))
{
float h = (float)(proom->lowceiling) / (float)proom->height_max;
if (h < 0.8)
return false;
}
DAS_SetDiffusion( proom );
}
DAS_DisplayRoomDEBUG( proom, bnewroom, -1.0 );
return bnewroom;
}
extern int DSP_ConstructPreset( bool bskyabove, int width, int length, int height, float fdiffusion, float freflectivity, float *psurf_refl, int inode, int cnodes );
// select new dsp_room based on size, wall materials
// (or modulate params for current dsp)
// returns new preset # for dsp_automatic
int DAS_GetRoomDSP( das_room_t *proom, int inode )
{
// preset constructor
// call dsp module with params, get dsp preset back
bool bskyabove = proom->bskyabove;
int width = proom->width_max;
int length = proom->length_max;
int height = proom->height_max;
float fdiffusion = proom->diffusion;
float freflectivity = proom->refl_avg;
float surf_refl[6];
// fill array of surface reflectivities - for left,right,front,back,ceiling,floor
for (int i = 0; i < 6; i++)
surf_refl[i] = proom->refl_walls[i];
return DSP_ConstructPreset( bskyabove, width, length, height, fdiffusion, freflectivity, surf_refl, inode, DAS_CNODES );
}
// main entry point: call once per frame to update dsp_automatic
// for automatic room detection. dsp_room must be set to DSP_AUTOMATIC to enable.
// NOTE: this routine accumulates traceline information over several frames - it
// never traces more than 3 times per call, and normally just once per call.
void DAS_CheckNewRoomDSP( )
{
VPROF("DAS_CheckNewRoomDSP");
das_room_t *proom = &g_das_room;
int dsp_preset;
bool bRoom_ready = false;
// if listener has not been updated, do nothing
if ((listener_origin == vec3_origin) &&
(listener_forward == vec3_origin) &&
(listener_right == vec3_origin) &&
(listener_up == vec3_origin) )
return;
if ( !SND_IsInGame() )
return;
// make sure we init nodes & vectors first time this is called
if ( !g_bdas_init_nodes )
{
g_bdas_init_nodes = 1;
DAS_InitNodes();
}
if ( !DSP_CheckDspAutoEnabled())
{
// make sure room params are reinitialized each time autoroom is selected
g_bdas_room_init = 0;
return;
}
if ( !g_bdas_room_init )
{
g_bdas_room_init = 1;
DAS_InitAutoRoom( proom );
}
// get time
double dtime = g_pSoundServices->GetHostTime();
// compare to previous time - don't check for new room until timer expires
// ie: wait at least DAS_AUTO_WAIT seconds between preset changes
if ( fabs(dtime - proom->last_dsp_change) < DAS_AUTO_WAIT )
return;
// first, update room size parameters, see if room is ready to check - if room is updated, return true right away
// 3 traces per frame while accumulating room size info
for (int i = 0 ; i < 3; i++)
bRoom_ready = DAS_UpdateRoomSize( proom );
if (!bRoom_ready)
return;
if ( !g_bdas_create_new_node )
{
// next, check all nodes for line of sight to player - if all checked, return true right away
if ( !DAS_CheckNextNode( proom ) )
{
// check all nodes first
return;
}
// find out if any previously stored nodes can see player,
// if so, get closest node's dsp preset
dsp_preset = DAS_GetDspPreset( proom->bskyabove );
if (dsp_preset != -1)
{
// an existing node can see player - just set preset and return
if (dsp_preset != dsp_room_GetInt())
{
// changed preset, so update timestamp
proom->last_dsp_change = g_pSoundServices->GetHostTime();
if (g_pdas_last_node)
DAS_DisplayRoomDEBUG( &(g_pdas_last_node->room), false, (float)dsp_preset );
}
DSP_SetDspAuto( dsp_preset );
goto check_new_room_exit;
}
}
g_bdas_create_new_node = true;
// no nodes can see player, need to try to create a new one
// check for 'new' room around player
if ( DAS_CheckNewRoom( proom ) )
{
// new room found - update dsp_automatic
dsp_preset = DAS_GetRoomDSP( proom, DAS_GetNextNodeIndex() );
DSP_SetDspAuto( dsp_preset );
// changed preset, so update timestamp
proom->last_dsp_change = g_pSoundServices->GetHostTime();
// save room as new node
DAS_StoreNode( proom, dsp_preset );
goto check_new_room_exit;
}
check_new_room_exit:
// reset new node creation flag - start checking for visible nodes again
g_bdas_create_new_node = false;
// reset room checking flag - start checking room around player again
proom->broomready = false;
// reset node checking flag - start checking nodes around player again
DAS_ResetNodes();
return;
}
// remap contents of volumes[] arrary if sound originates from player, or is music, and is 100% 'mono'
// ie: same volume in all channels
void RemapPlayerOrMusicVols( channel_t *ch, int volumes[CCHANVOLUMES/2], bool fplayersound, bool fmusicsound, float mono )
{
VPROF_("RemapPlayerOrMusicVols", 2, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
if ( !fplayersound && !fmusicsound )
return; // no remapping
if ( ch->flags.bSpeaker )
return; // don't remap speaker sounds rebroadcast on player
// get total volume
float vol_total = 0.0;
int k;
for (k = 0; k < CCHANVOLUMES/2; k++)
vol_total += (float)volumes[k];
if ( !g_AudioDevice->IsSurround() )
{
if (mono < 1.0)
return;
// remap 2 chan non-spatialized versions of player and music sounds
// note: this is required to keep volumes same as 4 & 5 ch cases!
float vol_dist_music[] = {1.0, 1.0}; // FL, FR music volumes
float vol_dist_player[] = {1.0, 1.0}; // FL, FR player volumes
float *pvol_dist;
pvol_dist = (fplayersound ? vol_dist_player : vol_dist_music);
for (k = 0; k < 2; k++)
volumes[k] = clamp((int)(vol_total * pvol_dist[k]), 0, 255);
return;
}
// surround sound configuration...
if ( fplayersound ) // && (ch->bstereowav && ch->wavtype != CHAR_DIRECTIONAL && ch->wavtype != CHAR_DISTVARIANT) )
{
// NOTE: player sounds also get n% overall volume boost.
//float vol_dist5[] = {0.29, 0.29, 0.09, 0.09, 0.63}; // FL, FR, RL, RR, FC - 5 channel (mono source) volume distribution
//float vol_dist5st[] = {0.29, 0.29, 0.09, 0.09, 0.63}; // FL, FR, RL, RR, FC - 5 channel (stereo source) volume distribution
float vol_dist5[] = {0.30, 0.30, 0.09, 0.09, 0.59}; // FL, FR, RL, RR, FC - 5 channel (mono source) volume distribution
float vol_dist5st[] = {0.30, 0.30, 0.09, 0.09, 0.59}; // FL, FR, RL, RR, FC - 5 channel (stereo source) volume distribution
float vol_dist4[] = {0.50, 0.50, 0.15, 0.15, 0.00}; // FL, FR, RL, RR, 0 - 4 channel (mono source) volume distribution
float vol_dist4st[] = {0.50, 0.50, 0.15, 0.15, 0.00}; // FL, FR, RL, RR, 0 - 4 channel (stereo source)volume distribution
float *pvol_dist;
if ( ch->flags.bstereowav && (ch->wavtype == CHAR_OMNI || ch->wavtype == CHAR_SPATIALSTEREO || ch->wavtype == 0))
{
pvol_dist = (g_AudioDevice->IsSurroundCenter() ? vol_dist5st : vol_dist4st);
}
else
{
pvol_dist = (g_AudioDevice->IsSurroundCenter() ? vol_dist5 : vol_dist4);
}
for (k = 0; k < 5; k++)
volumes[k] = clamp((int)(vol_total * pvol_dist[k]), 0, 255);
return;
}
// Special case for music in surround mode
if ( fmusicsound )
{
float vol_dist5[] = {0.5, 0.5, 0.25, 0.25, 0.0}; // FL, FR, RL, RR, FC - 5 channel distribution
float vol_dist4[] = {0.5, 0.5, 0.25, 0.25, 0.0}; // FL, FR, RL, RR, 0 - 4 channel distribution
float *pvol_dist;
pvol_dist = (g_AudioDevice->IsSurroundCenter() ? vol_dist5 : vol_dist4);
for (k = 0; k < 5; k++)
volumes[k] = clamp((int)(vol_total * pvol_dist[k]), 0, 255);
return;
}
return;
}
static int s_nSoundGuid = 0;
void SND_ActivateChannel( channel_t *pChannel )
{
Q_memset( pChannel, 0, sizeof(*pChannel) );
g_ActiveChannels.Add( pChannel );
pChannel->guid = ++s_nSoundGuid;
}
/*
=================
SND_Spatialize
=================
*/
void SND_Spatialize(channel_t *ch)
{
VPROF("SND_Spatialize");
vec_t dist;
Vector source_vec;
Vector source_vec_DL;
Vector source_vec_DR;
Vector source_doppler_left;
Vector source_doppler_right;
bool fdopplerwav = false;
bool fplaydopplerwav = false;
bool fvalidentity;
float gain;
float scale = 1.0;
bool fplayersound = false;
bool fmusicsound = false;
float mono = 0.0;
bool bAttenuated = true;
ch->dspface = 1.0; // default facing direction: always facing player
ch->dspmix = 0; // default mix 0% dsp_room fx
ch->distmix = 0; // default 100% left (near) wav
#if !defined( _X360 )
if ( ch->sfx &&
ch->sfx->pSource &&
ch->sfx->pSource->GetType() == CAudioSource::AUDIO_SOURCE_VOICE )
{
Voice_Spatialize( ch );
}
#endif
if ( IsSoundSourceLocalPlayer( ch->soundsource ) && !toolframework->InToolMode() )
{
// sounds coming from listener actually come from a short distance directly in front of listener
// in tool mode however, the view entity is meaningless, since we're viewing from arbitrary locations in space
fplayersound = true;
}
// assume 'dry', playeverwhere sounds are 'music' or 'voiceover'
if ( ch->flags.bdry && ch->dist_mult <= 0 )
{
fmusicsound = true;
fplayersound = false;
}
// update channel's position in case ent that made the sound is moving.
QAngle source_angles;
source_angles.Init(0.0, 0.0, 0.0);
Vector entOrigin = ch->origin;
bool looping = false;
CAudioSource *pSource = ch->sfx ? ch->sfx->pSource : NULL;
if ( pSource )
{
looping = pSource->IsLooped();
}
SpatializationInfo_t si;
si.info.Set(
ch->soundsource,
ch->entchannel,
ch->sfx ? ch->sfx->getname() : "",
ch->origin,
ch->direction,
ch->master_vol,
DIST_MULT_TO_SNDLVL( ch->dist_mult ),
looping,
ch->pitch,
listener_origin,
ch->speakerentity );
si.type = SpatializationInfo_t::SI_INSPATIALIZATION;
si.pOrigin = &entOrigin;
si.pAngles = &source_angles;
si.pflRadius = NULL;
if ( ch->soundsource != 0 && ch->radius == 0 )
{
si.pflRadius = &ch->radius;
}
{
VPROF_("SoundServices->GetSoundSpatializtion", 2, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
fvalidentity = g_pSoundServices->GetSoundSpatialization( ch->soundsource, si );
}
if ( ch->flags.bUpdatePositions )
{
AngleVectors( source_angles, &ch->direction );
ch->origin = entOrigin;
}
else
{
VectorAngles( ch->direction, source_angles );
}
if ( ch->userdata != 0 )
{
g_pSoundServices->GetToolSpatialization( ch->userdata, ch->guid, si );
if ( ch->flags.bUpdatePositions )
{
AngleVectors( source_angles, &ch->direction );
ch->origin = entOrigin;
}
}
#if 0
// !!!UNDONE - above code assumes the ENT hasn't been removed or respawned as another ent!
// !!!UNDONE - fix this by flagging some entities (ie: glass) as immobile. Don't spatialize them.
if ( !fvalidendity)
{
// Turn off the sound while the entity doesn't exist or is not in the PVS.
goto ClearAllVolumes;
}
#endif // 0
fdopplerwav = ((ch->wavtype == CHAR_DOPPLER) && !fplayersound);
if ( fdopplerwav )
{
VPROF_("SND_Spatialize doppler", 2, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
Vector vnearpoint; // point of closest approach to listener,
// along sound source forward direction (doppler wavs)
vnearpoint = ch->origin; // default nearest sound approach point
// calculate point of closest approach for CHAR_DOPPLER wavs, replace source_vec
fplaydopplerwav = SND_GetClosestPoint( ch, source_angles, vnearpoint );
// if doppler sound was 'shot' away from listener, don't play it
if ( !fplaydopplerwav )
goto ClearAllVolumes;
// find location of doppler left & doppler right points
SND_GetDopplerPoints( ch, source_angles, vnearpoint, source_doppler_left, source_doppler_right);
// source_vec_DL is vector from listener to doppler left point
// source_vec_DR is vector from listener to doppler right point
VectorSubtract(source_doppler_left, listener_origin, source_vec_DL );
VectorSubtract(source_doppler_right, listener_origin, source_vec_DR );
// normalized vectors to left and right doppler locations
dist = VectorNormalize( source_vec_DL );
VectorNormalize( source_vec_DR );
// don't play doppler if out of range
// unless recording in the tool, since we may play back in range
if ( dist > DOPPLER_RANGE_MAX && !toolframework->IsToolRecording() )
goto ClearAllVolumes;
}
else
{
// source_vec is vector from listener to sound source
if ( fplayersound )
{
// get 2d forward direction vector, ignoring pitch angle
Vector listener_forward2d;
ConvertListenerVectorTo2D( &listener_forward2d, &listener_right );
// player sounds originate from 1' in front of player, 2d
VectorMultiply(listener_forward2d, 12.0, source_vec );
}
else
{
VectorSubtract(ch->origin, listener_origin, source_vec);
}
// normalize source_vec and get distance from listener to source
dist = VectorNormalize( source_vec );
}
// calculate dsp mix based on distance to listener & sound level (linear approximation)
ch->dspmix = SND_GetDspMix( ch, dist );
// calculate sound source facing direction for CHAR_DIRECTIONAL wavs
if ( !fplayersound )
{
ch->dspface = SND_GetFacingDirection( ch, source_angles );
// calculate mixing parameter for CHAR_DISTVAR wavs
ch->distmix = SND_GetDistanceMix( ch, dist );
}
// for sounds with a radius, spatialize left/right/front/rear evenly within the radius
if ( ch->radius > 0 && dist < ch->radius && !fdopplerwav )
{
float interval = ch->radius * 0.5;
mono = dist - interval;
if ( mono < 0.0 )
mono = 0.0;
mono /= interval;
mono = 1.0 - mono;
// mono is 0.0 -> 1.0 from radius 100% to radius 50%
}
// don't pan sounds with no attenuation
if ( ch->dist_mult <= 0 && !fdopplerwav )
{
// sound is centered left/right/front/back
mono = 1.0;
bAttenuated = false;
}
if ( ch->wavtype == CHAR_OMNI )
{
// omni directional sound sources are mono mix, all speakers
// ie: they only attenuate by distance, not by source direction.
mono = 1.0;
bAttenuated = false;
}
// calculate gain based on distance, atmospheric attenuation, interposed objects
// perform compression as gain approaches 1.0
gain = SND_GetGain( ch, fplayersound, fmusicsound, looping, dist, bAttenuated );
// map gain through global mixer by soundtype
// gain *= SND_GetVolFromSoundtype( ch->soundtype );
int last_mixgroupid;
gain *= MXR_GetVolFromMixGroup( ch->mixgroups, &last_mixgroupid );
// if playing a word, get volume scale of word - scale gain
scale = VOX_GetChanVol(ch);
gain *= scale;
// save spatialized volume and mixgroupid for display later
ch->last_mixgroupid = last_mixgroupid;
if ( fdopplerwav )
{
VPROF_("SND_Spatialize doppler", 2, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
// fill out channel volumes for both doppler sound source locations
int volumes[CCHANVOLUMES/2];
// left doppler location
g_AudioDevice->SpatializeChannel( volumes, ch->master_vol, source_vec_DL, gain, mono );
// load volumes into channel as crossfade targets
ChannelSetVolTargets( ch, volumes, IFRONT_LEFT, CCHANVOLUMES/2 );
// right doppler location
g_AudioDevice->SpatializeChannel( volumes, ch->master_vol, source_vec_DR, gain, mono );
// load volumes into channel as crossfade targets
ChannelSetVolTargets( ch, volumes, IFRONT_LEFTD, CCHANVOLUMES/2 );
}
else
{
// fill out channel volumes for single sound source location
int volumes[CCHANVOLUMES/2];
g_AudioDevice->SpatializeChannel( volumes, ch->master_vol, source_vec, gain, mono );
// Special case for stereo sounds originating from player in surround mode
// and special case for musci: remap volumes directly to channels.
RemapPlayerOrMusicVols( ch, volumes, fplayersound, fmusicsound, mono );
// load volumes into channel as crossfade volume targets
ChannelSetVolTargets( ch, volumes, IFRONT_LEFT, CCHANVOLUMES/2 );
}
// prevent left/right/front/rear/center volumes from changing too quickly & producing pops
ChannelUpdateVolXfade( ch );
// end of first time spatializing sound
if ( SND_IsInGame() || toolframework->InToolMode() )
{
ch->flags.bfirstpass = false;
}
// calculate total volume for display later
ch->last_vol = gain * (ch->master_vol/255.0);
return;
ClearAllVolumes:
// Clear all volumes and return.
// This shuts the sound off permanently.
ChannelClearVolumes( ch );
// end of first time spatializing sound
ch->flags.bfirstpass = false;
}
ConVar snd_defer_trace("snd_defer_trace","1");
void SND_SpatializeFirstFrameNoTrace( channel_t *pChannel)
{
if ( snd_defer_trace.GetBool() )
{
// set up tracing state to be non-obstructed
pChannel->flags.bfirstpass = false;
pChannel->flags.bTraced = true;
pChannel->ob_gain = 1.0;
pChannel->ob_gain_inc = 1.0;
pChannel->ob_gain_target = 1.0;
// now spatialize without tracing
SND_Spatialize(pChannel);
// now reset tracing state to firstpass so the trace gets done on next spatialize
pChannel->ob_gain = 0.0;
pChannel->ob_gain_inc = 0.0;
pChannel->ob_gain_target = 0.0;
pChannel->flags.bfirstpass = true;
pChannel->flags.bTraced = false;
}
else
{
pChannel->ob_gain = 0.0;
pChannel->ob_gain_inc = 0.0;
pChannel->ob_gain_target = 0.0;
pChannel->flags.bfirstpass = true;
pChannel->flags.bTraced = false;
SND_Spatialize(pChannel);
}
}
// search through all channels for a channel that matches this
// soundsource, entchannel and sfx, and perform alteration on channel
// as indicated by 'flags' parameter. If shut down request and
// sfx contains a sentence name, shut off the sentence.
// returns TRUE if sound was altered,
// returns FALSE if sound was not found (sound is not playing)
int S_AlterChannel( int soundsource, int entchannel, CSfxTable *sfx, int vol, int pitch, int flags )
{
THREAD_LOCK_SOUND();
int ch_idx;
const char *name = sfx->getname();
if ( name && TestSoundChar( name, CHAR_SENTENCE ) )
{
// This is a sentence name.
// For sentences: assume that the entity is only playing one sentence
// at a time, so we can just shut off
// any channel that has ch->isentence >= 0 and matches the
// soundsource.
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
ch_idx = list.GetChannelIndex(i);
if (channels[ch_idx].soundsource == soundsource
&& channels[ch_idx].entchannel == entchannel
&& channels[ch_idx].sfx != NULL )
{
if (flags & SND_CHANGE_PITCH)
channels[ch_idx].basePitch = pitch;
if (flags & SND_CHANGE_VOL)
channels[ch_idx].master_vol = vol;
if (flags & SND_STOP)
{
S_FreeChannel(&channels[ch_idx]);
}
return TRUE;
}
}
// channel not found
return FALSE;
}
// regular sound or streaming sound
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
bool bSuccess = false;
for ( int i = 0; i < list.Count(); i++ )
{
ch_idx = list.GetChannelIndex(i);
if ( channels[ch_idx].soundsource == soundsource &&
( ( flags & SND_IGNORE_NAME ) ||
( channels[ch_idx].entchannel == entchannel && channels[ch_idx].sfx == sfx ) ) )
{
if (flags & SND_CHANGE_PITCH)
channels[ch_idx].basePitch = pitch;
if (flags & SND_CHANGE_VOL)
channels[ch_idx].master_vol = vol;
if (flags & SND_STOP)
{
S_FreeChannel(&channels[ch_idx]);
}
if ( ( flags & SND_IGNORE_NAME ) == 0 )
return TRUE;
else
bSuccess = true;
}
}
return ( bSuccess ) ? ( TRUE ) : ( FALSE );
}
// set channel flags during initialization based on
// source name
void S_SetChannelWavtype( channel_t *target_chan, CSfxTable *pSfx )
{
// if 1st or 2nd character of name is CHAR_DRYMIX, sound should be mixed dry with no dsp (ie: music)
if ( TestSoundChar(pSfx->getname(), CHAR_DRYMIX) )
target_chan->flags.bdry = true;
else
target_chan->flags.bdry = false;
if ( TestSoundChar(pSfx->getname(), CHAR_FAST_PITCH) )
target_chan->flags.bfast_pitch = true;
else
target_chan->flags.bfast_pitch = false;
// get sound spatialization encoding
target_chan->wavtype = 0;
if ( TestSoundChar( pSfx->getname(), CHAR_DOPPLER ))
target_chan->wavtype = CHAR_DOPPLER;
if ( TestSoundChar( pSfx->getname(), CHAR_DIRECTIONAL ))
target_chan->wavtype = CHAR_DIRECTIONAL;
if ( TestSoundChar( pSfx->getname(), CHAR_DISTVARIANT ))
target_chan->wavtype = CHAR_DISTVARIANT;
if ( TestSoundChar( pSfx->getname(), CHAR_OMNI ))
target_chan->wavtype = CHAR_OMNI;
if ( TestSoundChar( pSfx->getname(), CHAR_SPATIALSTEREO ))
target_chan->wavtype = CHAR_SPATIALSTEREO;
}
// Sets bstereowav flag in channel if source is true stere wav
// sets default wavtype for stereo wavs to CHAR_DISTVARIANT -
// ie: sound varies with distance (left is close, right is far)
// Must be called after S_SetChannelWavtype
void S_SetChannelStereo( channel_t *target_chan, CAudioSource *pSource )
{
if ( !pSource )
{
target_chan->flags.bstereowav = false;
return;
}
// returns true only if source data is a stereo wav file.
// ie: mp3, voice, sentence are all excluded.
target_chan->flags.bstereowav = pSource->IsStereoWav();
// Default stereo wavtype:
// just player standard stereo wavs on player entity - no override.
if ( IsSoundSourceLocalPlayer( target_chan->soundsource ) )
return;
// default wavtype for stereo wavs is OMNI - except for drymix or sounds with 0 attenuation
if ( target_chan->flags.bstereowav && !target_chan->wavtype && !target_chan->flags.bdry && target_chan->dist_mult )
// target_chan->wavtype = CHAR_DISTVARIANT;
target_chan->wavtype = CHAR_OMNI;
}
// =======================================================================
// Channel volume management routines:
// channel volumes crossfade between values over time
// to prevent pops due to rapid spatialization changes
// =======================================================================
// return true if all volumes and target volumes for channel are less/equal to 'vol'
bool BChannelLowVolume( channel_t *pch, int vol_min )
{
int max = -1;
int max_target = -1;
int vol;
int vol_target;
for (int i = 0; i < CCHANVOLUMES; i++)
{
vol = (int)(pch->fvolume[i]);
vol_target = (int)(pch->fvolume_target[i]);
if (vol > max)
max = vol;
if (vol_target > max_target)
max_target = vol_target;
}
return (max <= vol_min && max_target <= vol_min);
}
// Get the loudest actual volume for a channel (not counting targets).
float ChannelLoudestCurVolume( const channel_t * RESTRICT pch )
{
float loudest = pch->fvolume[0];
for (int i = 1; i < CCHANVOLUMES; i++)
{
loudest = fpmax(loudest, pch->fvolume[i]);
}
return loudest;
}
// clear all volumes, targets, crossfade increments
void ChannelClearVolumes( channel_t *pch )
{
for (int i = 0; i < CCHANVOLUMES; i++)
{
pch->fvolume[i] = 0.0;
pch->fvolume_target[i] = 0.0;
pch->fvolume_inc[i] = 0.0;
}
}
// return current volume as integer
int ChannelGetVol( channel_t *pch, int ivol )
{
Assert(ivol < CCHANVOLUMES);
return (int)(pch->fvolume[ivol]);
}
// return maximum current output volume
int ChannelGetMaxVol( channel_t *pch )
{
float max = 0.0;
for (int i = 0; i < CCHANVOLUMES; i++)
{
if (pch->fvolume[i] > max)
max = pch->fvolume[i];
}
return (int)max;
}
// set current volume (clears crossfading - instantaneous value change)
void ChannelSetVol( channel_t *pch, int ivol, int vol )
{
Assert(ivol < CCHANVOLUMES);
pch->fvolume[ivol] = (float)(clamp(vol, 0, 255));
pch->fvolume_target[ivol] = pch->fvolume[ivol];
pch->fvolume_inc[ivol] = 0.0;
}
// copy current channel volumes into target array, starting at ivol, copying cvol entries
void ChannelCopyVolumes( channel_t *pch, int *pvolume_dest, int ivol_start, int cvol )
{
Assert (ivol_start < CCHANVOLUMES);
Assert (ivol_start + cvol <= CCHANVOLUMES);
for (int i = 0; i < cvol; i++)
pvolume_dest[i] = (int)(pch->fvolume[i + ivol_start]);
}
// volume has hit target, shut off crossfading increment
inline void ChannelStopVolXfade( channel_t *pch, int ivol )
{
pch->fvolume[ivol] = pch->fvolume_target[ivol];
pch->fvolume_inc[ivol] = 0.0;
}
#define VOL_XFADE_TIME 0.070 // channel volume crossfade time in seconds
#define VOL_INCR_MAX 20.0 // never change volume by more than +/-N units per frame
// set volume target and volume increment (for crossfade) for channel & speaker
void ChannelSetVolTarget( channel_t *pch, int ivol, int volume_target )
{
float frametime = g_pSoundServices->GetHostFrametime();
float speed;
float vol_target = (float)(clamp(volume_target, 0, 255));
float vol_current;
Assert(ivol < CCHANVOLUMES);
// set volume target
pch->fvolume_target[ivol] = vol_target;
// current volume
vol_current = pch->fvolume[ivol];
// if first time spatializing, set target = volume with no crossfade
// if current & target volumes are close - don't bother crossfading
if ( pch->flags.bfirstpass || (fabs(vol_target - vol_current) < 5.0))
{
// set current volume = target, no increment
ChannelStopVolXfade( pch, ivol);
return;
}
// get crossfade increment 'speed' (volume change per frame)
speed = ( frametime / VOL_XFADE_TIME ) * (vol_target - vol_current);
// make sure we never increment by more than +/- VOL_INCR_MAX volume units per frame
speed = clamp(speed, (float) -VOL_INCR_MAX, (float) VOL_INCR_MAX);
pch->fvolume_inc[ivol] = speed;
}
// set volume targets, using array pvolume as source volumes.
// set into channel volumes starting at ivol_offset index
// set cvol volumes
void ChannelSetVolTargets( channel_t *pch, int *pvolumes, int ivol_offset, int cvol )
{
int volume_target;
Assert(ivol_offset + cvol <= CCHANVOLUMES);
for (int i = 0; i < cvol; i++)
{
volume_target = pvolumes[i];
ChannelSetVolTarget( pch, ivol_offset + i, volume_target );
}
}
// Call once per frame, per channel:
// update all volume crossfades, from fvolume -> fvolume_target
// if current volume reaches target, set increment to 0
void ChannelUpdateVolXfade( channel_t *pch )
{
float fincr;
for (int i = 0; i < CCHANVOLUMES; i++)
{
fincr = pch->fvolume_inc[i];
if (fincr != 0.0)
{
pch->fvolume[i] += fincr;
// test for hit target
if (fincr > 0.0)
{
if (pch->fvolume[i] >= pch->fvolume_target[i])
ChannelStopVolXfade( pch, i );
}
else
{
if (pch->fvolume[i] <= pch->fvolume_target[i])
ChannelStopVolXfade( pch, i );
}
}
}
}
// =======================================================================
// S_StartDynamicSound
// =======================================================================
// Start a sound effect for the given entity on the given channel (ie; voice, weapon etc).
// Try to grab a channel out of the 8 dynamic spots available.
// Currently used for looping sounds, streaming sounds, sentences, and regular entity sounds.
// NOTE: volume is 0.0 - 1.0 and attenuation is 0.0 - 1.0 when passed in.
// Pitch changes playback pitch of wave by % above or below 100. Ignored if pitch == 100
// NOTE: it's not a good idea to play looping sounds through StartDynamicSound, because
// if the looping sound starts out of range, or is bumped from the buffer by another sound
// it will never be restarted. Use StartStaticSound (pass CHAN_STATIC to EMIT_SOUND or
// SV_StartSound.
int S_StartDynamicSound( StartSoundParams_t& params )
{
Assert( params.staticsound == false );
channel_t *target_chan;
int vol;
if ( !g_AudioDevice || !g_AudioDevice->IsActive())
return 0;
if (!params.pSfx)
return 0;
// For debugging to see the actual name of the sound...
char sndname[ MAX_OSPATH ];
Q_strncpy( sndname, params.pSfx->getname(), sizeof( sndname ) );
// Msg("Start sound %s\n", pSfx->getname() );
// override the entchannel to CHAN_STREAM if this is a
// non-voice stream sound.
if ( TestSoundChar(sndname, CHAR_STREAM ) && params.entchannel != CHAN_VOICE && params.entchannel != CHAN_VOICE2 )
params.entchannel = CHAN_STREAM;
vol = params.fvol*255;
if (vol > 255)
{
DevMsg("S_StartDynamicSound: %s volume > 255", sndname );
vol = 255;
}
THREAD_LOCK_SOUND();
if ( params.flags & (SND_STOP|SND_CHANGE_VOL|SND_CHANGE_PITCH) )
{
if ( S_AlterChannel( params.soundsource, params.entchannel, params.pSfx, vol, params.pitch, params.flags) )
return 0;
if ( params.flags & SND_STOP )
return 0;
// fall through - if we're not trying to stop the sound,
// and we didn't find it (it's not playing), go ahead and start it up
}
if (params.pitch == 0)
{
DevMsg ("Warning: S_StartDynamicSound (%s) Ignored, called with pitch 0\n", sndname );
return 0;
}
// pick a channel to play on
target_chan = SND_PickDynamicChannel(params.soundsource, params.entchannel, params.origin, params.pSfx, params.delay, (params.flags & SND_DO_NOT_OVERWRITE_EXISTING_ON_CHANNEL) != 0 );
if ( !target_chan )
return 0;
int channelIndex = (int)( target_chan - channels );
g_AudioDevice->ChannelReset( params.soundsource, channelIndex, target_chan->dist_mult );
#ifdef DEBUG_CHANNELS
{
char szTmp[128];
Q_snprintf(szTmp, sizeof( szTmp ), "Sound %s playing on Dynamic game channel %d\n", sndname, IWavstreamOfCh(target_chan));
Plat_DebugString(szTmp);
}
#endif
bool bIsSentence = TestSoundChar( sndname, CHAR_SENTENCE );
SND_ActivateChannel( target_chan );
ChannelClearVolumes( target_chan );
target_chan->userdata = params.userdata;
target_chan->initialStreamPosition = params.initialStreamPosition;
VectorCopy(params.origin, target_chan->origin);
VectorCopy(params.direction, target_chan->direction);
// never update positions if source entity is 0
target_chan->flags.bUpdatePositions = params.bUpdatePositions && (params.soundsource == 0 ? 0 : 1);
// reference_dist / (reference_power_level / actual_power_level)
target_chan->flags.m_bCompatibilityAttenuation = SNDLEVEL_IS_COMPATIBILITY_MODE( params.soundlevel );
if ( target_chan->flags.m_bCompatibilityAttenuation )
{
// Translate soundlevel from its 'encoded' value to a real soundlevel that we can use in the sound system.
params.soundlevel = SNDLEVEL_FROM_COMPATIBILITY_MODE( params.soundlevel );
}
target_chan->dist_mult = SNDLVL_TO_DIST_MULT( params.soundlevel );
S_SetChannelWavtype( target_chan, params.pSfx );
target_chan->master_vol = vol;
target_chan->soundsource = params.soundsource;
target_chan->entchannel = params.entchannel;
target_chan->basePitch = params.pitch;
target_chan->flags.isSentence = false;
target_chan->radius = 0;
target_chan->sfx = params.pSfx;
target_chan->special_dsp = params.specialdsp;
target_chan->flags.fromserver = params.fromserver;
target_chan->flags.bSpeaker = (params.flags & SND_SPEAKER) ? 1 : 0;
target_chan->speakerentity = params.speakerentity;
target_chan->flags.m_bShouldPause = (params.flags & SND_SHOULDPAUSE) ? 1 : 0;
// initialize dsp room mixing params
target_chan->dsp_mix_min = -1;
target_chan->dsp_mix_max = -1;
CAudioSource *pSource = NULL;
if ( bIsSentence )
{
// this is a sentence
// link all words and load the first word
// NOTE: sentence names stored in the cache lookup are
// prepended with a '!'. Sentence names stored in the
// sentence file do not have a leading '!'.
VOX_LoadSound( target_chan, PSkipSoundChars( sndname ) );
}
else
{
// regular or streamed sound fx
pSource = S_LoadSound( params.pSfx, target_chan );
if ( pSource && !IsValidSampleRate( pSource->SampleRate() ) )
{
Warning( "*** Invalid sample rate (%d) for sound '%s'.\n", pSource->SampleRate(), sndname );
}
if ( !pSource && !params.pSfx->m_bIsLateLoad )
{
Warning( "Failed to load sound \"%s\", file probably missing from disk/repository\n", sndname );
}
}
if (!target_chan->pMixer)
{
// couldn't load the sound's data, or sentence has 0 words (this is not an error)
S_FreeChannel( target_chan );
return 0;
}
int nSndShowStart = snd_showstart.GetInt();
// TODO: Support looping sounds through speakers.
// If the sound is from a speaker, and it's looping, ignore it.
if ( target_chan->flags.bSpeaker )
{
if ( params.pSfx->pSource && params.pSfx->pSource->IsLooped() )
{
if (nSndShowStart > 0 && nSndShowStart < 7 && nSndShowStart != 4)
{
DevMsg("DynamicSound : Speaker ignored looping sound: %s\n", sndname );
}
S_FreeChannel( target_chan );
return 0;
}
}
S_SetChannelStereo( target_chan, pSource );
if (nSndShowStart == 5)
{
snd_showstart.SetValue(6); // debug: show gain for next spatialize only
nSndShowStart = 6;
}
// get sound type before we spatialize
MXR_GetMixGroupFromSoundsource( target_chan, params.soundsource, params.soundlevel );
// skip the trace on the first spatialization. This channel may be stolen
// by another sound played this frame. Defer the trace to the mix loop
SND_SpatializeFirstFrameNoTrace(target_chan);
if (nSndShowStart > 0 && nSndShowStart < 7 && nSndShowStart != 4)
{
channel_t *pTargetChan = target_chan;
DevMsg( "DynamicSound %s : src %d : channel %d : %d dB : vol %.2f : time %.3f\n", sndname, params.soundsource, params.entchannel, params.soundlevel, params.fvol, g_pSoundServices->GetHostTime() );
if (nSndShowStart == 2 || nSndShowStart == 5)
DevMsg( "\t dspmix %1.2f : distmix %1.2f : dspface %1.2f : lvol %1.2f : cvol %1.2f : rvol %1.2f : rlvol %1.2f : rrvol %1.2f\n",
pTargetChan->dspmix, pTargetChan->distmix, pTargetChan->dspface,
pTargetChan->fvolume[IFRONT_LEFT], pTargetChan->fvolume[IFRONT_CENTER], pTargetChan->fvolume[IFRONT_RIGHT], pTargetChan->fvolume[IREAR_LEFT], pTargetChan->fvolume[IREAR_RIGHT] );
if (nSndShowStart == 3)
DevMsg( "\t x: %4f y: %4f z: %4f\n", pTargetChan->origin.x, pTargetChan->origin.y, pTargetChan->origin.z );
if ( snd_visualize.GetInt() )
{
CDebugOverlay::AddTextOverlay( pTargetChan->origin, 2.0f, sndname );
}
}
// If a client can't hear a sound when they FIRST receive the StartSound message,
// the client will never be able to hear that sound. This is so that out of
// range sounds don't fill the playback buffer. For streaming sounds, we bypass this optimization.
if ( BChannelLowVolume( target_chan, 0 ) && !toolframework->IsToolRecording() )
{
// Looping sounds don't use this optimization because they should stick around until they're killed.
// Also bypass for speech (GetSentence)
if ( !params.pSfx->pSource || (!params.pSfx->pSource->IsLooped() && !params.pSfx->pSource->GetSentence()) )
{
// if this is long sound, play the whole thing.
if (!SND_IsLongWave( target_chan ))
{
// DevMsg("S_StartDynamicSound: spatialized to 0 vol & ignored %s", sndname);
S_FreeChannel( target_chan );
return 0; // not audible at all
}
}
}
// Init client entity mouth movement vars
target_chan->flags.m_bIgnorePhonemes = ( params.flags & SND_IGNORE_PHONEMES ) != 0;
SND_InitMouth(target_chan);
if ( IsX360() && params.delay < 0 )
{
params.delay = 0;
target_chan->flags.delayed_start = true;
}
// Pre-startup delay. Compute # of samples over which to mix in zeros from data source before
// actually reading first set of samples
if ( params.delay != 0.0f )
{
Assert( target_chan->sfx );
Assert( target_chan->sfx->pSource );
// delay count is computed at the sampling rate of the source because the output rate will
// match the source rate when the sound is mixed
float rate = target_chan->sfx->pSource->SampleRate();
int delaySamples = (int)( params.delay * rate );
if ( params.delay > 0 )
{
target_chan->pMixer->SetStartupDelaySamples( delaySamples );
target_chan->flags.delayed_start = true;
}
else
{
int skipSamples = -delaySamples;
int totalSamples = target_chan->sfx->pSource->SampleCount();
if ( target_chan->sfx->pSource->IsLooped() )
{
skipSamples = skipSamples % totalSamples;
}
if ( skipSamples >= totalSamples )
{
S_FreeChannel( target_chan );
return 0;
}
target_chan->pitch = target_chan->basePitch * 0.01f;
target_chan->pMixer->SkipSamples( target_chan, skipSamples, rate, 0 );
target_chan->ob_gain_target = 1.0f;
target_chan->ob_gain = 1.0f;
target_chan->ob_gain_inc = 0.0;
target_chan->flags.bfirstpass = false;
target_chan->flags.delayed_start = true;
}
}
g_pSoundServices->OnSoundStarted( target_chan->guid, params, sndname );
return target_chan->guid;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : *name -
// Output : CSfxTable
//-----------------------------------------------------------------------------
CSfxTable *S_DummySfx( const char *name )
{
dummySfx.setname( name );
return &dummySfx;
}
/*
=================
S_StartStaticSound
=================
Start playback of a sound, loaded into the static portion of the channel array.
Currently, this should be used for looping ambient sounds, looping sounds
that should not be interrupted until complete, non-creature sentences,
and one-shot ambient streaming sounds. Can also play 'regular' sounds one-shot,
in case designers want to trigger regular game sounds.
Pitch changes playback pitch of wave by % above or below 100. Ignored if pitch == 100
NOTE: volume is 0.0 - 1.0 and attenuation is 0.0 - 1.0 when passed in.
*/
int S_StartStaticSound( StartSoundParams_t& params )
{
Assert( params.staticsound == true );
channel_t *ch;
CAudioSource *pSource = NULL;
if ( !g_AudioDevice->IsActive() )
return 0;
if ( !params.pSfx )
return 0;
// For debugging to see the actual name of the sound...
char sndname[ MAX_OSPATH ];
Q_strncpy( sndname, params.pSfx->getname(), sizeof( sndname ) );
// Msg("Start static sound %s\n", pSfx->getname() );
int vol = params.fvol * 255;
if ( vol > 255 )
{
DevMsg( "S_StartStaticSound: %s volume > 255", sndname );
vol = 255;
}
int nSndShowStart = snd_showstart.GetInt();
if ((params.flags & SND_STOP) && nSndShowStart > 0)
DevMsg("S_StartStaticSound: %s Stopped.\n", sndname);
if ((params.flags & SND_STOP) || (params.flags & SND_CHANGE_VOL) || (params.flags & SND_CHANGE_PITCH))
{
if (S_AlterChannel(params.soundsource, params.entchannel, params.pSfx, vol, params.pitch, params.flags) || (params.flags & SND_STOP))
return 0;
}
if ( params.pitch == 0 )
{
DevMsg( "Warning: S_StartStaticSound Ignored, called with pitch 0\n");
return 0;
}
// First, make sure the sound source entity is even in the PVS.
float flSoundRadius = 0.0f;
bool looping = false;
/*
CAudioSource *pSource = pSfx ? pSfx->pSource : NULL;
if ( pSource )
{
looping = pSource->IsLooped();
}
*/
SpatializationInfo_t si;
si.info.Set(
params.soundsource,
params.entchannel,
params.pSfx ? sndname : "",
params.origin,
params.direction,
vol,
params.soundlevel,
looping,
params.pitch,
listener_origin,
params.speakerentity );
si.type = SpatializationInfo_t::SI_INCREATION;
si.pOrigin = NULL;
si.pAngles = NULL;
si.pflRadius = &flSoundRadius;
g_pSoundServices->GetSoundSpatialization( params.soundsource, si );
// pick a channel to play on from the static area
THREAD_LOCK_SOUND();
ch = SND_PickStaticChannel(params.soundsource, params.pSfx); // Autolooping sounds are always fixed origin(?)
if ( !ch )
return 0;
SND_ActivateChannel( ch );
ChannelClearVolumes( ch );
ch->userdata = params.userdata;
ch->initialStreamPosition = params.initialStreamPosition;
if ( ch->userdata != 0 )
{
g_pSoundServices->GetToolSpatialization( ch->userdata, ch->guid, si );
}
int channelIndex = ch - channels;
g_AudioDevice->ChannelReset( params.soundsource, channelIndex, ch->dist_mult );
#ifdef DEBUG_CHANNELS
{
char szTmp[128];
Q_snprintf(szTmp, sizeof( szTmp ), "Sound %s playing on Static game channel %d\n", sfxin->name, IWavstreamOfCh(ch));
Plat_DebugString(szTmp);
}
#endif
if ( TestSoundChar(sndname, CHAR_SENTENCE) )
{
// this is a sentence. link words to play in sequence.
// NOTE: sentence names stored in the cache lookup are
// prepended with a '!'. Sentence names stored in the
// sentence file do not have a leading '!'.
// link all words and load the first word
VOX_LoadSound( ch, PSkipSoundChars(sndname) );
}
else
{
// load regular or stream sound
pSource = S_LoadSound( params.pSfx, ch );
if ( pSource && !IsValidSampleRate( pSource->SampleRate() ) )
{
Warning( "*** Invalid sample rate (%d) for sound '%s'.\n", pSource->SampleRate(), sndname );
}
if ( !pSource && !params.pSfx->m_bIsLateLoad )
{
Warning( "Failed to load sound \"%s\", file probably missing from disk/repository\n", sndname );
}
ch->sfx = params.pSfx;
ch->flags.isSentence = false;
}
if ( !ch->pMixer )
{
// couldn't load sounds' data, or sentence has 0 words (not an error)
S_FreeChannel( ch );
return 0;
}
VectorCopy (params.origin, ch->origin);
VectorCopy (params.direction, ch->direction);
// never update positions if source entity is 0
ch->flags.bUpdatePositions = params.bUpdatePositions && (params.soundsource == 0 ? 0 : 1);
ch->master_vol = vol;
ch->flags.m_bCompatibilityAttenuation = SNDLEVEL_IS_COMPATIBILITY_MODE( params.soundlevel );
if ( ch->flags.m_bCompatibilityAttenuation )
{
// Translate soundlevel from its 'encoded' value to a real soundlevel that we can use in the sound system.
params.soundlevel = SNDLEVEL_FROM_COMPATIBILITY_MODE( params.soundlevel );
}
ch->dist_mult = SNDLVL_TO_DIST_MULT( params.soundlevel );
S_SetChannelWavtype( ch, params.pSfx );
ch->basePitch = params.pitch;
ch->soundsource = params.soundsource;
ch->entchannel = params.entchannel;
ch->special_dsp = params.specialdsp;
ch->flags.fromserver = params.fromserver;
ch->flags.bSpeaker = (params.flags & SND_SPEAKER) ? 1 : 0;
ch->speakerentity = params.speakerentity;
ch->flags.m_bShouldPause = (params.flags & SND_SHOULDPAUSE) ? 1 : 0;
// TODO: Support looping sounds through speakers.
// If the sound is from a speaker, and it's looping, ignore it.
if ( ch->flags.bSpeaker )
{
if ( params.pSfx->pSource && params.pSfx->pSource->IsLooped() )
{
if (nSndShowStart > 0 && nSndShowStart < 7 && nSndShowStart != 4)
{
DevMsg("StaticSound : Speaker ignored looping sound: %s\n", sndname);
}
S_FreeChannel( ch );
return 0;
}
}
// set the default radius
ch->radius = flSoundRadius;
S_SetChannelStereo( ch, pSource );
// initialize dsp room mixing params
ch->dsp_mix_min = -1;
ch->dsp_mix_max = -1;
if (nSndShowStart == 5)
{
snd_showstart.SetValue(6); // display gain once only
nSndShowStart = 6;
}
// get sound type before we spatialize
MXR_GetMixGroupFromSoundsource( ch, params.soundsource, params.soundlevel );
// skip the trace on the first spatialization. This channel may be stolen
// by another sound played this frame. Defer the trace to the mix loop
SND_SpatializeFirstFrameNoTrace(ch);
// Init client entity mouth movement vars
ch->flags.m_bIgnorePhonemes = ( params.flags & SND_IGNORE_PHONEMES ) != 0;
SND_InitMouth( ch );
if ( IsX360() && params.delay < 0 )
{
// X360TEMP: Can't support yet, but going to.
params.delay = 0;
}
// Pre-startup delay. Compute # of samples over which to mix in zeros from data source before
// actually reading first set of samples
if ( params.delay != 0.0f )
{
Assert( ch->sfx );
Assert( ch->sfx->pSource );
float rate = ch->sfx->pSource->SampleRate();
int delaySamples = (int)( params.delay * rate * params.pitch * 0.01f );
ch->pMixer->SetStartupDelaySamples( delaySamples );
if ( params.delay > 0 )
{
ch->pMixer->SetStartupDelaySamples( delaySamples );
ch->flags.delayed_start = true;
}
else
{
int skipSamples = -delaySamples;
int totalSamples = ch->sfx->pSource->SampleCount();
if ( ch->sfx->pSource->IsLooped() )
{
skipSamples = skipSamples % totalSamples;
}
if ( skipSamples >= totalSamples )
{
S_FreeChannel( ch );
return 0;
}
ch->pitch = ch->basePitch * 0.01f;
ch->pMixer->SkipSamples( ch, skipSamples, rate, 0 );
ch->ob_gain_target = 1.0f;
ch->ob_gain = 1.0f;
ch->ob_gain_inc = 0.0f;
ch->flags.bfirstpass = false;
}
}
if ( S_IsMusic( ch ) )
{
// See if we have "music" of same name playing from "world" which means we save/restored this sound already. If so,
// kill the new version and update the soundsource
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = list.GetChannel(i);
// Don't mess with the channel we just created, of course
if ( ch == pChannel )
continue;
if ( ch->sfx != pChannel->sfx )
continue;
if ( pChannel->soundsource != SOUND_FROM_WORLD )
continue;
if ( !S_IsMusic( pChannel ) )
continue;
DevMsg( 1, "Hooking duplicate restored song track %s\n", sndname );
// the new channel will have an updated soundsource and probably
// has an updated pitch or volume since we are receiving this sound message
// after the sound has started playing (usually a volume change)
// copy that data out of the source
pChannel->soundsource = ch->soundsource;
pChannel->master_vol = ch->master_vol;
pChannel->basePitch = ch->basePitch;
pChannel->pitch = ch->pitch;
S_FreeChannel( ch );
return 0;
}
}
g_pSoundServices->OnSoundStarted( ch->guid, params, sndname );
if (nSndShowStart > 0 && nSndShowStart < 7 && nSndShowStart != 4)
{
DevMsg( "StaticSound %s : src %d : channel %d : %d dB : vol %.2f : radius %.0f : time %.3f\n", sndname, params.soundsource, params.entchannel, params.soundlevel, params.fvol, flSoundRadius, g_pSoundServices->GetHostTime() );
if (nSndShowStart == 2 || nSndShowStart == 5)
DevMsg( "\t dspmix %1.2f : distmix %1.2f : dspface %1.2f : lvol %1.2f : cvol %1.2f : rvol %1.2f : rlvol %1.2f : rrvol %1.2f\n",
ch->dspmix, ch->distmix, ch->dspface,
ch->fvolume[IFRONT_LEFT], ch->fvolume[IFRONT_CENTER], ch->fvolume[IFRONT_RIGHT], ch->fvolume[IREAR_LEFT], ch->fvolume[IREAR_RIGHT] );
if (nSndShowStart == 3)
DevMsg( "\t x: %4f y: %4f z: %4f\n", ch->origin.x, ch->origin.y, ch->origin.z );
}
return ch->guid;
}
#ifdef STAGING_ONLY
static ConVar snd_filter( "snd_filter", "", FCVAR_CHEAT );
#endif // STAGING_ONLY
int S_StartSound( StartSoundParams_t& params )
{
if( ! params.pSfx )
{
return 0;
}
#ifdef STAGING_ONLY
if ( snd_filter.GetString()[ 0 ] && !Q_stristr( params.pSfx->getname(), snd_filter.GetString() ) )
{
return 0;
}
#endif // STAGING_ONLY
if ( IsX360() && params.delay < 0 && !params.initialStreamPosition && params.pSfx )
{
// calculate an initial stream position from the expected sample position
float rate = params.pSfx->pSource->SampleRate();
int samplePosition = (int)( -params.delay * rate * params.pitch * 0.01f );
params.initialStreamPosition = params.pSfx->pSource->SampleToStreamPosition( samplePosition );
}
if ( params.staticsound )
{
VPROF_( "StartStaticSound", 0, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
return S_StartStaticSound( params );
}
else
{
VPROF_( "StartDynamicSound", 0, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
return S_StartDynamicSound( params );
}
}
// Restart all the sounds on the specified channel
inline bool IsChannelLooped( int iChannel )
{
return (channels[iChannel].sfx &&
channels[iChannel].sfx->pSource &&
channels[iChannel].sfx->pSource->IsLooped() );
}
int S_GetCurrentStaticSounds( SoundInfo_t *pResult, int nSizeResult, int entchannel )
{
int nSpaceRemaining = nSizeResult;
for (int i = MAX_DYNAMIC_CHANNELS; i < total_channels && nSpaceRemaining; i++)
{
if ( channels[i].entchannel == entchannel && channels[i].sfx )
{
pResult->Set( channels[i].soundsource,
channels[i].entchannel,
channels[i].sfx->getname(),
channels[i].origin,
channels[i].direction,
( (float)channels[i].master_vol / 255.0 ),
DIST_MULT_TO_SNDLVL( channels[i].dist_mult ),
IsChannelLooped( i ),
channels[i].basePitch,
listener_origin,
channels[i].speakerentity );
pResult++;
nSpaceRemaining--;
}
}
return (nSizeResult - nSpaceRemaining);
}
// Stop all sounds for entity on a channel.
void S_StopSound(int soundsource, int entchannel)
{
THREAD_LOCK_SOUND();
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = list.GetChannel(i);
if (pChannel->soundsource == soundsource
&& pChannel->entchannel == entchannel)
{
S_FreeChannel( pChannel );
}
}
}
channel_t *S_FindChannelByGuid( int guid )
{
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = list.GetChannel(i);
if ( pChannel->guid == guid )
{
return pChannel;
}
}
return NULL;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : guid -
//-----------------------------------------------------------------------------
void S_StopSoundByGuid( int guid )
{
THREAD_LOCK_SOUND();
channel_t *pChannel = S_FindChannelByGuid( guid );
if ( pChannel )
{
S_FreeChannel( pChannel );
}
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : guid -
//-----------------------------------------------------------------------------
float S_SoundDurationByGuid( int guid )
{
channel_t *pChannel = S_FindChannelByGuid( guid );
if ( !pChannel || !pChannel->sfx )
return 0.0f;
// NOTE: Looping sounds will return the length of a single loop
// Use S_IsLoopingSoundByGuid to see if they are looped
float flRate = pChannel->sfx->pSource->SampleRate() * pChannel->basePitch * 0.01f;
int nTotalSamples = pChannel->sfx->pSource->SampleCount();
return (flRate != 0.0f) ? nTotalSamples / flRate : 0.0f;
}
//-----------------------------------------------------------------------------
// Is this sound a looping sound?
//-----------------------------------------------------------------------------
bool S_IsLoopingSoundByGuid( int guid )
{
channel_t *pChannel = S_FindChannelByGuid( guid );
if ( !pChannel || !pChannel->sfx )
return false;
return( pChannel->sfx->pSource->IsLooped() );
}
//-----------------------------------------------------------------------------
// Purpose: Note that the guid is preincremented, so we can just return the current value as the "last sound" indicator
// Input : -
// Output : int
//-----------------------------------------------------------------------------
int S_GetGuidForLastSoundEmitted()
{
return s_nSoundGuid;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : guid -
// Output : Returns true on success, false on failure.
//-----------------------------------------------------------------------------
bool S_IsSoundStillPlaying( int guid )
{
channel_t *pChannel = S_FindChannelByGuid( guid );
return pChannel != NULL ? true : false;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : guid -
// fvol -
//-----------------------------------------------------------------------------
void S_SetVolumeByGuid( int guid, float fvol )
{
channel_t *pChannel = S_FindChannelByGuid( guid );
pChannel->master_vol = 255.0f * clamp( fvol, 0.0f, 1.0f );
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : guid -
// Output : float
//-----------------------------------------------------------------------------
float S_GetElapsedTimeByGuid( int guid )
{
channel_t *pChannel = S_FindChannelByGuid( guid );
if ( !pChannel )
return 0.0f;
CAudioMixer *mixer = pChannel->pMixer;
if ( !mixer )
return 0.0f;
float elapsed = mixer->GetSamplePosition() / ( mixer->GetSource()->SampleRate() * pChannel->pitch * 0.01f );
return elapsed;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : sndlist -
//-----------------------------------------------------------------------------
void S_GetActiveSounds( CUtlVector< SndInfo_t >& sndlist )
{
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *ch = list.GetChannel(i);
SndInfo_t info;
info.m_nGuid = ch->guid;
info.m_filenameHandle = ch->sfx ? ch->sfx->GetFileNameHandle() : NULL;
info.m_nSoundSource = ch->soundsource;
info.m_nChannel = ch->entchannel;
// If a sound is being played through a speaker entity (e.g., on a monitor,), this is the
// entity upon which to show the lips moving, if the sound has sentence data
info.m_nSpeakerEntity = ch->speakerentity;
info.m_flVolume = (float)ch->master_vol / 255.0f;
info.m_flLastSpatializedVolume = ch->last_vol;
// Radius of this sound effect (spatialization is different within the radius)
info.m_flRadius = ch->radius;
info.m_nPitch = ch->basePitch;
info.m_pOrigin = &ch->origin;
info.m_pDirection = &ch->direction;
// if true, assume sound source can move and update according to entity
info.m_bUpdatePositions = ch->flags.bUpdatePositions;
// true if playing linked sentence
info.m_bIsSentence = ch->flags.isSentence;
// if true, bypass all dsp processing for this sound (ie: music)
info.m_bDryMix = ch->flags.bdry;
// true if sound is playing through in-game speaker entity.
info.m_bSpeaker = ch->flags.bSpeaker;
// true if sound is using special DSP effect
info.m_bSpecialDSP = ( ch->special_dsp != 0 );
// for snd_show, networked sounds get colored differently than local sounds
info.m_bFromServer = ch->flags.fromserver;
sndlist.AddToTail( info );
}
}
void S_StopAllSounds( bool bClear )
{
THREAD_LOCK_SOUND();
int i;
if ( !g_AudioDevice )
return;
if ( !g_AudioDevice->IsActive() )
return;
total_channels = MAX_DYNAMIC_CHANNELS; // no statics
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( i = 0; i < list.Count(); i++ )
{
channel_t *pChannel = list.GetChannel(i);
if ( channels[i].sfx )
{
DevMsg( 1, "%2d:Stopped sound %s\n", i, channels[i].sfx->getname() );
}
S_FreeChannel( pChannel );
}
Q_memset( channels, 0, MAX_CHANNELS * sizeof(channel_t) );
if ( bClear )
{
S_ClearBuffer();
}
// Clear any remaining soundfade
memset( &soundfade, 0, sizeof( soundfade ) );
g_AudioDevice->StopAllSounds();
Assert( g_ActiveChannels.GetActiveCount() == 0 );
}
void S_StopAllSoundsC( void )
{
S_StopAllSounds( true );
}
void S_OnLoadScreen( bool value )
{
s_bOnLoadScreen = value;
}
void S_ClearBuffer( void )
{
if ( !g_AudioDevice )
return;
g_AudioDevice->ClearBuffer();
DSP_ClearState();
MIX_ClearAllPaintBuffers( PAINTBUFFER_SIZE, true );
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : percent -
// holdtime -
// intime -
// outtime -
//-----------------------------------------------------------------------------
void S_SoundFade( float percent, float holdtime, float intime, float outtime )
{
soundfade.starttime = g_pSoundServices->GetHostTime();
soundfade.initial_percent = percent;
soundfade.fadeouttime = outtime;
soundfade.holdtime = holdtime;
soundfade.fadeintime = intime;
}
//-----------------------------------------------------------------------------
// Purpose: Modulates sound volume on the client.
//-----------------------------------------------------------------------------
void S_UpdateSoundFade(void)
{
float totaltime;
float f;
// Determine current fade value.
// Assume no fading remains
soundfade.percent = 0;
totaltime = soundfade.fadeouttime + soundfade.fadeintime + soundfade.holdtime;
float elapsed = g_pSoundServices->GetHostTime() - soundfade.starttime;
// Clock wrapped or reset (BUG) or we've gone far enough
if ( elapsed < 0.0f || elapsed >= totaltime || totaltime <= 0.0f )
{
return;
}
// We are in the fade time, so determine amount of fade.
if ( soundfade.fadeouttime > 0.0f && ( elapsed < soundfade.fadeouttime ) )
{
// Ramp up
f = elapsed / soundfade.fadeouttime;
}
// Inside the hold time
else if ( elapsed <= ( soundfade.fadeouttime + soundfade.holdtime ) )
{
// Stay
f = 1.0f;
}
else
{
// Ramp down
f = ( elapsed - ( soundfade.fadeouttime + soundfade.holdtime ) ) / soundfade.fadeintime;
// backward interpolated...
f = 1.0f - f;
}
// Spline it.
f = SimpleSpline( f );
f = clamp( f, 0.0f, 1.0f );
soundfade.percent = soundfade.initial_percent * f;
}
//=============================================================================
// Global Voice Ducker - enabled in vcd scripts, when characters deliver important dialog. Overrides all
// other mixer ducking, and ducks all other sounds except dialog.
ConVar snd_ducktovolume( "snd_ducktovolume", "0.55", FCVAR_ARCHIVE );
ConVar snd_duckerattacktime( "snd_duckerattacktime", "0.5", FCVAR_ARCHIVE );
ConVar snd_duckerreleasetime( "snd_duckerreleasetime", "2.5", FCVAR_ARCHIVE );
ConVar snd_duckerthreshold("snd_duckerthreshold", "0.15", FCVAR_ARCHIVE );
static void S_UpdateVoiceDuck( int voiceChannelCount, int voiceChannelMaxVolume, float frametime )
{
float volume_when_ducked = snd_ducktovolume.GetFloat();
int volume_threshold = (int)(snd_duckerthreshold.GetFloat() * 255.0);
float duckTarget = 1.0;
if ( voiceChannelCount > 0 )
{
voiceChannelMaxVolume = clamp(voiceChannelMaxVolume, 0, 255);
// duckTarget = RemapVal( voiceChannelMaxVolume, 0, 255, 1.0, volume_when_ducked );
// KB: Change: ducker now active if any character is speaking above threshold volume.
// KB: Active ducker drops all volumes to volumes * snd_duckvolume
if ( voiceChannelMaxVolume > volume_threshold )
duckTarget = volume_when_ducked;
}
float rate = ( duckTarget < g_DuckScale ) ? snd_duckerattacktime.GetFloat() : snd_duckerreleasetime.GetFloat();
g_DuckScale = Approach( duckTarget, g_DuckScale, frametime * ((1-volume_when_ducked) / rate) );
}
// set 2d forward vector, given 3d right vector.
// NOTE: this should only be used for a listener forward
// vector from a listener right vector. It is not a general use routine.
void ConvertListenerVectorTo2D( Vector *pvforward, Vector *pvright )
{
// get 2d forward direction vector, ignoring pitch angle
QAngle angles2d;
Vector source2d;
Vector listener_forward2d;
source2d = *pvright;
source2d.z = 0.0;
VectorNormalize(source2d);
// convert right vector to euler angles (yaw & pitch)
VectorAngles(source2d, angles2d);
// get forward angle of listener
angles2d[PITCH] = 0;
angles2d[YAW] += 90; // rotate 90 ccw
angles2d[ROLL] = 0;
if (angles2d[YAW] >= 360)
angles2d[YAW] -= 360;
AngleVectors(angles2d, &listener_forward2d);
VectorNormalize(listener_forward2d);
*pvforward = listener_forward2d;
}
// If this is nonzero, we will only spatialize some of the static
// channels each frame. The round robin will spatialize 1 / (2 ^ x)
// of the spatial channels each frame.
ConVar snd_spatialize_roundrobin( "snd_spatialize_roundrobin", "0", FCVAR_ALLOWED_IN_COMPETITIVE, "Lowend optimization: if nonzero, spatialize only a fraction of sound channels each frame. 1/2^x of channels will be spatialized per frame." );
/*
============
S_Update
Called once each time through the main loop
============
*/
void S_Update( const AudioState_t *pAudioState )
{
VPROF("S_Update");
channel_t *ch;
channel_t *combine;
static unsigned int s_roundrobin = 0 ; ///< number of times this function is called.
///< used instead of host_frame because that number
///< isn't necessarily available here (sez Yahn).
if ( !g_AudioDevice->IsActive() )
return;
g_SndMutex.Lock();
// Update any client side sound fade
S_UpdateSoundFade();
if ( pAudioState )
{
VectorCopy( pAudioState->m_Origin, listener_origin );
AngleVectors( pAudioState->m_Angles, &listener_forward, &listener_right, &listener_up );
s_bIsListenerUnderwater = pAudioState->m_bIsUnderwater;
}
else
{
VectorCopy( vec3_origin, listener_origin );
VectorCopy( vec3_origin, listener_forward );
VectorCopy( vec3_origin, listener_right );
VectorCopy( vec3_origin, listener_up );
s_bIsListenerUnderwater = false;
}
g_AudioDevice->UpdateListener( listener_origin, listener_forward, listener_right, listener_up );
combine = NULL;
int voiceChannelCount = 0;
int voiceChannelMaxVolume = 0;
// reset traceline counter for this frame
g_snd_trace_count = 0;
// calculate distance to nearest walls, update dsp_spatial
// updates one wall only per frame (one trace per frame)
SND_SetSpatialDelays();
// updates dsp_room if automatic room detection enabled
DAS_CheckNewRoomDSP();
// update spatialization for static and dynamic sounds
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
if (snd_spatialize_roundrobin.GetInt() == 0)
{
// spatialize each channel each time
for ( int i = 0; i < list.Count(); i++ )
{
ch = list.GetChannel(i);
Assert(ch->sfx);
Assert(ch->activeIndex > 0);
SND_Spatialize(ch); // respatialize channel
if ( ch->sfx->pSource && ch->sfx->pSource->IsVoiceSource() )
{
voiceChannelCount++;
voiceChannelMaxVolume = max(voiceChannelMaxVolume, ChannelGetMaxVol( ch) );
}
}
}
else // lowend performance improvement: spatialize only some channels each frame.
{
unsigned int robinmask = (1 << snd_spatialize_roundrobin.GetInt()) - 1;
// now do static channels
for ( int i = 0 ; i < list.Count() ; ++i )
{
ch = list.GetChannel(i);
Assert(ch->sfx);
Assert(ch->activeIndex > 0);
// need to check bfirstpass because sound tracing may have been deferred
if ( ch->flags.bfirstpass || (robinmask & s_roundrobin) == ( i & robinmask ) )
{
SND_Spatialize(ch); // respatialize channel
}
if ( ch->sfx->pSource && ch->sfx->pSource->IsVoiceSource() )
{
voiceChannelCount++;
voiceChannelMaxVolume = max( voiceChannelMaxVolume, ChannelGetMaxVol( ch) );
}
}
++s_roundrobin;
}
SND_ChannelTraceReset();
// set new target for voice ducking
float frametime = g_pSoundServices->GetHostFrametime();
S_UpdateVoiceDuck( voiceChannelCount, voiceChannelMaxVolume, frametime );
// update x360 music volume
g_DashboardMusicMixValue = Approach( g_DashboardMusicMixTarget, g_DashboardMusicMixValue, g_DashboardMusicFadeRate * frametime );
//
// debugging output
//
if (snd_show.GetInt())
{
con_nprint_t np;
np.time_to_live = 2.0f;
np.fixed_width_font = true;
int total = 0;
CChannelList activeChannels;
g_ActiveChannels.GetActiveChannels( activeChannels );
for ( int i = 0; i < activeChannels.Count(); i++ )
{
channel_t *channel = activeChannels.GetChannel(i);
if ( !channel->sfx )
continue;
np.index = total + 2;
if ( channel->flags.fromserver )
{
np.color[0] = 1.0;
np.color[1] = 0.8;
np.color[2] = 0.1;
}
else
{
np.color[0] = 0.1;
np.color[1] = 0.9;
np.color[2] = 1.0;
}
unsigned int sampleCount = RemainingSamples( channel );
float timeleft = (float)sampleCount / (float)channel->sfx->pSource->SampleRate();
bool bLooping = channel->sfx->pSource->IsLooped();
if (snd_surround.GetInt() < 4)
{
Con_NXPrintf ( &np, "%02i l(%03d) r(%03d) vol(%03d) ent(%03d) pos(%6d %6d %6d) timeleft(%f) looped(%d) %50s",
total+ 1,
(int)channel->fvolume[IFRONT_LEFT],
(int)channel->fvolume[IFRONT_RIGHT],
channel->master_vol,
channel->soundsource,
(int)channel->origin[0],
(int)channel->origin[1],
(int)channel->origin[2],
timeleft,
bLooping,
channel->sfx->getname());
}
else
{
Con_NXPrintf ( &np, "%02i l(%03d) c(%03d) r(%03d) rl(%03d) rr(%03d) vol(%03d) ent(%03d) pos(%6d %6d %6d) timeleft(%f) looped(%d) %50s",
total+ 1,
(int)channel->fvolume[IFRONT_LEFT],
(int)channel->fvolume[IFRONT_CENTER],
(int)channel->fvolume[IFRONT_RIGHT],
(int)channel->fvolume[IREAR_LEFT],
(int)channel->fvolume[IREAR_RIGHT],
channel->master_vol,
channel->soundsource,
(int)channel->origin[0],
(int)channel->origin[1],
(int)channel->origin[2],
timeleft,
bLooping,
channel->sfx->getname());
}
if ( snd_visualize.GetInt() )
{
CDebugOverlay::AddTextOverlay( channel->origin, 0.05f, channel->sfx->getname() );
}
total++;
}
while ( total <= 128 )
{
Con_NPrintf( total + 2, "" );
total++;
}
}
g_SndMutex.Unlock();
if ( s_bOnLoadScreen )
return;
// not time to update yet?
double tNow = Plat_FloatTime();
// this is the last time we ran a sound frame
g_LastSoundFrame = tNow;
// this is the last time we did mixing (extraupdate also advances this if it mixes)
g_LastMixTime = tNow;
// mix some sound
// try to stay at least one frame + mixahead ahead in the mix.
g_EstFrameTime = (g_EstFrameTime * 0.9f) + (g_pSoundServices->GetHostFrametime() * 0.1f);
S_Update_( g_EstFrameTime + snd_mixahead.GetFloat() );
}
CON_COMMAND( snd_dumpclientsounds, "Dump sounds to VXConsole" )
{
con_nprint_t np;
np.time_to_live = 2.0f;
np.fixed_width_font = true;
int total = 0;
CChannelList list;
g_ActiveChannels.GetActiveChannels( list );
for ( int i = 0; i < list.Count(); i++ )
{
channel_t *ch = list.GetChannel(i);
if ( !ch->sfx )
continue;
unsigned int sampleCount = RemainingSamples( ch );
float timeleft = (float)sampleCount / (float)ch->sfx->pSource->SampleRate();
bool bLooping = ch->sfx->pSource->IsLooped();
const char *pszclassname = GetClientClassname(ch->soundsource);
Msg( "%02i %s l(%03d) c(%03d) r(%03d) rl(%03d) rr(%03d) vol(%03d) pos(%6d %6d %6d) timeleft(%f) looped(%d) %50s chan:%d ent(%03d):%s\n",
total+ 1,
ch->flags.fromserver ? "SERVER" : "CLIENT",
(int)ch->fvolume[IFRONT_LEFT],
(int)ch->fvolume[IFRONT_CENTER],
(int)ch->fvolume[IFRONT_RIGHT],
(int)ch->fvolume[IREAR_LEFT],
(int)ch->fvolume[IREAR_RIGHT],
ch->master_vol,
(int)ch->origin[0],
(int)ch->origin[1],
(int)ch->origin[2],
timeleft,
bLooping,
ch->sfx->getname(),
ch->entchannel,
ch->soundsource,
pszclassname ? pszclassname : "NULL" );
total++;
}
}
//-----------------------------------------------------------------------------
// Set g_soundtime to number of full samples that have been transfered out to hardware
// since start.
//-----------------------------------------------------------------------------
void GetSoundTime(void)
{
int fullsamples;
int sampleOutCount;
// size of output buffer in *full* 16 bit samples
// A 2 channel device has a *full* sample consisting of a 16 bit LR pair.
// A 1 channel device has a *full* sample consiting of a 16 bit single sample.
fullsamples = g_AudioDevice->DeviceSampleCount() / g_AudioDevice->DeviceChannels();
// NOTE: it is possible to miscount buffers if it has wrapped twice between
// calls to S_Update. However, since the output buffer size is > 1 second of sound,
// this should only occur for framerates lower than 1hz
// sampleOutCount is counted in 16 bit *full* samples, of number of samples output to hardware
// for current output buffer
sampleOutCount = g_AudioDevice->GetOutputPosition();
if ( sampleOutCount < s_oldsampleOutCount )
{
// buffer wrapped
s_buffers++;
if ( g_paintedtime > 0x70000000 )
{
// time to chop things off to avoid 32 bit limits
s_buffers = 0;
g_paintedtime = fullsamples;
S_StopAllSounds( true );
}
}
s_oldsampleOutCount = sampleOutCount;
if ( cl_movieinfo.IsRecording() || IsReplayRendering() )
{
// when recording a replay, we look at the record frame rate, not the engine frame rate
#if defined( REPLAY_ENABLED )
extern IClientReplayContext *g_pClientReplayContext;
if ( IsReplayRendering() )
{
IReplayMovieRenderer *pMovieRenderer = (g_pClientReplayContext != NULL) ? g_pClientReplayContext->GetMovieRenderer() : NULL;
if ( pMovieRenderer && pMovieRenderer->IsAudioSyncFrame() )
{
float t = g_pSoundServices->GetHostTime();
if ( s_lastsoundtime != t )
{
float frameTime = pMovieRenderer->GetRecordingFrameDuration();
float fSamples = frameTime * (float) g_AudioDevice->DeviceDmaSpeed() + g_ReplaySoundTimeFracAccumulator;
float intPart = (float) floor( fSamples );
g_ReplaySoundTimeFracAccumulator = fSamples - intPart;
g_soundtime += (int) intPart;
s_lastsoundtime = t;
}
}
}
else // cl_movieinfo.IsRecording()
// in movie, just mix one frame worth of sound
#endif
{
float t = g_pSoundServices->GetHostTime();
if ( s_lastsoundtime != t )
{
g_soundtime += g_pSoundServices->GetHostFrametime() * g_AudioDevice->DeviceDmaSpeed();
s_lastsoundtime = t;
}
}
}
else
{
// g_soundtime indicates how many *full* samples have actually been
// played out to dma
g_soundtime = s_buffers*fullsamples + sampleOutCount;
}
}
void S_ExtraUpdate( void )
{
if ( !g_AudioDevice || !g_pSoundServices )
return;
if ( !g_AudioDevice->IsActive() )
return;
if ( s_bOnLoadScreen )
return;
if ( snd_noextraupdate.GetInt() || cl_movieinfo.IsRecording() || IsReplayRendering() )
return; // don't pollute timings
// If listener position and orientation has not yet been updated (ie: no call to S_Update since level load)
// then don't mix. Important - mixing with listener at 'false' origin causes
// some sounds to incorrectly spatialize to 0 volume, killing them before they can play.
if ((listener_origin == vec3_origin) &&
(listener_forward == vec3_origin) &&
(listener_right == vec3_origin) &&
(listener_up == vec3_origin) )
return;
// Only mix if you have used up 90% of the mixahead buffer
double tNow = Plat_FloatTime();
float delta = (tNow - g_LastMixTime);
// we know we were at least snd_mixahead seconds ahead of the output the last time we did mixing
// if we're not close to running out just exit to avoid small mix batches
if ( delta > 0 && delta < (snd_mixahead.GetFloat() * 0.9f) )
return;
g_LastMixTime = tNow;
g_pSoundServices->OnExtraUpdate();
// Shouldn't have to do any work here if your framerate hasn't dropped
S_Update_( snd_mixahead.GetFloat() );
}
extern void DEBUG_StartSoundMeasure(int type, int samplecount );
extern void DEBUG_StopSoundMeasure(int type, int samplecount );
void S_Update_Guts( float mixAheadTime )
{
VPROF( "S_Update_Guts" );
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s", __FUNCTION__ );
DEBUG_StartSoundMeasure(4, 0);
// Update our perception of audio time.
// 'g_soundtime' tells how many samples have
// been played out of the dma buffer since sound system startup.
// 'g_paintedtime' indicates how many samples we've actually mixed
// and sent to the dma buffer since sound system startup.
GetSoundTime();
// if ( g_soundtime > g_paintedtime )
// {
// // if soundtime > paintedtime, then the dma buffer
// // has played out more sound than we've actually
// // mixed. We need to call S_Update_ more often.
//
// DevMsg ("S_Update_ : Underflow\n");
// paintedtime = g_soundtime;
// }
// (kdb) above code doesn't handle underflow correctly
// should actually zero out the paintbuffer to advance to the new
// time.
// mix ahead of current position
unsigned endtime = g_AudioDevice->PaintBegin( mixAheadTime, g_soundtime, g_paintedtime );
int samples = endtime - g_paintedtime;
samples = samples < 0 ? 0 : samples;
if ( samples )
{
THREAD_LOCK_SOUND();
DEBUG_StartSoundMeasure( 2, samples );
MIX_PaintChannels( endtime, s_bIsListenerUnderwater );
MXR_DebugShowMixVolumes();
MXR_UpdateAllDuckerVolumes();
DEBUG_StopSoundMeasure( 2, 0 );
}
g_AudioDevice->PaintEnd();
DEBUG_StopSoundMeasure( 4, samples );
}
#if !defined( _X360 )
#define THREADED_MIX_TIME 33
#else
#define THREADED_MIX_TIME XMA_POLL_RATE
#endif
ConVar snd_ShowThreadFrameTime( "snd_ShowThreadFrameTime", "0" );
bool g_bMixThreadExit;
ThreadHandle_t g_hMixThread;
void S_Update_Thread()
{
float frameTime = THREADED_MIX_TIME * 0.001f;
double lastFrameTime = Plat_FloatTime();
while ( !g_bMixThreadExit )
{
// mixing (for 360) needs to be updated at a steady rate
// large update times causes the mixer to demand more audio data
// the 360 decoder has finite latency and cannot fulfill spike requests
float t0 = Plat_FloatTime();
S_Update_Guts( frameTime + snd_mixahead.GetFloat() );
int updateTime = ( Plat_FloatTime() - t0 ) * 1000.0f;
// try to maintain a steadier rate by compensating for fluctuating mix times
int sleepTime = THREADED_MIX_TIME - updateTime;
if ( sleepTime > 0 )
{
ThreadSleep( sleepTime );
}
// mimic a frametime needed for sound update
double t1 = Plat_FloatTime();
frameTime = t1 - lastFrameTime;
lastFrameTime = t1;
if ( snd_ShowThreadFrameTime.GetBool() )
{
Msg( "S_Update_Thread: frameTime: %d ms\n", (int)( frameTime * 1000.0f ) );
}
}
}
void S_ShutdownMixThread()
{
if ( g_hMixThread )
{
g_bMixThreadExit = true;
ThreadJoin( g_hMixThread );
ReleaseThreadHandle( g_hMixThread );
g_hMixThread = NULL;
}
}
void S_Update_( float mixAheadTime )
{
if ( !IsConsole() || !snd_mix_async.GetBool() )
{
S_ShutdownMixThread();
S_Update_Guts( mixAheadTime );
}
else
{
if ( !g_hMixThread )
{
g_bMixThreadExit = false;
g_hMixThread = ThreadExecuteSolo( "SndMix", S_Update_Thread );
if ( IsX360() )
{
ThreadSetAffinity( g_hMixThread, XBOX_PROCESSOR_5 );
}
}
}
}
//-----------------------------------------------------------------------------
// Threaded mixing enable. Purposely hiding enable/disable details.
//-----------------------------------------------------------------------------
void S_EnableThreadedMixing( bool bEnable )
{
if ( snd_mix_async.GetBool() != bEnable )
{
snd_mix_async.SetValue( bEnable );
}
}
/*
===============================================================================
console functions
===============================================================================
*/
extern void DSP_DEBUGSetParams(int ipreset, int iproc, float *pvalues, int cparams);
extern void DSP_DEBUGReloadPresetFile( void );
void S_DspParms( const CCommand &args )
{
if ( args.ArgC() == 1)
{
// if dsp_parms with no arguments, reload entire preset file
DSP_DEBUGReloadPresetFile();
return;
}
if ( args.ArgC() < 4 )
{
Msg( "Usage: dsp_parms PRESET# PROC# param0 param1 ...up to param15 \n" );
return;
}
int cparam = min( args.ArgC() - 4, 16);
float params[16];
Q_memset( params, 0, sizeof(float) * 16 );
// get preset & proc
int idsp, iproc;
idsp = Q_atof( args[1] );
iproc = Q_atof( args[2] );
// get params
for (int i = 0; i < cparam; i++)
{
params[i] = Q_atof( args[i+4] );
}
// set up params & switch preset
DSP_DEBUGSetParams(idsp, iproc, params, cparam);
}
static ConCommand dsp_parm("dsp_reload", S_DspParms );
void S_Play( const char *pszName, bool flush = false )
{
int inCache;
char szName[256];
CSfxTable *pSfx;
Q_strncpy( szName, pszName, sizeof( szName ) );
if ( !Q_strrchr( pszName, '.' ) )
{
Q_strncat( szName, ".wav", sizeof( szName ), COPY_ALL_CHARACTERS );
}
pSfx = S_FindName( szName, &inCache );
if ( inCache && flush )
{
pSfx->pSource->CacheUnload();
}
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = g_pSoundServices->GetViewEntity();
params.entchannel = CHAN_REPLACE;
params.pSfx = pSfx;
params.origin = listener_origin;
params.fvol = 1.0f;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
S_StartSound( params );
}
static void S_Play( const CCommand &args )
{
bool bFlush = !Q_stricmp( args[0], "playflush" );
for ( int i = 1; i < args.ArgC(); ++i )
{
S_Play( args[i], bFlush );
}
}
static void S_PlayVol( const CCommand &args )
{
static int hash=543;
float vol;
char name[256];
CSfxTable *pSfx;
for ( int i = 1; i<args.ArgC(); i += 2 )
{
if ( !Q_strrchr( args[i], '.') )
{
Q_strncpy( name, args[i], sizeof( name ) );
Q_strncat( name, ".wav", sizeof( name ), COPY_ALL_CHARACTERS );
}
else
{
Q_strncpy( name, args[i], sizeof( name ) );
}
pSfx = S_PrecacheSound( name );
vol = Q_atof( args[i+1] );
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = hash++;
params.entchannel = CHAN_AUTO;
params.pSfx = pSfx;
params.origin = listener_origin;
params.fvol = vol;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
S_StartDynamicSound( params );
}
}
static void S_PlayDelay( const CCommand &args )
{
if ( args.ArgC() != 3 )
{
Msg( "Usage: sndplaydelay delay_in_sec (negative to skip ahead) soundname\n" );
return;
}
char szName[256];
CSfxTable *pSfx;
float delay = Q_atof( args[ 1 ] );
Q_strncpy(szName, args[ 2 ], sizeof( szName ) );
if ( !Q_strrchr( args[ 2 ], '.' ) )
{
Q_strncat( szName, ".wav", sizeof( szName ), COPY_ALL_CHARACTERS );
}
pSfx = S_FindName( szName, NULL );
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = g_pSoundServices->GetViewEntity();
params.entchannel = CHAN_REPLACE;
params.pSfx = pSfx;
params.origin = listener_origin;
params.fvol = 1.0f;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
params.delay = delay;
S_StartSound( params );
}
static ConCommand sndplaydelay( "sndplaydelay", S_PlayDelay, "Usage: sndplaydelay delay_in_sec (negative to skip ahead) soundname", FCVAR_SERVER_CAN_EXECUTE );
static bool SortByNameLessFunc( const int &lhs, const int &rhs )
{
CSfxTable *pSfx1 = s_Sounds[lhs].pSfx;
CSfxTable *pSfx2 = s_Sounds[rhs].pSfx;
return CaselessStringLessThan( pSfx1->getname(), pSfx2->getname() );
}
void S_SoundList(void)
{
CSfxTable *sfx;
CAudioSource *pSource;
int size, total;
total = 0;
for ( int i = s_Sounds.FirstInorder(); i != s_Sounds.InvalidIndex(); i = s_Sounds.NextInorder( i ) )
{
sfx = s_Sounds[i].pSfx;
pSource = sfx->pSource;
if ( !pSource || !pSource->IsCached() )
continue;
size = pSource->SampleSize() * pSource->SampleCount();
total += size;
if ( pSource->IsLooped() )
Msg ("L");
else
Msg (" ");
Msg("(%2db) %6i : %s\n", pSource->SampleSize(), size, sfx->getname());
}
Msg( "Total resident: %i\n", total );
}
#if defined( _X360 )
CON_COMMAND( vx_soundlist, "Dump sounds to VXConsole" )
{
CSfxTable *sfx;
CAudioSource *pSource;
int dataSize;
char *pFormatStr;
int sampleRate;
int sampleBits;
int streamed;
int looped;
int channels;
int numSamples;
int numSounds = s_Sounds.Count();
xSoundList_t* pSoundList = new xSoundList_t[numSounds];
int i = 0;
for ( int iSrcSound=s_Sounds.FirstInorder(); iSrcSound != s_Sounds.InvalidIndex(); iSrcSound = s_Sounds.NextInorder( iSrcSound ) )
{
dataSize = -1;
sampleRate = -1;
sampleBits = -1;
pFormatStr = "???";
streamed = -1;
looped = -1;
channels = -1;
numSamples = -1;
sfx = s_Sounds[iSrcSound].pSfx;
pSource = sfx->pSource;
if ( pSource && pSource->IsCached() )
{
numSamples = pSource->SampleCount();
dataSize = pSource->DataSize();
sampleRate = pSource->SampleRate();
streamed = pSource->IsStreaming();
looped = pSource->IsLooped();
channels = pSource->IsStereoWav() ? 2 : 1;
if ( pSource->Format() == WAVE_FORMAT_ADPCM )
{
pFormatStr = "ADPCM";
sampleBits = 16;
}
else if ( pSource->Format() == WAVE_FORMAT_PCM )
{
pFormatStr = "PCM";
sampleBits = (pSource->SampleSize() * 8)/channels;
}
else if ( pSource->Format() == WAVE_FORMAT_XMA )
{
pFormatStr = "XMA";
sampleBits = 16;
}
}
V_strncpy( pSoundList[i].name, sfx->getname(), sizeof( pSoundList[i].name ) );
V_strncpy( pSoundList[i].formatName, pFormatStr, sizeof( pSoundList[i].formatName ) );
pSoundList[i].rate = sampleRate;
pSoundList[i].bits = sampleBits;
pSoundList[i].channels = channels;
pSoundList[i].looped = looped;
pSoundList[i].dataSize = dataSize;
pSoundList[i].numSamples = numSamples;
pSoundList[i].streamed = streamed;
++i;
}
XBX_rSoundList( numSounds, pSoundList );
delete [] pSoundList;
}
#endif
extern unsigned g_snd_time_debug;
extern unsigned g_snd_call_time_debug;
extern unsigned g_snd_count_debug;
extern unsigned g_snd_samplecount;
extern unsigned g_snd_frametime;
extern unsigned g_snd_frametime_total;
extern int g_snd_profile_type;
// start measuring sound perf, 100 reps
// type 1 - dsp, 2 - mix, 3 - load sound, 4 - all sound
// set type via ConVar snd_profile
void DEBUG_StartSoundMeasure(int type, int samplecount )
{
if (type != g_snd_profile_type)
return;
if (samplecount)
g_snd_samplecount += samplecount;
g_snd_call_time_debug = Plat_MSTime();
}
// show sound measurement after 25 reps - show as % of total frame
// type 1 - dsp, 2 - mix, 3 - load sound, 4 - all sound
// BUGBUG: snd_profile 4 reports a lower average because it's average cost
// PER CALL and most calls (via SoundExtraUpdate()) don't do any work and
// bring the average down. If you want an average PER FRAME instead, it's generally higher.
void DEBUG_StopSoundMeasure(int type, int samplecount )
{
if (type != g_snd_profile_type)
return;
if (samplecount)
g_snd_samplecount += samplecount;
// add total time since last frame
g_snd_frametime_total += Plat_MSTime() - g_snd_frametime;
// performance timing
g_snd_time_debug += Plat_MSTime() - g_snd_call_time_debug;
if (++g_snd_count_debug >= 100)
{
switch (g_snd_profile_type)
{
case 1:
Msg("dsp: (%2.2f) millisec ", ((float)g_snd_time_debug) / 100.0);
Msg("(%2.2f) pct of frame \n", 100.0 * ((float)g_snd_time_debug) / ((float)g_snd_frametime_total));
break;
case 2:
Msg("mix+dsp:(%2.2f) millisec ", ((float)g_snd_time_debug) / 100.0);
Msg("(%2.2f) pct of frame \n", 100.0 * ((float)g_snd_time_debug) / ((float)g_snd_frametime_total));
break;
case 3:
//if ( (((float)g_snd_time_debug) / 100.0) < 0.01 )
// break;
Msg("snd load: (%2.2f) millisec ", ((float)g_snd_time_debug) / 100.0);
Msg("(%2.2f) pct of frame \n", 100.0 * ((float)g_snd_time_debug) / ((float)g_snd_frametime_total));
break;
case 4:
Msg("sound: (%2.2f) millisec ", ((float)g_snd_time_debug) / 100.0);
Msg("(%2.2f) pct of frame (%d samples) \n", 100.0 * ((float)g_snd_time_debug) / ((float)g_snd_frametime_total), g_snd_samplecount);
break;
}
g_snd_count_debug = 0;
g_snd_time_debug = 0;
g_snd_samplecount = 0;
g_snd_frametime_total = 0;
}
g_snd_frametime = Plat_MSTime();
}
// speak a sentence from console; works by passing in "!sentencename"
// or "sentence"
extern ConVar dsp_room;
static void S_Say( const CCommand &args )
{
CSfxTable *pSfx;
if ( !g_AudioDevice->IsActive() )
return;
char sound[256];
Q_strncpy( sound, args[1], sizeof( sound ) );
// DEBUG - test performance of dsp code
if ( !Q_stricmp( sound, "dsp" ) )
{
unsigned time;
int i;
int count = 10000;
int idsp;
for (i = 0; i < PAINTBUFFER_SIZE; i++)
{
g_paintbuffer[i].left = RandomInt(0,2999);
g_paintbuffer[i].right = RandomInt(0,2999);
}
Msg ("Start profiling 10,000 calls to DSP\n");
idsp = dsp_room.GetInt();
// get system time
time = Plat_MSTime();
for (i = 0; i < count; i++)
{
// SX_RoomFX(PAINTBUFFER_SIZE, TRUE, TRUE);
DSP_Process(idsp, g_paintbuffer, NULL, NULL, PAINTBUFFER_SIZE);
}
// display system time delta
Msg("%d milliseconds \n", Plat_MSTime() - time);
return;
}
if ( !Q_stricmp(sound, "paint") )
{
unsigned time;
int count = 10000;
static int hash=543;
int psav = g_paintedtime;
Msg ("Start profiling MIX_PaintChannels\n");
pSfx = S_PrecacheSound("ambience/labdrone1.wav");
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = hash++;
params.entchannel = CHAN_AUTO;
params.pSfx = pSfx;
params.origin = listener_origin;
params.fvol = 1.0f;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
S_StartDynamicSound( params );
// get system time
time = Plat_MSTime();
// paint a boatload of sound
MIX_PaintChannels( g_paintedtime + 512*count, s_bIsListenerUnderwater );
// display system time delta
Msg("%d milliseconds \n", Plat_MSTime() - time);
g_paintedtime = psav;
return;
}
// DEBUG
if ( !TestSoundChar( sound, CHAR_SENTENCE ) )
{
// build a fake sentence name, then play the sentence text
Q_strncpy(sound, "xxtestxx ", sizeof( sound ) );
Q_strncat(sound, args[1], sizeof( sound ), COPY_ALL_CHARACTERS );
int addIndex = g_Sentences.AddToTail();
sentence_t *pSentence = &g_Sentences[addIndex];
pSentence->pName = sound;
pSentence->length = 0;
// insert null terminator after sentence name
sound[8] = 0;
pSfx = S_PrecacheSound ("!xxtestxx");
if (!pSfx)
{
Msg ("S_Say: can't cache %s\n", sound);
return;
}
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = g_pSoundServices->GetViewEntity();
params.entchannel = CHAN_REPLACE;
params.pSfx = pSfx;
params.origin = vec3_origin;
params.fvol = 1.0f;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
S_StartDynamicSound ( params );
// remove last
g_Sentences.Remove( g_Sentences.Size() - 1 );
}
else
{
pSfx = S_FindName(sound, NULL);
if (!pSfx)
{
Msg ("S_Say: can't find sentence name %s\n", sound);
return;
}
StartSoundParams_t params;
params.staticsound = false;
params.soundsource = g_pSoundServices->GetViewEntity();
params.entchannel = CHAN_REPLACE;
params.pSfx = pSfx;
params.origin = vec3_origin;
params.fvol = 1.0f;
params.soundlevel = SNDLVL_NONE;
params.flags = 0;
params.pitch = PITCH_NORM;
S_StartDynamicSound( params );
}
}
//------------------------------------------------------------------------------
//
// Sound Mixers
//
// Sound mixers are referenced by name from Soundscapes, and are used to provide
// custom volume control over various sound categories, called 'mix groups'
//
// see scripts/soundmixers.txt for data format
//------------------------------------------------------------------------------
#define CMXRGROUPMAX 64 // up to n mixgroups
#define CMXRGROUPRULESMAX (CMXRGROUPMAX + 16) // max number of group rules
#define CMXRSOUNDMIXERSMAX 32 // up to n sound mixers per project
// mix groups - these equivalent to submixes on an audio mixer
// list of rules for determining sound membership in mix groups.
// All conditions which are not null are ANDed together
#define CMXRCLASSMAX 16
#define CMXRNAMEMAX 32
struct classlistelem_t
{
char szclassname[CMXRNAMEMAX]; // name of entities' class, such as CAI_BaseNPC or CHL2_Player
};
struct grouprule_t
{
char szmixgroup[CMXRNAMEMAX]; // mix group name
int mixgroupid; // mix group unique id
char szdir[CMXRNAMEMAX]; // substring to search for in ch->sfx
int classId; // index of classname
int chantype; // channel type (CHAN_WEAPON, etc)
int soundlevel_min; // min soundlevel
int soundlevel_max; // max soundlevel
int priority; // 0..100 higher priority sound groups duck all lower pri groups if enabled
int is_ducked; // if 1, sound group is ducked by all higher priority 'causes_duck" sounds
int causes_ducking; // if 1, sound group ducks other 'is_ducked' sounds of lower priority
float duck_target_pct; // if sound group is ducked, target percent of original volume
float total_vol; // total volume of all sounds in this group, if group can cause ducking
float ducker_threshold; // ducking is caused by this group if total_vol > ducker_threshold
// and causes_ducking is enabled.
float duck_target_vol; // target volume while ducking
float duck_ramp_val; // current value of ramp - moves towards duck_target_vol
};
// sound mixer
struct soundmixer_t
{
char szsoundmixer[CMXRNAMEMAX]; // name of this soundmixer
float mapMixgroupidToValue[CMXRGROUPMAX]; // sparse array of mix group values for this soundmixer
};
int g_mapMixgroupidToGrouprulesid[CMXRGROUPMAX]; // map mixgroupid (one per unique group name)
// back to 1st entry of this name in g_grouprules
// sound mixer globals
classlistelem_t g_groupclasslist[CMXRCLASSMAX];
soundmixer_t g_soundmixers[CMXRSOUNDMIXERSMAX]; // all sound mixers
grouprule_t g_grouprules[CMXRGROUPRULESMAX]; // all rules for determining mix group membership
// set current soundmixer index g_isoundmixer, search for match in soundmixers
// Only change current soundmixer if new name is different from current name.
int g_isoundmixer = -1; // index of current sound mixer
char g_szsoundmixer_cur[64]; // current soundmixer name
ConVar snd_soundmixer("snd_soundmixer", "Default_Mix"); // current soundmixer name
void MXR_SetCurrentSoundMixer( const char *szsoundmixer )
{
// if soundmixer name is not different from current name, return
if ( !Q_stricmp(szsoundmixer, g_szsoundmixer_cur) )
{
return;
}
for (int i = 0; i < g_csoundmixers; i++)
{
if ( !Q_stricmp(g_soundmixers[i].szsoundmixer, szsoundmixer) )
{
g_isoundmixer = i;
// save new current sound mixer name
V_strcpy_safe(g_szsoundmixer_cur, szsoundmixer);
return;
}
}
}
ConVar snd_showclassname("snd_showclassname", "0"); // if 1, show classname of ent making sound
// if 2, show all mixgroup matches
// if 3, show all mixgroup matches with current soundmixer for ent
// get the client class name if an entity was specified
const char *GetClientClassname( SoundSource soundsource )
{
IClientEntity *pClientEntity = NULL;
if ( entitylist )
{
pClientEntity = entitylist->GetClientEntity( soundsource );
if ( pClientEntity )
{
ClientClass *pClientClass = pClientEntity->GetClientClass();
// check npc sounds
if ( pClientClass )
{
return pClientClass->GetName();
}
}
}
return NULL;
}
// builds a cached list of rules that match the directory name on the sound
int MXR_GetMixGroupListFromDirName( const char *pDirname, byte *pList, int listMax )
{
// if we call this before the groups are parsed we'll get bad data
Assert(g_cgrouprules>0);
int count = 0;
for ( int i = 0; i < listMax; i++ )
{
pList[i] = 255;
}
for ( int i = 0; i < g_cgrouprules; i++ )
{
grouprule_t *prule = &g_grouprules[i];
if ( prule->szdir[ 0 ] && Q_stristr( pDirname, prule->szdir ) )
{
pList[count] = i;
count++;
if ( count >= listMax )
return count;
}
}
return count;
}
// determine which mixgroups sound is in, and save those mixgroupids in sound.
// use current soundmixer indicated with g_isoundmixer, and contents of g_rgpgrouprules.
// Algorithm:
// 1. all conditions in a row are AND conditions,
// 2. all rows sharing the same groupname are OR conditions.
// so - if a sound matches all conditions of a row, it is given that row's mixgroup id
// if a sound doesn't match all conditions of a row, the next row is checked.
// returns 0, default mixgroup if no match
void MXR_GetMixGroupFromSoundsource( channel_t *pchan, SoundSource soundsource, soundlevel_t soundlevel)
{
int i;
grouprule_t *prule;
bool fmatch;
bool classMatch[CMXRCLASSMAX];
// init all mixgroups for channel
for ( i = 0; i < 8; i++ )
{
pchan->mixgroups[i] = -1;
}
char sndname[MAX_OSPATH];
Q_strncpy( sndname, pchan->sfx->getname(), sizeof( sndname ) );
// Use forward slashes here
Q_FixSlashes( sndname, '/' );
const char *pszclassname = GetClientClassname(soundsource);
for ( i = 0; i < g_cgroupclass; i++ )
{
classMatch[i] = false;
if ( pszclassname && Q_stristr(pszclassname, g_groupclasslist[i].szclassname ) )
{
classMatch[i] = true;
}
}
if ( snd_showclassname.GetInt() == 1)
{
// utility: show classname of ent making sound
if (pszclassname)
{
DevMsg("(%s:%s) \n", pszclassname, sndname);
}
}
// check all group rules for a match, save
// up to 8 matches in channel mixgroup.
int cmixgroups = 0;
if (!pchan->sfx->m_bMixGroupsCached)
{
pchan->sfx->OnNameChanged( pchan->sfx->getname() );
}
// since this is a sorted list (in group rule order) we only need to test against the next matching rule
// this avoids a search inside the loop
int currentDirRuleIndex = 0;
int currentDirRule = pchan->sfx->m_mixGroupList[0];
for (i = 0; i < g_cgrouprules; i++)
{
prule = &g_grouprules[i];
fmatch = true;
// check directory or name substring
#if _DEBUG
// check dir table is correct in CSfxTable cache
if ( prule->szdir[ 0 ] && Q_stristr( sndname, prule->szdir ) )
{
Assert(currentDirRule == i);
}
else
{
Assert(currentDirRule != i);
}
if ( prule->classId >= 0 )
{
// rule has a valid class id and table is correct
Assert(prule->classId < g_cgroupclass);
if ( pszclassname && Q_stristr(pszclassname, g_groupclasslist[prule->classId].szclassname) )
{
Assert(classMatch[prule->classId] == true);
}
else
{
Assert(classMatch[prule->classId] == false);
}
}
#endif
// this is the next matching dir for this sound, no need to search
// becuse the list is sorted and we visit all elements
if ( currentDirRule == i )
{
Assert(prule->szdir[0]);
currentDirRuleIndex++;
currentDirRule = 255;
if ( currentDirRuleIndex < pchan->sfx->m_mixGroupCount )
{
currentDirRule = pchan->sfx->m_mixGroupList[currentDirRuleIndex];
}
}
else if ( prule->szdir[ 0 ] )
{
fmatch = false; // substring doesn't match, keep looking
}
// check class name
if ( fmatch && prule->classId >= 0 )
{
fmatch = classMatch[prule->classId];
}
// check channel type
if ( fmatch && prule->chantype >= 0)
{
if ( pchan->entchannel != prule->chantype )
fmatch = false; // channel type doesn't match, keep looking
}
// check sndlvlmin/max
if ( fmatch && prule->soundlevel_min >= 0)
{
if ( soundlevel < prule->soundlevel_min )
fmatch = false; // soundlevel is less than min, keep looking
}
if ( fmatch && prule->soundlevel_max >= 0)
{
if ( soundlevel > prule->soundlevel_max )
fmatch = false; // soundlevel is greater than max, keep looking
}
if ( fmatch )
{
pchan->mixgroups[cmixgroups] = prule->mixgroupid;
cmixgroups++;
if (cmixgroups >= 8)
return; // too many matches, stop looking
}
if (fmatch && snd_showclassname.GetInt() >= 2)
{
// show all mixgroups for this sound
if (cmixgroups == 1)
{
DevMsg("\n%s:%s: ", g_szsoundmixer_cur, sndname);
}
if (prule->szmixgroup[0])
{
// int rgmixgroupid[8];
// for (int i = 0; i < 8; i++)
// rgmixgroupid[i] = -1;
// rgmixgroupid[0] = prule->mixgroupid;
// float vol = MXR_GetVolFromMixGroup( rgmixgroupid );
// DevMsg("%s(%1.2f) ", prule->szmixgroup, vol);
DevMsg("%s ", prule->szmixgroup);
}
}
}
}
struct debug_showvols_t
{
char *psz; // group name
int mixgroupid; // groupid
float vol; // group volume
float totalvol; // total volume of all sounds playing in this group
};
// display routine for MXR_DebugShowMixVolumes
#define MXR_DEBUG_INCY (1.0/40.0) // vertical text spacing
#define MXR_DEBUG_GREENSTART 0.3 // start position on screen of bar
#define MXR_DEBUG_MAXVOL 1.0 // max volume scale
#define MXR_DEBUG_REDLIMIT 1.0 // volume limit into yellow
#define MXR_DEBUG_YELLOWLIMIT 0.7 // volume limit into red
#define MXR_DEBUG_VOLSCALE 48 // length of graph in characters
#define MXR_DEBUG_CHAR '-' // bar character
extern ConVar dsp_volume;
int g_debug_mxr_displaycount = 0;
void MXR_DebugGraphMixVolumes( debug_showvols_t *groupvols, int cgroups)
{
float flXpos, flYpos, flXposBar, duration;
int r,g,b,a;
int rb, gb, bb, ab;
flXpos = 0;
flYpos = 0;
char text[128];
char bartext[MXR_DEBUG_VOLSCALE*3];
duration = 0.01;
g_debug_mxr_displaycount++;
if (!(g_debug_mxr_displaycount % 10))
return; // only display every 10 frames
r = 96; g = 86; b = 226; a = 255; ab = 255;
// show volume, dsp_volume
Q_snprintf( text, 128, "Game Volume: %1.2f", volume.GetFloat());
CDebugOverlay::AddScreenTextOverlay(flXpos, flYpos, duration, r, g, b,a, text);
flYpos += MXR_DEBUG_INCY;
Q_snprintf( text, 128, "DSP Volume: %1.2f", dsp_volume.GetFloat());
CDebugOverlay::AddScreenTextOverlay(flXpos, flYpos, duration, r, g, b,a, text);
flYpos += MXR_DEBUG_INCY;
for (int i = 0; i < cgroups; i++)
{
// r += 64; g += 64; b += 16;
r = r % 255; g = g % 255; b = b % 255;
Q_snprintf( text, 128, "%s: %1.2f (%1.2f)", groupvols[i].psz,
groupvols[i].vol * g_DuckScale, groupvols[i].totalvol * g_DuckScale);
CDebugOverlay::AddScreenTextOverlay(flXpos, flYpos, duration, r, g, b,a, text);
// draw volume bar graph
float vol = (groupvols[i].totalvol * g_DuckScale) / MXR_DEBUG_MAXVOL;
// draw first 70% green
float vol1 = 0.0;
float vol2 = 0.0;
float vol3 = 0.0;
int cbars;
vol1 = clamp(vol, 0.0f, 0.7f);
vol2 = clamp(vol, 0.0f, 0.95f);
vol3 = vol;
flXposBar = flXpos + MXR_DEBUG_GREENSTART;
if (vol1 > 0.0)
{
//flXposBar = flXpos + MXR_DEBUG_GREENSTART;
rb = 0; gb= 255; bb = 0; // green bar
Q_memset(bartext, 0, sizeof(bartext));
cbars = (int)((float)vol1 * (float)MXR_DEBUG_VOLSCALE);
cbars = clamp(cbars, 0, MXR_DEBUG_VOLSCALE*3-1);
Q_memset(bartext, MXR_DEBUG_CHAR, cbars);
CDebugOverlay::AddScreenTextOverlay(flXposBar, flYpos, duration, rb, gb, bb,ab, bartext);
}
// yellow bar
if (vol2 > MXR_DEBUG_YELLOWLIMIT)
{
rb = 255; gb = 255; bb = 0;
Q_memset(bartext, 0, sizeof(bartext));
cbars = (int)((float)vol2 * (float)MXR_DEBUG_VOLSCALE);
cbars = clamp(cbars, 0, MXR_DEBUG_VOLSCALE*3-1);
Q_memset(bartext, MXR_DEBUG_CHAR, cbars);
CDebugOverlay::AddScreenTextOverlay(flXposBar, flYpos, duration, rb, gb, bb,ab, bartext);
}
// red bar
if (vol3 > MXR_DEBUG_REDLIMIT)
{
//flXposBar = flXpos + MXR_DEBUG_REDSTART;
rb = 255; gb = 0; bb = 0;
Q_memset(bartext, 0, sizeof(bartext));
cbars = (int)((float)vol3 * (float)MXR_DEBUG_VOLSCALE);
cbars = clamp(cbars, 0, MXR_DEBUG_VOLSCALE*3-1);
Q_memset(bartext, MXR_DEBUG_CHAR, cbars);
CDebugOverlay::AddScreenTextOverlay(flXposBar, flYpos, duration, rb, gb, bb,ab, bartext);
}
flYpos += MXR_DEBUG_INCY;
}
}
ConVar snd_disable_mixer_duck("snd_disable_mixer_duck", "0"); // if 1, soundmixer ducking is disabled
// given mix group id, return current duck volume
float MXR_GetDuckVolume( int mixgroupid )
{
if ( snd_disable_mixer_duck.GetInt() )
return 1.0;
Assert ( mixgroupid < g_cgrouprules );
int grouprulesid = g_mapMixgroupidToGrouprulesid[mixgroupid];
// if this mixgroup is not ducked, return 1.0
if ( !g_grouprules[grouprulesid].is_ducked )
return 1.0;
// return current duck value for this group, scaled by current fade in/out ramp
return g_grouprules[grouprulesid].duck_ramp_val;
}
#define SND_DUCKER_UPDATETIME 0.1 // seconds to wait between ducker updates
double g_mxr_ducktime = 0.0; // time of last update to ducker
// Get total volume currently playing in all groups,
// process duck volumes for all groups
// Call once per frame - updates occur at 10hz
void MXR_UpdateAllDuckerVolumes( void )
{
if ( snd_disable_mixer_duck.GetInt() )
return;
// check timer since last update, only update at 10hz
int i;
double dtime = g_pSoundServices->GetHostTime();
// don't update until timer expires
if (fabs(dtime - g_mxr_ducktime) < SND_DUCKER_UPDATETIME)
return;
g_mxr_ducktime = dtime;
// clear out all total volume values for groups
for ( i = 0; i < g_cgrouprules; i++)
g_grouprules[i].total_vol = 0.0;
// for every channel in a mix group which can cause ducking:
// get total volume, store total in grouprule:
CChannelList list;
int ch_idx;
channel_t *pchan;
bool b_found_ducked_channel = false;
g_ActiveChannels.GetActiveChannels( list );
for ( i = 0; i < list.Count(); i++ )
{
ch_idx = list.GetChannelIndex(i);
pchan = &channels[ch_idx];
if (pchan->last_vol > 0.0)
{
// account for all mix groups this channel belongs to...
for (int j = 0; j < 8; j++)
{
int imixgroup = pchan->mixgroups[j];
if (imixgroup < 0)
continue;
int grouprulesid = g_mapMixgroupidToGrouprulesid[imixgroup];
if (g_grouprules[grouprulesid].causes_ducking)
g_grouprules[grouprulesid].total_vol += pchan->last_vol;
if (g_grouprules[grouprulesid].is_ducked)
b_found_ducked_channel = true;
}
}
}
// if no channels playing which may be ducked, do nothing
if ( !b_found_ducked_channel )
return;
// for all groups that can be ducked:
// see if a higher priority sound group has a volume > threshold,
// if so, then duck this group by setting duck_target_vol to duck_target_pct.
// if no sound group is causing ducking in this group, reset duck_target_vol to 1.0
for (i = 0; i < g_cgrouprules; i++)
{
if (g_grouprules[i].is_ducked)
{
int priority = g_grouprules[i].priority;
float duck_volume = 1.0; // clear to 1.0 if no channel causing ducking
// make sure we interact appropriately with global voice ducking...
// if global voice ducking is active, skip sound group ducking and just set duck_volume target to 1.0
if ( g_DuckScale >= 1.0 )
{
// check all sound groups for higher priority duck trigger
for (int j = 0; j < g_cgrouprules; j++)
{
if (g_grouprules[j].priority > priority &&
g_grouprules[j].causes_ducking &&
g_grouprules[j].total_vol > g_grouprules[j].ducker_threshold)
{
// a higher priority group is causing this group to be ducked
// set duck volume target to the ducked group's duck target percent
// and break
duck_volume = g_grouprules[i].duck_target_pct;
// UNDONE: to prevent edge condition caused by crossing threshold, may need to have secondary
// UNDONE: timer which allows ducking at 0.2 hz
break;
}
}
}
g_grouprules[i].duck_target_vol = duck_volume;
}
}
// update all ducker ramps if current duck value is not target
// if ramp is greater than duck_volume, approach at 'attack rate'
// if ramp is less than duck_volume, approach at 'decay rate'
for (i = 0; i < g_cgrouprules; i++)
{
float target = g_grouprules[i].duck_target_vol;
float current = g_grouprules[i].duck_ramp_val;
if (g_grouprules[i].is_ducked && (current != target))
{
float ramptime = target < current ? snd_duckerattacktime.GetFloat() : snd_duckerreleasetime.GetFloat();
// delta is volume change per update (we can do this
// since we run at an approximate fixed update rate of 10hz)
float delta = (1.0 - g_grouprules[i].duck_target_pct);
delta *= ( SND_DUCKER_UPDATETIME / ramptime );
if (current > target)
delta = -delta;
// update ramps
current += delta;
if (current < target && delta < 0)
current = target;
if (current > target && delta > 0)
current = target;
g_grouprules[i].duck_ramp_val = current;
}
}
}
ConVar snd_showmixer("snd_showmixer", "0"); // set to 1 to show mixer every frame
// show the current soundmixer output
void MXR_DebugShowMixVolumes( void )
{
if (snd_showmixer.GetInt() == 0)
return;
// for the current soundmixer:
// make a totalvolume bucket for each mixgroup type in the soundmixer.
// for every active channel, add its spatialized volume to
// totalvolume bucket for that channel's selected mixgroup
// display all mixgroup/volume/totalvolume values as horizontal bars
debug_showvols_t groupvols[CMXRGROUPMAX];
int i;
int cgroups = 0;
if (g_isoundmixer < 0)
{
DevMsg("No sound mixer selected!");
return;
}
soundmixer_t *pmixer = &g_soundmixers[g_isoundmixer];
// for every entry in mapMixgroupidToValue which is not -1,
// set up groupvols
for (i = 0; i < CMXRGROUPMAX; i++)
{
if (pmixer->mapMixgroupidToValue[i] >= 0)
{
groupvols[cgroups].mixgroupid = i;
groupvols[cgroups].psz = MXR_GetGroupnameFromId( i );
groupvols[cgroups].totalvol = 0.0;
groupvols[cgroups].vol = pmixer->mapMixgroupidToValue[i];
cgroups++;
}
}
// for every active channel, get its volume and
// the selected mixgroupid, add to groupvols totalvol
CChannelList list;
int ch_idx;
channel_t *pchan;
g_ActiveChannels.GetActiveChannels( list );
for ( i = 0; i < list.Count(); i++ )
{
ch_idx = list.GetChannelIndex(i);
pchan = &channels[ch_idx];
if (pchan->last_vol > 0.0)
{
// find entry in groupvols
for (int j = 0; j < CMXRGROUPMAX; j++)
{
if (pchan->last_mixgroupid == groupvols[j].mixgroupid)
{
groupvols[j].totalvol += pchan->last_vol;
break;
}
}
}
}
// groupvols is now fully initialized - just display it
MXR_DebugGraphMixVolumes( groupvols, cgroups);
}
#ifdef _DEBUG
// set the named mixgroup volume to vol for the current soundmixer
static void MXR_DebugSetMixGroupVolume( const CCommand &args )
{
if ( args.ArgC() != 3 )
{
DevMsg("Parameters: mix group name, volume");
return;
}
const char *szgroupname = args[1];
float vol = atof( args[2] );
int imixgroup = MXR_GetMixgroupFromName( szgroupname );
if ( g_isoundmixer < 0 )
return;
soundmixer_t *pmixer = &g_soundmixers[g_isoundmixer];
pmixer->mapMixgroupidToValue[imixgroup] = vol;
}
#endif //_DEBUG
// given array of groupids (ie: the sound is in these groups),
// return a mix volume.
// return first mixgroup id in the provided array
// which maps to a non -1 volume value for this
// sound mixer
float MXR_GetVolFromMixGroup( int rgmixgroupid[8], int *plast_mixgroupid )
{
// if no soundmixer currently set, return 1.0 volume
if (g_isoundmixer < 0)
{
*plast_mixgroupid = 0;
return 1.0;
}
float duckgain = 1.0;
if (g_csoundmixers)
{
soundmixer_t *pmixer = &g_soundmixers[g_isoundmixer];
if (pmixer)
{
// search mixgroupid array, return first match (non -1)
for (int i = 0; i < 8; i++)
{
int imixgroup = rgmixgroupid[i];
if (imixgroup < 0)
continue;
// save lowest duck gain value for any of the mix groups this sound is in
float duckgain_new = MXR_GetDuckVolume( imixgroup );
if ( duckgain_new < duckgain)
duckgain = duckgain_new;
Assert(imixgroup < CMXRGROUPMAX);
// return first mixgroup id in the passed in array
// that maps to a non -1 volume value for this
// sound mixer
if ( pmixer->mapMixgroupidToValue[imixgroup] >= 0)
{
*plast_mixgroupid = imixgroup;
// get gain due to mixer settings
float gain = pmixer->mapMixgroupidToValue[imixgroup];
// modify gain with ducker settings for this group
return gain * duckgain;
}
}
}
}
*plast_mixgroupid = 0;
return duckgain;
}
// get id of mixgroup name
int MXR_GetMixgroupFromName( const char *pszgroupname )
{
// scan group rules for mapping from name to id
if ( !pszgroupname )
return -1;
if ( Q_strlen(pszgroupname) == 0 )
return -1;
for (int i = 0; i < g_cgrouprules; i++)
{
if ( !Q_stricmp(g_grouprules[i].szmixgroup, pszgroupname ) )
return g_grouprules[i].mixgroupid;
}
return -1;
}
// get mixgroup name from id
char *MXR_GetGroupnameFromId( int mixgroupid)
{
// scan group rules for mapping from name to id
if (mixgroupid < 0)
return NULL;
for (int i = 0; i < g_cgrouprules; i++)
{
if ( g_grouprules[i].mixgroupid == mixgroupid)
return g_grouprules[i].szmixgroup;
}
return NULL;
}
// assign a unique mixgroup id to each unique named mix group
// within grouprules. Note: all mixgroupids in grouprules must be -1
// when this routine starts.
void MXR_AssignGroupIds( void )
{
int cmixgroupid = 0;
for (int i = 0; i < g_cgrouprules; i++)
{
int mixgroupid = MXR_GetMixgroupFromName( g_grouprules[i].szmixgroup );
if (mixgroupid == -1)
{
// groupname is not yet assigned, provide a unique mixgroupid.
g_grouprules[i].mixgroupid = cmixgroupid;
// save reverse mapping, from mixgroupid to the first grouprules entry for this name
g_mapMixgroupidToGrouprulesid[cmixgroupid] = i;
cmixgroupid++;
}
}
}
int MXR_AddClassname( const char *pName )
{
char szclassname[CMXRNAMEMAX];
Q_strncpy( szclassname, pName, CMXRNAMEMAX );
for ( int i = 0; i < g_cgroupclass; i++ )
{
if ( !Q_stricmp( szclassname, g_groupclasslist[i].szclassname ) )
return i;
}
if ( g_cgroupclass >= CMXRCLASSMAX )
{
Assert(g_cgroupclass < CMXRCLASSMAX);
return -1;
}
Q_memcpy(g_groupclasslist[g_cgroupclass].szclassname, pName, min((size_t)CMXRNAMEMAX-1, strlen(pName)));
g_cgroupclass++;
return g_cgroupclass-1;
}
#define CHAR_LEFT_PAREN '{'
#define CHAR_RIGHT_PAREN '}'
// load group rules and sound mixers from file
bool MXR_LoadAllSoundMixers( void )
{
// init soundmixer globals
g_isoundmixer = -1;
g_szsoundmixer_cur[0] = 0;
g_csoundmixers = 0; // total number of soundmixers found
g_cgrouprules = 0; // total number of group rules found
Q_memset(g_soundmixers, 0, sizeof(g_soundmixers));
Q_memset(g_grouprules, 0, sizeof(g_grouprules));
// load file
// build rules
// build array of sound mixers
char szFile[MAX_OSPATH];
const char *pstart;
bool bResult = false;
char *pbuffer;
Q_snprintf( szFile, sizeof( szFile ), "scripts/soundmixers.txt" );
pbuffer = (char *)COM_LoadFile( szFile, 5, NULL ); // Use malloc - free at end of this routine
if ( !pbuffer )
{
Error( "MXR_LoadAllSoundMixers: unable to open '%s'\n", szFile );
return bResult;
}
pstart = pbuffer;
// first pass: load g_grouprules[]
// starting at top of file,
// scan for first '{', skipping all comment lines
// get strings for: groupname, directory, classname, chan, sndlvl_min, sndlvl_max
// convert chan to CHAN_ lookup
// convert sndlvl_min, sndl_max to ints
// store all in g_grouprules, update g_cgrouprules;
// get next line
// when hit '}' we're done with grouprules
// check for first CHAR_LEFT_PAREN
while (1)
{
pstart = COM_Parse( pstart );
if ( strlen(com_token) <= 0)
break; // eof
if ( com_token[0] != CHAR_LEFT_PAREN )
continue;
break;
}
while (1)
{
pstart = COM_Parse( pstart );
if (com_token[0] == CHAR_RIGHT_PAREN)
break;
grouprule_t *pgroup = &g_grouprules[g_cgrouprules];
// copy mixgroup name, directory, classname
// if no value specified, set to 0 length string
if (com_token[0])
Q_memcpy(pgroup->szmixgroup, com_token, min((size_t)CMXRNAMEMAX-1, strlen(com_token)));
pstart = COM_Parse( pstart );
if (com_token[0])
Q_memcpy(pgroup->szdir, com_token, min((size_t)CMXRNAMEMAX-1, strlen(com_token)));
pgroup->classId = -1;
pstart = COM_Parse( pstart );
if (com_token[0])
{
pgroup->classId = MXR_AddClassname( com_token );
}
// make sure all copied strings are null terminated
pgroup->szmixgroup[CMXRNAMEMAX-1] = 0;
pgroup->szdir[CMXRNAMEMAX-1] = 0;
// lookup chan
pstart = COM_Parse( pstart );
if (com_token[0])
{
if (!Q_stricmp(com_token, "CHAN_STATIC"))
pgroup->chantype = CHAN_STATIC;
else if (!Q_stricmp(com_token, "CHAN_WEAPON"))
pgroup->chantype = CHAN_WEAPON;
else if (!Q_stricmp(com_token, "CHAN_VOICE"))
pgroup->chantype = CHAN_VOICE;
else if (!Q_stricmp(com_token, "CHAN_VOICE2"))
pgroup->chantype = CHAN_VOICE2;
else if (!Q_stricmp(com_token, "CHAN_BODY"))
pgroup->chantype = CHAN_BODY;
else if (!Q_stricmp(com_token, "CHAN_ITEM"))
pgroup->chantype = CHAN_ITEM;
}
else
pgroup->chantype = -1;
// get sndlvls
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->soundlevel_min = atoi(com_token);
else
pgroup->soundlevel_min = -1;
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->soundlevel_max = atoi(com_token);
else
pgroup->soundlevel_max = -1;
// get duck priority, IsDucked, Causes_ducking, duck_target_pct
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->priority = atoi(com_token);
else
pgroup->priority = 50;
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->is_ducked = atoi(com_token);
else
pgroup->is_ducked = 0;
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->causes_ducking = atoi(com_token);
else
pgroup->causes_ducking = 0;
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->duck_target_pct = ((float)(atoi(com_token))) / 100.0f;
else
pgroup->duck_target_pct = 0.5f;
pstart = COM_Parse( pstart );
if (com_token[0])
pgroup->ducker_threshold = ((float)(atoi(com_token))) / 100.0f;
else
pgroup->ducker_threshold = 0.5f;
pgroup->duck_ramp_val = 1.0;
pgroup->duck_target_vol = 1.0;
pgroup->total_vol = 0.0;
// set mixgroup id to -1
pgroup->mixgroupid = -1;
// update rule count
g_cgrouprules++;
if (g_cgrouprules >= CMXRGROUPRULESMAX)
{
// UNDONE: error! too many rules
break;
}
}
// now process all groupids in groups, such that
// each mixgroup gets a unique id.
MXR_AssignGroupIds();
// now load g_soundmixers
// while not at end of file...
// scan for "<name>", if found save as new soundmixer name
// while not '}'
// scan for "<name>", save as groupname
// scan for "<num>", save as mix value
while(1)
{
pstart = COM_Parse( pstart );
if ( strlen(com_token) <= 0)
break; // eof
// save name in soundmixer
soundmixer_t *pmixer = &g_soundmixers[g_csoundmixers];
Q_memcpy(pmixer->szsoundmixer, com_token, min((size_t)CMXRNAMEMAX-1, strlen(com_token)));
// init all mixer values to -1.
for (int j = 0; j < CMXRGROUPMAX; j++)
{
pmixer->mapMixgroupidToValue[j] = -1.0;
}
// load all groupnames for this soundmixer
while (1)
{
pstart = COM_Parse( pstart );
if (com_token[0] == CHAR_LEFT_PAREN)
continue; // skip {
if (com_token[0] == CHAR_RIGHT_PAREN)
break; // finished with this sounmixer
// lookup mixgroupid for groupname
int mixgroupid = MXR_GetMixgroupFromName( com_token );
float value;
// get mix value
pstart = COM_Parse( pstart );
value = atof( com_token );
// store value for mixgroupid
Assert(mixgroupid <= CMXRGROUPMAX);
pmixer->mapMixgroupidToValue[mixgroupid] = value;
}
g_csoundmixers++;
if (g_csoundmixers >= CMXRSOUNDMIXERSMAX)
{
// UNDONE: error! to many sound mixers
break;
}
}
bResult = true;
// loadmxr_exit:
free( pbuffer );
return bResult;
}
void MXR_ReleaseMemory( void )
{
// free all resources
}
float S_GetMono16Samples( const char *pszName, CUtlVector< short >& sampleList )
{
CSfxTable *pSfx = S_PrecacheSound( PSkipSoundChars( pszName ) );
if ( !pSfx )
return 0.0f;
CAudioSource *pWave = pSfx->pSource;
if ( !pWave )
return 0.0f;
int nType = pWave->GetType();
if ( nType != CAudioSource::AUDIO_SOURCE_WAV )
return 0.0f;
CAudioMixer *pMixer = pWave->CreateMixer();
if ( !pMixer )
return 0.0f;
float duration = AudioSource_GetSoundDuration( pSfx );
// Determine start/stop positions
int totalsamples = (int)( duration * pWave->SampleRate() );
if ( totalsamples <= 0 )
return 0;
bool bStereo = pWave->IsStereoWav();
int mix_sample_size = pMixer->GetMixSampleSize();
int nNumChannels = bStereo ? 2 : 1;
char *pData = NULL;
int pos = 0;
int remaining = totalsamples;
while ( remaining > 0 )
{
int blockSize = min( remaining, 1000 );
char copyBuf[AUDIOSOURCE_COPYBUF_SIZE];
int copied = pWave->GetOutputData( (void **)&pData, pos, blockSize, copyBuf );
if ( !copied )
{
break;
}
remaining -= copied;
pos += copied;
// Now get samples out of output data
switch ( nNumChannels )
{
default:
case 1:
{
for ( int i = 0; i < copied; ++i )
{
int offset = i * mix_sample_size;
short sample = 0;
if ( mix_sample_size == 1 )
{
char s = *( char * )( pData + offset );
// Upscale it to fit into a short
sample = s << 8;
}
else if ( mix_sample_size == 2 )
{
sample = *( short * )( pData + offset );
}
else if ( mix_sample_size == 4 )
{
// Not likely to have 4 bytes mono!!!
Assert( 0 );
int s = *( int * )( pData + offset );
sample = s >> 16;
}
else
{
Assert( 0 );
}
sampleList.AddToTail( sample );
}
}
break;
case 2:
{
for ( int i = 0; i < copied; ++i )
{
int offset = i * mix_sample_size;
short left = 0;
short right = 0;
if ( mix_sample_size == 1 )
{
// Not possible!!!, must be at least 2 bytes!!!
Assert( 0 );
char v = *( char * )( pData + offset );
left = right = ( v << 8 );
}
else if ( mix_sample_size == 2 )
{
// One byte per channel
left = (short)( ( *(char *)( pData + offset ) ) << 8 );
right = (short)( ( *(char *)( pData + offset + 1 ) ) << 8 );
}
else if ( mix_sample_size == 4 )
{
// 2 bytes per channel
left = *( short * )( pData + offset );
right = *( short * )( pData + offset + 2 );
}
else
{
Assert( 0 );
}
short sample = ( left + right ) >> 1;
sampleList.AddToTail( sample );
}
}
break;
}
}
delete pMixer;
return duration;
}