mirror of https://github.com/FWGS/xash3d-fwgs
engine: client: sound: slight refactoring, removed unused s_listener.velocity
This commit is contained in:
parent
3723ac60ef
commit
13274655d4
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@ -1605,7 +1605,6 @@ void SND_UpdateSound( void )
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// release raw-channels that no longer used more than 10 secs
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S_FreeIdleRawChannels();
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VectorCopy( cl.simvel, s_listener.velocity );
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s_listener.frametime = (cl.time - cl.oldtime);
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s_listener.waterlevel = cl.local.waterlevel;
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s_listener.active = CL_IsInGame();
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@ -17,30 +17,56 @@ GNU General Public License for more details.
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#include "sound.h"
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#include "client.h"
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#define IPAINTBUFFER 0
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#define IROOMBUFFER 1
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#define ISTREAMBUFFER 2
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enum
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{
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IPAINTBUFFER = 0,
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IROOMBUFFER,
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ISTREAMBUFFER,
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CPAINTBUFFERS,
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};
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#define FILTERTYPE_NONE 0
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#define FILTERTYPE_LINEAR 1
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#define FILTERTYPE_CUBIC 2
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enum
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{
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FILTERTYPE_NONE = 0,
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FILTERTYPE_LINEAR,
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FILTERTYPE_CUBIC,
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};
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#define CCHANVOLUMES 2
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#define SND_SCALE_BITS 7
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#define SND_SCALE_SHIFT (8 - SND_SCALE_BITS)
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#define SND_SCALE_LEVELS (1 << SND_SCALE_BITS)
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#define SND_SCALE_BITS 7
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#define SND_SCALE_SHIFT ( 8 - SND_SCALE_BITS )
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#define SND_SCALE_LEVELS ( 1 << SND_SCALE_BITS )
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portable_samplepair_t *g_curpaintbuffer;
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portable_samplepair_t streambuffer[(PAINTBUFFER_SIZE+1)];
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portable_samplepair_t paintbuffer[(PAINTBUFFER_SIZE+1)];
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portable_samplepair_t roombuffer[(PAINTBUFFER_SIZE+1)];
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portable_samplepair_t facingbuffer[(PAINTBUFFER_SIZE+1)];
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portable_samplepair_t temppaintbuffer[(PAINTBUFFER_SIZE+1)];
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paintbuffer_t paintbuffers[CPAINTBUFFERS];
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// sound mixing buffer
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#define CPAINTFILTERMEM 3
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#define CPAINTFILTERS 4 // maximum number of consecutive upsample passes per paintbuffer
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int snd_scaletable[SND_SCALE_LEVELS][256];
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// fixed point stuff for real-time resampling
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#define FIX_BITS 28
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#define FIX_SCALE ( 1 << FIX_BITS )
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#define FIX_MASK (( 1 << FIX_BITS ) - 1 )
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#define FIX_FLOAT( a ) ((int)(( a ) * FIX_SCALE ))
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#define FIX( a ) (((int)( a )) << FIX_BITS )
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#define FIX_INTPART( a ) (((int)( a )) >> FIX_BITS )
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#define FIX_FRACPART( a ) (( a ) & FIX_MASK )
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typedef struct
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{
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qboolean factive; // if true, mix to this paintbuffer using flags
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portable_samplepair_t *pbuf; // front stereo mix buffer, for 2 or 4 channel mixing
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int ifilter; // current filter memory buffer to use for upsampling pass
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portable_samplepair_t fltmem[CPAINTFILTERS][CPAINTFILTERMEM];
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} paintbuffer_t;
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static portable_samplepair_t *g_curpaintbuffer;
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static portable_samplepair_t streambuffer[(PAINTBUFFER_SIZE+1)];
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static portable_samplepair_t paintbuffer[(PAINTBUFFER_SIZE+1)];
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static portable_samplepair_t roombuffer[(PAINTBUFFER_SIZE+1)];
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static portable_samplepair_t temppaintbuffer[(PAINTBUFFER_SIZE+1)];
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static paintbuffer_t paintbuffers[CPAINTBUFFERS];
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static int snd_scaletable[SND_SCALE_LEVELS][256];
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void S_InitScaletable( void )
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{
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int i, j;
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@ -67,7 +93,7 @@ static void S_TransferPaintBuffer( int endtime )
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dword *pbuf;
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pbuf = (dword *)dma.buffer;
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snd_p = (int *)PAINTBUFFER;
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snd_p = (int *)g_curpaintbuffer;
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lpaintedtime = paintedtime;
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sampleMask = ((dma.samples >> 1) - 1);
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@ -568,6 +594,7 @@ static void MIX_MixChannelsToPaintbuffer( int endtime, int rate, int outputRate
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// Don't mix sound data for sounds with zero volume. If it's a non-looping sound,
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// just remove the sound when its volume goes to zero.
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bZeroVolume = !ch->leftvol && !ch->rightvol;
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if( !bZeroVolume )
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@ -839,6 +866,14 @@ static void MIX_MixPaintbuffers( int ibuf1, int ibuf2, int ibuf3, int count, flo
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pbuf2 = paintbuffers[ibuf2].pbuf;
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pbuf3 = paintbuffers[ibuf3].pbuf;
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if( !gain )
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{
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// do not mix buf2 into buf3, just copy
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if( pbuf1 != pbuf3 )
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memcpy( pbuf3, pbuf1, sizeof( *pbuf1 ) * count );
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return;
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}
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// destination buffer stereo - average n chans down to stereo
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// destination 2ch:
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@ -21,205 +21,170 @@ extern poolhandle_t sndpool;
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#include "xash3d_mathlib.h"
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// sound engine rate defines
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#define SOUND_DMA_SPEED 44100 // hardware playback rate
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#define SOUND_11k 11025 // 11khz sample rate
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#define SOUND_16k 16000 // 16khz sample rate
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#define SOUND_22k 22050 // 22khz sample rate
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#define SOUND_32k 32000 // 32khz sample rate
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#define SOUND_44k 44100 // 44khz sample rate
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#define DMA_MSEC_PER_SAMPLE ((float)(1000.0 / SOUND_DMA_SPEED))
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// fixed point stuff for real-time resampling
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#define FIX_BITS 28
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#define FIX_SCALE (1 << FIX_BITS)
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#define FIX_MASK ((1 << FIX_BITS)-1)
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#define FIX_FLOAT(a) ((int)((a) * FIX_SCALE))
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#define FIX(a) (((int)(a)) << FIX_BITS)
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#define FIX_INTPART(a) (((int)(a)) >> FIX_BITS)
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#define FIX_FRACTION(a,b) (FIX(a)/(b))
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#define FIX_FRACPART(a) ((a) & FIX_MASK)
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#define SOUND_11k 11025 // 11khz sample rate
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#define SOUND_16k 16000 // 16khz sample rate
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#define SOUND_22k 22050 // 22khz sample rate
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#define SOUND_32k 32000 // 32khz sample rate
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#define SOUND_44k 44100 // 44khz sample rate
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#define SOUND_DMA_SPEED SOUND_44k // hardware playback rate
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// NOTE: clipped sound at 32760 to avoid overload
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#define CLIP( x ) (( x ) > 32760 ? 32760 : (( x ) < -32760 ? -32760 : ( x )))
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#define CLIP( x ) (( x ) > 32760 ? 32760 : (( x ) < -32760 ? -32760 : ( x )))
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#define PAINTBUFFER_SIZE 1024 // 44k: was 512
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#define PAINTBUFFER (g_curpaintbuffer)
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#define CPAINTBUFFERS 3
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#define PAINTBUFFER_SIZE 1024 // 44k: was 512
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// sound mixing buffer
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#define CPAINTFILTERMEM 3
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#define CPAINTFILTERS 4 // maximum number of consecutive upsample passes per paintbuffer
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#define S_RAW_SOUND_IDLE_SEC 10 // time interval for idling raw sound before it's freed
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#define S_RAW_SOUND_BACKGROUNDTRACK -2
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#define S_RAW_SOUND_SOUNDTRACK -1
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#define S_RAW_SAMPLES_PRECISION_BITS 14
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#define CIN_FRAMETIME (1.0f / 30.0f)
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#define S_RAW_SOUND_IDLE_SEC 10 // time interval for idling raw sound before it's freed
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#define S_RAW_SOUND_BACKGROUNDTRACK -2
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#define S_RAW_SOUND_SOUNDTRACK -1
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#define S_RAW_SAMPLES_PRECISION_BITS 14
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typedef struct
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{
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int left;
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int right;
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int left;
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int right;
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} portable_samplepair_t;
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typedef struct
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{
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qboolean factive; // if true, mix to this paintbuffer using flags
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portable_samplepair_t *pbuf; // front stereo mix buffer, for 2 or 4 channel mixing
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int ifilter; // current filter memory buffer to use for upsampling pass
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portable_samplepair_t fltmem[CPAINTFILTERS][CPAINTFILTERMEM];
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} paintbuffer_t;
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typedef struct sfx_s
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{
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char name[MAX_QPATH];
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wavdata_t *cache;
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char name[MAX_QPATH];
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wavdata_t *cache;
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int servercount;
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uint hashValue;
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struct sfx_s *hashNext;
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int servercount;
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uint hashValue;
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struct sfx_s *hashNext;
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} sfx_t;
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extern portable_samplepair_t paintbuffer[];
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extern portable_samplepair_t roombuffer[];
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extern portable_samplepair_t temppaintbuffer[];
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extern portable_samplepair_t *g_curpaintbuffer;
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extern paintbuffer_t paintbuffers[];
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// structure used for fading in and out client sound volume.
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typedef struct
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{
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float initial_percent;
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float percent; // how far to adjust client's volume down by.
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float starttime; // GetHostTime() when we started adjusting volume
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float fadeouttime; // # of seconds to get to faded out state
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float holdtime; // # of seconds to hold
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float fadeintime; // # of seconds to restore
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float initial_percent;
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float percent; // how far to adjust client's volume down by.
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float starttime; // GetHostTime() when we started adjusting volume
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float fadeouttime; // # of seconds to get to faded out state
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float holdtime; // # of seconds to hold
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float fadeintime; // # of seconds to restore
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} soundfade_t;
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typedef struct
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{
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float percent;
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float percent;
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} musicfade_t;
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typedef struct snd_format_s
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{
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unsigned int speed;
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unsigned int width;
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unsigned int channels;
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uint speed;
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byte width;
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byte channels;
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} snd_format_t;
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typedef struct
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{
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snd_format_t format;
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int samples; // mono samples in buffer
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int samplepos; // in mono samples
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byte *buffer;
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qboolean initialized; // sound engine is active
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const char *backendName;
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snd_format_t format;
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int samples; // mono samples in buffer
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int samplepos; // in mono samples
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qboolean initialized; // sound engine is active
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byte *buffer;
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const char *backendName;
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} dma_t;
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#include "vox.h"
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typedef struct
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{
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double sample;
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wavdata_t *pData;
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double forcedEndSample;
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qboolean finished;
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double sample;
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wavdata_t *pData;
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double forcedEndSample;
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qboolean finished;
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} mixer_t;
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typedef struct rawchan_s
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{
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int entnum;
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int master_vol;
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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float dist_mult; // distance multiplier (attenuation/clipK)
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vec3_t origin; // only use if fixed_origin is set
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volatile uint s_rawend;
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wavdata_t sound_info; // advance play position
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float oldtime; // catch time jumps
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size_t max_samples; // buffer length
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portable_samplepair_t rawsamples[1]; // variable sized
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int entnum;
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int master_vol;
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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float dist_mult; // distance multiplier (attenuation/clipK)
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vec3_t origin; // only use if fixed_origin is set
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volatile uint s_rawend;
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float oldtime; // catch time jumps
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wavdata_t sound_info; // advance play position
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size_t max_samples; // buffer length
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portable_samplepair_t rawsamples[1]; // variable sized
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} rawchan_t;
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typedef struct channel_s
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{
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char name[16]; // keept sentence name
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sfx_t *sfx; // sfx number
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char name[16]; // keep sentence name
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sfx_t *sfx; // sfx number
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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int leftvol; // 0-255 left volume
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int rightvol; // 0-255 right volume
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int entnum; // entity soundsource
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int entchannel; // sound channel (CHAN_STREAM, CHAN_VOICE, etc.)
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vec3_t origin; // only use if fixed_origin is set
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float dist_mult; // distance multiplier (attenuation/clipK)
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int master_vol; // 0-255 master volume
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qboolean isSentence; // bit who indicated sentence
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int basePitch; // base pitch percent (100% is normal pitch playback)
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float pitch; // real-time pitch after any modulation or shift by dynamic data
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qboolean use_loop; // don't loop default and local sounds
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qboolean staticsound; // use origin instead of fetching entnum's origin
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qboolean localsound; // it's a local menu sound (not looped, not paused)
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mixer_t pMixer;
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int entnum; // entity soundsource
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int entchannel; // sound channel (CHAN_STREAM, CHAN_VOICE, etc.)
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vec3_t origin; // only use if fixed_origin is set
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float dist_mult; // distance multiplier (attenuation/clipK)
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int master_vol; // 0-255 master volume
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int basePitch; // base pitch percent (100% is normal pitch playback)
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float pitch; // real-time pitch after any modulation or shift by dynamic data
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qboolean use_loop; // don't loop default and local sounds
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qboolean staticsound; // use origin instead of fetching entnum's origin
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qboolean localsound; // it's a local menu sound (not looped, not paused)
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mixer_t pMixer;
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// sentence mixer
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int wordIndex;
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mixer_t *currentWord; // NULL if sentence is finished
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voxword_t words[CVOXWORDMAX];
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qboolean isSentence; // bit indicating sentence
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int wordIndex;
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mixer_t *currentWord; // NULL if sentence is finished
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voxword_t words[CVOXWORDMAX];
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} channel_t;
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typedef struct
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{
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vec3_t origin; // simorg + view_ofs
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vec3_t velocity;
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vec3_t forward;
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vec3_t right;
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vec3_t up;
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vec3_t origin; // simorg + view_ofs
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vec3_t forward;
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vec3_t right;
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vec3_t up;
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int entnum;
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int waterlevel;
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float frametime; // used for sound fade
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qboolean active;
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qboolean inmenu; // listener in-menu ?
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qboolean paused;
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qboolean streaming; // playing AVI-file
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qboolean stream_paused; // pause only background track
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int entnum;
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int waterlevel;
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float frametime; // used for sound fade
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qboolean active;
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qboolean inmenu; // listener in-menu ?
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qboolean paused;
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qboolean streaming; // playing AVI-file
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qboolean stream_paused; // pause only background track
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} listener_t;
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typedef struct
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{
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string current; // a currently playing track
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string loopName; // may be empty
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stream_t *stream;
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int source; // may be game, menu, etc
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string current; // a currently playing track
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string loopName; // may be empty
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stream_t *stream;
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int source; // may be game, menu, etc
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} bg_track_t;
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//====================================================================
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#define MAX_DYNAMIC_CHANNELS (60 + NUM_AMBIENTS)
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#define MAX_CHANNELS (256 + MAX_DYNAMIC_CHANNELS) // Scourge Of Armagon has too many static sounds on hip2m4.bsp
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#define MAX_RAW_CHANNELS 48
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#define MAX_RAW_SAMPLES 8192
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#define MAX_DYNAMIC_CHANNELS (60 + NUM_AMBIENTS)
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#define MAX_CHANNELS (256 + MAX_DYNAMIC_CHANNELS) // Scourge Of Armagon has too many static sounds on hip2m4.bsp
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#define MAX_RAW_CHANNELS 48
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#define MAX_RAW_SAMPLES 8192
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extern sound_t ambient_sfx[NUM_AMBIENTS];
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extern qboolean snd_ambient;
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extern channel_t channels[MAX_CHANNELS];
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extern rawchan_t *raw_channels[MAX_RAW_CHANNELS];
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extern int total_channels;
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extern int paintedtime;
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extern int soundtime;
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extern listener_t s_listener;
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extern int idsp_room;
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extern dma_t dma;
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extern sound_t ambient_sfx[NUM_AMBIENTS];
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extern qboolean snd_ambient;
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extern channel_t channels[MAX_CHANNELS];
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extern rawchan_t *raw_channels[MAX_RAW_CHANNELS];
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extern int total_channels;
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extern int paintedtime;
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extern int soundtime;
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extern listener_t s_listener;
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extern int idsp_room;
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extern dma_t dma;
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extern convar_t s_musicvolume;
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extern convar_t s_lerping;
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extern convar_t s_test; // cvar to testify new effects
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extern convar_t s_musicvolume;
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extern convar_t s_lerping;
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extern convar_t s_test; // cvar to test new effects
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extern convar_t s_samplecount;
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extern convar_t s_warn_late_precache;
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@ -16,21 +16,20 @@ GNU General Public License for more details.
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#ifndef VOX_H
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#define VOX_H
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#define CVOXWORDMAX 64
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#define SENTENCE_INDEX -99999 // unique sentence index
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#define CVOXWORDMAX 64
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#define SENTENCE_INDEX -99999 // unique sentence index
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typedef struct voxword_s
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{
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int volume; // increase percent, ie: 125 = 125% increase
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int pitch; // pitch shift up percent
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int start; // offset start of wave percent
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int end; // offset end of wave percent
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int cbtrim; // end of wave after being trimmed to 'end'
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int fKeepCached; // 1 if this word was already in cache before sentence referenced it
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int samplefrac; // if pitch shifting, this is position into wav * 256
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int timecompress; // % of wave to skip during playback (causes no pitch shift)
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sfx_t *sfx; // name and cache pointer
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int volume; // increase percent, ie: 125 = 125% increase
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int pitch; // pitch shift up percent
|
||||
int start; // offset start of wave percent
|
||||
int end; // offset end of wave percent
|
||||
int cbtrim; // end of wave after being trimmed to 'end'
|
||||
int fKeepCached; // 1 if this word was already in cache before sentence referenced it
|
||||
int samplefrac; // if pitch shifting, this is position into wav * 256
|
||||
int timecompress; // % of wave to skip during playback (causes no pitch shift)
|
||||
sfx_t *sfx; // name and cache pointer
|
||||
} voxword_t;
|
||||
|
||||
struct channel_s;
|
||||
|
|
|
|||
Loading…
Reference in New Issue