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engine: client: sound: switch to single-pass mixing, now channels gets upscaled directly to DMA speed, fixing the issue with s_lerping

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This commit is contained in:
Alibek Omarov
2026-05-10 02:48:23 +05:00
parent d4205867dd
commit 9c4edaed21
1 changed files with 92 additions and 75 deletions
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167
engine/client/sound/s_mix.c
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@@ -71,6 +71,39 @@ static portable_samplepair_t roombuffer[(PAINTBUFFER_SIZE+1)], paintbuffer[(PAIN
} \
} \
#define S_MakeMixMonoLerp( x ) \
static void S_MixMonoLerp ## x( portable_samplepair_t *pbuf, const int *volume, const void *buf, double offset_frac, double rate_scale, int num_samples ) \
{ \
const int##x##_t *data = buf; \
uint sample_idx = 0; \
for( int i = 0; i < num_samples; i++ ) \
{ \
int s = (int)( data[sample_idx] * ( 1.0 - offset_frac ) + data[sample_idx + 1] * offset_frac ); \
pbuf[i].left += ( s * volume[0] ) >> ( x - 8 ); \
pbuf[i].right += ( s * volume[1] ) >> ( x - 8 ); \
offset_frac += rate_scale; \
sample_idx += (uint)offset_frac; \
offset_frac -= (uint)offset_frac; \
} \
} \
#define S_MakeMixStereoLerp( x ) \
static void S_MixStereoLerp ## x( portable_samplepair_t *pbuf, const int *volume, const void *buf, double offset_frac, double rate_scale, int num_samples ) \
{ \
const int##x##_t *data = buf; \
uint sample_idx = 0; \
for( int i = 0; i < num_samples; i++ ) \
{ \
int sl = (int)( data[sample_idx + 0] * ( 1.0 - offset_frac ) + data[sample_idx + 2] * offset_frac ); \
int sr = (int)( data[sample_idx + 1] * ( 1.0 - offset_frac ) + data[sample_idx + 3] * offset_frac ); \
pbuf[i].left += ( sl * volume[0] ) >> ( x - 8 ); \
pbuf[i].right += ( sr * volume[1] ) >> ( x - 8 ); \
offset_frac += rate_scale; \
sample_idx += (uint)offset_frac << 1; \
offset_frac -= (uint)offset_frac; \
} \
} \
S_MakeMixMono( 8 )
S_MakeMixMono( 16 )
S_MakeMixStereo( 8 )
@@ -79,10 +112,14 @@ S_MakeMixMonoPitch( 8 )
S_MakeMixMonoPitch( 16 )
S_MakeMixStereoPitch( 8 )
S_MakeMixStereoPitch( 16 )
S_MakeMixMonoLerp( 8 )
S_MakeMixMonoLerp( 16 )
S_MakeMixStereoLerp( 8 )
S_MakeMixStereoLerp( 16 )
static void S_MixAudio( portable_samplepair_t *pbuf, const int *pvol, const void *buf, int channels, int width, double offset_frac, double rate_scale, int num_samples )
static void S_MixAudio( portable_samplepair_t *pbuf, const int *pvol, const void *buf, int channels, int width, double offset_frac, double rate_scale, int num_samples, qboolean lerp )
{
if( Q_equal( rate_scale, 1.0 ))
if( Q_equal( rate_scale, 1.0 ) && Q_equal( offset_frac, 0.0 ))
{
if( channels == 1 )
{
@@ -99,6 +136,23 @@ static void S_MixAudio( portable_samplepair_t *pbuf, const int *pvol, const void
S_MixStereo16( pbuf, pvol, buf, num_samples );
}
}
else if( lerp )
{
if( channels == 1 )
{
if( width == 1 )
S_MixMonoLerp8( pbuf, pvol, buf, offset_frac, rate_scale, num_samples );
else
S_MixMonoLerp16( pbuf, pvol, buf, offset_frac, rate_scale, num_samples );
}
else
{
if( width == 1 )
S_MixStereoLerp8( pbuf, pvol, buf, offset_frac, rate_scale, num_samples );
else
S_MixStereoLerp16( pbuf, pvol, buf, offset_frac, rate_scale, num_samples );
}
}
else
{
if( channels == 1 )
@@ -163,13 +217,16 @@ static int S_MixChannelToBuffer( portable_samplepair_t *pbuf, channel_t *chan, i
if( num_samples == 0 )
return 0;
// linear interpolation needs one sample of lookahead beyond the last read position
const int lookahead = s_lerping.value ? 1 : 0;
while( num_samples > 0 )
{
// calculate the last sample position
double end_sample = chan->sample + rate * num_samples * timecompress_rate;
// and get total amount of samples we want
int request_num_samples = (int)(ceil( end_sample ) - floor( chan->sample ));
// and get total amount of samples we want, including lookahead for interpolation
int request_num_samples = (int)(ceil( end_sample ) - floor( chan->sample )) + lookahead;
// get sample pointer and also amount of samples available
const void *audio = NULL;
@@ -181,13 +238,30 @@ static int S_MixChannelToBuffer( portable_samplepair_t *pbuf, channel_t *chan, i
double sample_frac = chan->sample - floor( chan->sample );
// can interpolate only when at least two source samples are available
qboolean lerp = lookahead && available >= 2;
// this is how much data we output
int out_count = num_samples;
if( request_num_samples > available ) // but we can't write more than we have
out_count = (int)ceil(( available - sample_frac ) / ( rate ));
if( request_num_samples > available )
{
// lerp needs sample_idx + 1 to stay within available, so cap by available - 1
if( lerp )
out_count = (int)floor(( available - 1 - sample_frac ) / rate );
else
out_count = (int)ceil(( available - sample_frac ) / rate );
}
// near a buffer boundary (e.g. just before a loop wrap) lerp may yield zero;
// fall back to nearest for one sample to keep chan->sample advancing
if( out_count <= 0 )
{
lerp = false;
out_count = 1;
}
const wavdata_t *wav = chan->sfx->cache;
S_MixAudio( pbuf + offset, pvol, audio, wav->channels, wav->width, sample_frac, rate, out_count );
S_MixAudio( pbuf + offset, pvol, audio, wav->channels, wav->width, sample_frac, rate, out_count, lerp );
chan->sample += out_count * rate * timecompress_rate;
offset += out_count;
@@ -231,11 +305,12 @@ static int VOX_MixChannelToBuffer( portable_samplepair_t *pbuf, channel_t *chan,
return offset;
}
static int S_MixNormalChannels( portable_samplepair_t *dst, int end, int rate )
static int S_MixNormalChannelsToRoombuffer( portable_samplepair_t *dst, int end )
{
const qboolean sp = Host_IsSinglePlayerGame();
const qboolean ingame = CL_IsInGame();
const int num_samples = ( end - snd.paintedtime ) / ( SOUND_DMA_SPEED / rate );
const int out_rate = snd.format.speed;
const int num_samples = end - snd.paintedtime;
// FWGS feature: make everybody sound like chipmunks when we're going fast
const float pitch_mult = ( sys_timescale.value + 1 ) / 2;
@@ -284,7 +359,7 @@ static int S_MixNormalChannels( portable_samplepair_t *dst, int end, int rate )
// if the sound is unaudible, skip it
// if it's also not looping, free it
if( ch->leftvol < 8 && ch->rightvol < 8 )
{
{
if( !FBitSet( sc->flags, SOUND_LOOPED ) || !FBitSet( ch->flags, FL_CHAN_USE_LOOP ))
{
if( ch->inauduble_free_time == 0.0f )
@@ -298,19 +373,18 @@ static int S_MixNormalChannels( portable_samplepair_t *dst, int end, int rate )
ch->inauduble_free_time = 0.0f;
if( rate != sc->rate )
continue;
if( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_STREAM )
{
cl_entity_t *ent = CL_GetEntityByIndex( ch->entnum );
if( ent != NULL )
{
int mouth_count = (int)( num_samples * (double)sc->rate / out_rate );
if( sc->width == 1 )
SND_MoveMouth8( &ent->mouth, ch->sample, sc, num_samples, FBitSet( ch->flags, FL_CHAN_USE_LOOP ));
SND_MoveMouth8( &ent->mouth, ch->sample, sc, mouth_count, FBitSet( ch->flags, FL_CHAN_USE_LOOP ));
else
SND_MoveMouth16( &ent->mouth, ch->sample, sc, num_samples, FBitSet( ch->flags, FL_CHAN_USE_LOOP ));
SND_MoveMouth16( &ent->mouth, ch->sample, sc, mouth_count, FBitSet( ch->flags, FL_CHAN_USE_LOOP ));
}
}
@@ -320,14 +394,14 @@ static int S_MixNormalChannels( portable_samplepair_t *dst, int end, int rate )
if( ch->words )
{
VOX_MixChannelToBuffer( dst, ch, num_samples, rate, pitch );
VOX_MixChannelToBuffer( dst, ch, num_samples, out_rate, pitch );
if( FBitSet( ch->flags, FL_CHAN_SENTENCE_FINISHED ))
S_FreeChannel( ch );
}
else
{
S_MixChannelToBuffer( dst, ch, num_samples, rate, pitch, 0, 0 );
S_MixChannelToBuffer( dst, ch, num_samples, out_rate, pitch, 0, 0 );
if( FBitSet( ch->flags, FL_CHAN_FINISHED ))
S_FreeChannel( ch );
@@ -337,63 +411,6 @@ static int S_MixNormalChannels( portable_samplepair_t *dst, int end, int rate )
return num_mixed_channels;
}
static int S_AverageSample( int a, int b )
{
return ( a >> 1 ) + ( b >> 1 ) + ((( a & 1 ) + ( b & 1 )) >> 1 );
}
static void S_UpsampleBuffer( portable_samplepair_t *dst, size_t num_samples )
{
if( s_lerping.value )
{
// copy even positions and average odd
for( size_t i = num_samples - 1; i > 0; i-- )
{
dst[i * 2] = dst[i];
dst[i * 2 + 1].left = S_AverageSample( dst[i].left, dst[i - 1].left );
dst[i * 2 + 1].right = S_AverageSample( dst[i].right, dst[i - 1].right );
}
}
else
{
// copy into even and odd positions
for( size_t i = num_samples - 1; i > 0; i-- )
{
dst[i * 2] = dst[i];
dst[i * 2 + 1] = dst[i];
}
}
dst[1] = dst[0];
}
static int S_MixNormalChannelsToRoombuffer( int end, int count )
{
// for room buffer we only support CD rates like 11k, 22k, and 44k
// TODO: 48k output would require support from platform-specific backends first
// until there is no real usecase, let's keep it simple
int num_mixed_channels = S_MixNormalChannels( roombuffer, end, SOUND_11k );
if( snd.format.speed >= SOUND_22k )
{
if( num_mixed_channels > 0 )
S_UpsampleBuffer( roombuffer, count / ( SOUND_22k / SOUND_11k ));
num_mixed_channels += S_MixNormalChannels( roombuffer, end, SOUND_22k );
}
if( snd.format.speed >= SOUND_44k )
{
if( num_mixed_channels > 0 )
S_UpsampleBuffer( roombuffer, count / ( SOUND_44k / SOUND_22k ));
num_mixed_channels += S_MixNormalChannels( roombuffer, end, SOUND_44k );
}
return num_mixed_channels;
}
static int S_MixRawChannels( int end )
{
int num_room_channels = 0;
@@ -537,7 +554,7 @@ void S_PaintChannels( int endtime )
S_ClearBuffers( num_samples );
int room_channels = S_MixNormalChannelsToRoombuffer( end, num_samples );
int room_channels = S_MixNormalChannelsToRoombuffer( roombuffer, end );
room_channels += S_MixRawChannels( end );