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997 lines
21 KiB
C++
997 lines
21 KiB
C++
/* (c) Magnus Auvinen. See licence.txt in the root of the distribution for more information. */
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/* If you are missing that file, acquire a complete release at teeworlds.com. */
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#include <base/math.h>
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#include <base/system.h>
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#include <engine/graphics.h>
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#include <engine/storage.h>
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#include <engine/shared/config.h>
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#include "SDL.h"
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#include "sound.h"
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extern "C"
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{
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#if defined(CONF_VIDEORECORDER)
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#include <engine/shared/video.h>
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#endif
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#include <opusfile.h>
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#include <wavpack.h>
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}
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#include <math.h>
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enum
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{
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NUM_SAMPLES = 512,
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NUM_VOICES = 256,
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NUM_CHANNELS = 16,
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};
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struct CSample
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{
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short *m_pData;
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int m_NumFrames;
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int m_Rate;
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int m_Channels;
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int m_LoopStart;
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int m_LoopEnd;
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int m_PausedAt;
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};
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struct CChannel
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{
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int m_Vol;
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int m_Pan;
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};
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struct CVoice
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{
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CSample *m_pSample;
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CChannel *m_pChannel;
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int m_Age; // increases when reused
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int m_Tick;
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int m_Vol; // 0 - 255
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int m_Flags;
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int m_X, m_Y;
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float m_Falloff; // [0.0, 1.0]
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int m_Shape;
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union
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{
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ISound::CVoiceShapeCircle m_Circle;
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ISound::CVoiceShapeRectangle m_Rectangle;
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};
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};
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static CSample m_aSamples[NUM_SAMPLES] = { {0} };
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static CVoice m_aVoices[NUM_VOICES] = { {0} };
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static CChannel m_aChannels[NUM_CHANNELS] = { {255, 0} };
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static LOCK m_SoundLock = 0;
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static int m_CenterX = 0;
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static int m_CenterY = 0;
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static int m_MixingRate = 48000;
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static volatile int m_SoundVolume = 100;
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static int m_NextVoice = 0;
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static int *m_pMixBuffer = 0; // buffer only used by the thread callback function
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static unsigned m_MaxFrames = 0;
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static const void *s_pWVBuffer = 0x0;
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static int s_WVBufferPosition = 0;
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static int s_WVBufferSize = 0;
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const int DefaultDistance = 1500;
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int m_LastBreak = 0;
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// TODO: there should be a faster way todo this
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static short Int2Short(int i)
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{
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if(i > 0x7fff)
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return 0x7fff;
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else if(i < -0x7fff)
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return -0x7fff;
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return i;
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}
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static int IntAbs(int i)
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{
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if(i<0)
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return -i;
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return i;
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}
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static void Mix(short *pFinalOut, unsigned Frames)
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{
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int MasterVol;
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mem_zero(m_pMixBuffer, m_MaxFrames*2*sizeof(int));
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Frames = minimum(Frames, m_MaxFrames);
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// acquire lock while we are mixing
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lock_wait(m_SoundLock);
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MasterVol = m_SoundVolume;
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for(unsigned i = 0; i < NUM_VOICES; i++)
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{
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if(m_aVoices[i].m_pSample)
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{
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// mix voice
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CVoice *v = &m_aVoices[i];
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int *pOut = m_pMixBuffer;
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int Step = v->m_pSample->m_Channels; // setup input sources
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short *pInL = &v->m_pSample->m_pData[v->m_Tick*Step];
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short *pInR = &v->m_pSample->m_pData[v->m_Tick*Step+1];
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unsigned End = v->m_pSample->m_NumFrames-v->m_Tick;
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int Rvol = (int)(v->m_pChannel->m_Vol*(v->m_Vol/255.0f));
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int Lvol = (int)(v->m_pChannel->m_Vol*(v->m_Vol/255.0f));
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// make sure that we don't go outside the sound data
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if(Frames < End)
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End = Frames;
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// check if we have a mono sound
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if(v->m_pSample->m_Channels == 1)
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pInR = pInL;
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// volume calculation
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if(v->m_Flags&ISound::FLAG_POS && v->m_pChannel->m_Pan)
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{
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// TODO: we should respect the channel panning value
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int dx = v->m_X - m_CenterX;
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int dy = v->m_Y - m_CenterY;
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//
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int p = IntAbs(dx);
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float FalloffX = 0.0f;
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float FalloffY = 0.0f;
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int RangeX = 0; // for panning
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bool InVoiceField = false;
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switch(v->m_Shape)
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{
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case ISound::SHAPE_CIRCLE:
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{
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float r = v->m_Circle.m_Radius;
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RangeX = r;
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int Dist = (int)sqrtf((float)dx*dx+dy*dy); // nasty float
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if(Dist < r)
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{
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InVoiceField = true;
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// falloff
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int FalloffDistance = r*v->m_Falloff;
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if(Dist > FalloffDistance)
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FalloffX = FalloffY = (r-Dist)/(r-FalloffDistance);
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else
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FalloffX = FalloffY = 1.0f;
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}
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else
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InVoiceField = false;
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break;
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}
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case ISound::SHAPE_RECTANGLE:
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{
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RangeX = v->m_Rectangle.m_Width/2.0f;
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int abs_dx = abs(dx);
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int abs_dy = abs(dy);
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int w = v->m_Rectangle.m_Width/2.0f;
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int h = v->m_Rectangle.m_Height/2.0f;
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if(abs_dx < w && abs_dy < h)
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{
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InVoiceField = true;
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// falloff
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int fx = v->m_Falloff * w;
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int fy = v->m_Falloff * h;
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FalloffX = abs_dx > fx ? (float)(w-abs_dx)/(w-fx) : 1.0f;
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FalloffY = abs_dy > fy ? (float)(h-abs_dy)/(h-fy) : 1.0f;
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}
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else
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InVoiceField = false;
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break;
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}
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};
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if(InVoiceField)
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{
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// panning
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if(!(v->m_Flags&ISound::FLAG_NO_PANNING))
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{
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if(dx > 0)
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Lvol = ((RangeX-p)*Lvol)/RangeX;
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else
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Rvol = ((RangeX-p)*Rvol)/RangeX;
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}
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{
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Lvol *= FalloffX * FalloffY;
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Rvol *= FalloffX * FalloffY;
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}
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}
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else
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{
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Lvol = 0;
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Rvol = 0;
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}
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}
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// process all frames
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for(unsigned s = 0; s < End; s++)
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{
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*pOut++ += (*pInL)*Lvol;
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*pOut++ += (*pInR)*Rvol;
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pInL += Step;
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pInR += Step;
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v->m_Tick++;
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}
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// free voice if not used any more
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if(v->m_Tick == v->m_pSample->m_NumFrames)
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{
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if(v->m_Flags&ISound::FLAG_LOOP)
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v->m_Tick = 0;
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else
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{
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v->m_pSample = 0;
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v->m_Age++;
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}
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}
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}
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}
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// release the lock
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lock_unlock(m_SoundLock);
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{
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// clamp accumulated values
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// TODO: this seams slow
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for(unsigned i = 0; i < Frames; i++)
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{
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int j = i<<1;
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int vl = ((m_pMixBuffer[j]*MasterVol)/101)>>8;
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int vr = ((m_pMixBuffer[j+1]*MasterVol)/101)>>8;
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pFinalOut[j] = Int2Short(vl);
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pFinalOut[j+1] = Int2Short(vr);
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// dbg_msg("sound", "the real shit: %d %d", pFinalOut[j], pFinalOut[j+1]);
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}
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}
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#if defined(CONF_ARCH_ENDIAN_BIG)
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swap_endian(pFinalOut, sizeof(short), Frames * 2);
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#endif
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}
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static void SdlCallback(void *pUnused, Uint8 *pStream, int Len)
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{
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(void)pUnused;
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#if defined(CONF_VIDEORECORDER)
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if (!(IVideo::Current() && g_Config.m_ClVideoSndEnable))
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Mix((short *)pStream, Len/2/2);
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else
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IVideo::Current()->NextAudioFrame(Mix);
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#else
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Mix((short *)pStream, Len/2/2);
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#endif
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}
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int CSound::Init()
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{
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m_SoundEnabled = 0;
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m_pGraphics = Kernel()->RequestInterface<IEngineGraphics>();
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m_pStorage = Kernel()->RequestInterface<IStorage>();
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SDL_AudioSpec Format, FormatOut;
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m_SoundLock = lock_create();
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if(!g_Config.m_SndEnable)
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return 0;
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if(SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
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{
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dbg_msg("gfx", "unable to init SDL audio: %s", SDL_GetError());
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return -1;
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}
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m_MixingRate = g_Config.m_SndRate;
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// Set 16-bit stereo audio at 22Khz
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Format.freq = g_Config.m_SndRate; // ignore_convention
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Format.format = AUDIO_S16; // ignore_convention
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Format.channels = 2; // ignore_convention
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Format.samples = g_Config.m_SndBufferSize; // ignore_convention
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Format.callback = SdlCallback; // ignore_convention
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Format.userdata = NULL; // ignore_convention
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// Open the audio device and start playing sound!
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m_Device = SDL_OpenAudioDevice(NULL, 0, &Format, &FormatOut, 0);
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if (m_Device == 0)
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{
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dbg_msg("client/sound", "unable to open audio: %s", SDL_GetError());
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return -1;
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}
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else
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dbg_msg("client/sound", "sound init successful using audio driver '%s'", SDL_GetCurrentAudioDriver());
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m_MaxFrames = FormatOut.samples*2;
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m_pMixBuffer = (int *)calloc(m_MaxFrames * 2, sizeof(int));
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SDL_PauseAudioDevice(m_Device, 0);
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m_SoundEnabled = 1;
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Update(); // update the volume
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return 0;
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}
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int CSound::Update()
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{
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// update volume
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int WantedVolume = g_Config.m_SndVolume;
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if(!m_pGraphics->WindowActive() && g_Config.m_SndNonactiveMute)
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WantedVolume = 0;
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if(WantedVolume != m_SoundVolume)
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{
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lock_wait(m_SoundLock);
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m_SoundVolume = WantedVolume;
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lock_unlock(m_SoundLock);
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}
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//#if defined(CONF_VIDEORECORDER)
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// if(IVideo::Current() && g_Config.m_ClVideoSndEnable)
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// IVideo::Current()->NextAudioFrame(Mix);
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//#endif
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return 0;
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}
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int CSound::Shutdown()
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{
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for(unsigned SampleID = 0; SampleID < NUM_SAMPLES; SampleID++)
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{
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UnloadSample(SampleID);
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}
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SDL_CloseAudioDevice(m_Device);
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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lock_destroy(m_SoundLock);
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free(m_pMixBuffer);
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m_pMixBuffer = 0;
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return 0;
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}
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int CSound::AllocID()
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{
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// TODO: linear search, get rid of it
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for(unsigned SampleID = 0; SampleID < NUM_SAMPLES; SampleID++)
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{
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if(m_aSamples[SampleID].m_pData == 0x0)
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return SampleID;
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}
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return -1;
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}
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void CSound::RateConvert(int SampleID)
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{
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CSample *pSample = &m_aSamples[SampleID];
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int NumFrames = 0;
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short *pNewData = 0;
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// make sure that we need to convert this sound
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if(!pSample->m_pData || pSample->m_Rate == m_MixingRate)
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return;
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// allocate new data
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NumFrames = (int)((pSample->m_NumFrames/(float)pSample->m_Rate)*m_MixingRate);
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pNewData = (short *)calloc(NumFrames * pSample->m_Channels, sizeof(short));
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for(int i = 0; i < NumFrames; i++)
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{
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// resample TODO: this should be done better, like linear at least
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float a = i/(float)NumFrames;
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int f = (int)(a*pSample->m_NumFrames);
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if(f >= pSample->m_NumFrames)
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f = pSample->m_NumFrames-1;
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// set new data
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if(pSample->m_Channels == 1)
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pNewData[i] = pSample->m_pData[f];
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else if(pSample->m_Channels == 2)
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{
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pNewData[i*2] = pSample->m_pData[f*2];
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pNewData[i*2+1] = pSample->m_pData[f*2+1];
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}
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}
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// free old data and apply new
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free(pSample->m_pData);
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pSample->m_pData = pNewData;
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pSample->m_NumFrames = NumFrames;
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pSample->m_Rate = m_MixingRate;
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}
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int CSound::DecodeOpus(int SampleID, const void *pData, unsigned DataSize)
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{
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if(SampleID == -1 || SampleID >= NUM_SAMPLES)
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return -1;
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CSample *pSample = &m_aSamples[SampleID];
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OggOpusFile *OpusFile = op_open_memory((const unsigned char *) pData, DataSize, NULL);
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if (OpusFile)
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{
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int NumChannels = op_channel_count(OpusFile, -1);
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int NumSamples = op_pcm_total(OpusFile, -1); // per channel!
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pSample->m_Channels = NumChannels;
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if(pSample->m_Channels > 2)
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{
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dbg_msg("sound/opus", "file is not mono or stereo.");
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return -1;
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}
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pSample->m_pData = (short *)calloc(NumSamples * NumChannels, sizeof(short));
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int Read;
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int Pos = 0;
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while (Pos < NumSamples)
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{
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Read = op_read(OpusFile, pSample->m_pData + Pos*NumChannels, NumSamples*NumChannels, NULL);
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Pos += Read;
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}
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pSample->m_NumFrames = NumSamples; // ?
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pSample->m_Rate = 48000;
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pSample->m_LoopStart = -1;
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pSample->m_LoopEnd = -1;
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pSample->m_PausedAt = 0;
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}
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else
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{
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dbg_msg("sound/opus", "failed to decode sample");
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return -1;
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}
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return SampleID;
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}
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static int ReadDataOld(void *pBuffer, int Size)
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{
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int ChunkSize = minimum(Size, s_WVBufferSize - s_WVBufferPosition);
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mem_copy(pBuffer, (const char *)s_pWVBuffer + s_WVBufferPosition, ChunkSize);
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s_WVBufferPosition += ChunkSize;
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return ChunkSize;
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}
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#if defined(CONF_WAVPACK_OPEN_FILE_INPUT_EX)
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static int ReadData(void *pId, void *pBuffer, int Size)
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{
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(void)pId;
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return ReadDataOld(pBuffer, Size);
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}
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static int ReturnFalse(void *pId)
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{
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(void)pId;
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return 0;
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}
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static unsigned int GetPos(void *pId)
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{
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(void)pId;
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return s_WVBufferPosition;
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}
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static unsigned int GetLength(void *pId)
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{
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(void)pId;
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return s_WVBufferSize;
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}
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static int PushBackByte(void *pId, int Char)
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{
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s_WVBufferPosition -= 1;
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return 0;
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}
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#endif
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int CSound::DecodeWV(int SampleID, const void *pData, unsigned DataSize)
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{
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if(SampleID == -1 || SampleID >= NUM_SAMPLES)
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return -1;
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CSample *pSample = &m_aSamples[SampleID];
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char aError[100];
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WavpackContext *pContext;
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s_pWVBuffer = pData;
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s_WVBufferSize = DataSize;
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s_WVBufferPosition = 0;
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#if defined(CONF_WAVPACK_OPEN_FILE_INPUT_EX)
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WavpackStreamReader Callback = {0};
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Callback.can_seek = ReturnFalse;
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Callback.get_length = GetLength;
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Callback.get_pos = GetPos;
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Callback.push_back_byte = PushBackByte;
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Callback.read_bytes = ReadData;
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pContext = WavpackOpenFileInputEx(&Callback, (void *)1, 0, aError, 0, 0);
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#else
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pContext = WavpackOpenFileInput(ReadDataOld, aError);
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#endif
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if(pContext)
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{
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int NumSamples = WavpackGetNumSamples(pContext);
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int BitsPerSample = WavpackGetBitsPerSample(pContext);
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unsigned int SampleRate = WavpackGetSampleRate(pContext);
|
|
int NumChannels = WavpackGetNumChannels(pContext);
|
|
int *pSrc;
|
|
short *pDst;
|
|
int i;
|
|
|
|
pSample->m_Channels = NumChannels;
|
|
pSample->m_Rate = SampleRate;
|
|
|
|
if(pSample->m_Channels > 2)
|
|
{
|
|
dbg_msg("sound/wv", "file is not mono or stereo.");
|
|
return -1;
|
|
}
|
|
|
|
if(BitsPerSample != 16)
|
|
{
|
|
dbg_msg("sound/wv", "bps is %d, not 16", BitsPerSample);
|
|
return -1;
|
|
}
|
|
|
|
int *pBuffer = (int *)calloc(NumSamples * NumChannels, sizeof(int));
|
|
WavpackUnpackSamples(pContext, pBuffer, NumSamples); // TODO: check return value
|
|
pSrc = pBuffer;
|
|
|
|
pSample->m_pData = (short *)calloc(NumSamples * NumChannels, sizeof(short));
|
|
pDst = pSample->m_pData;
|
|
|
|
for (i = 0; i < NumSamples*NumChannels; i++)
|
|
*pDst++ = (short)*pSrc++;
|
|
|
|
free(pBuffer);
|
|
#ifdef CONF_WAVPACK_CLOSE_FILE
|
|
WavpackCloseFile(pContext);
|
|
#endif
|
|
|
|
pSample->m_NumFrames = NumSamples;
|
|
pSample->m_LoopStart = -1;
|
|
pSample->m_LoopEnd = -1;
|
|
pSample->m_PausedAt = 0;
|
|
}
|
|
else
|
|
{
|
|
dbg_msg("sound/wv", "failed to decode sample (%s)", aError);
|
|
return -1;
|
|
}
|
|
|
|
return SampleID;
|
|
}
|
|
|
|
int CSound::LoadOpus(const char *pFilename)
|
|
{
|
|
// don't waste memory on sound when we are stress testing
|
|
#ifdef CONF_DEBUG
|
|
if(g_Config.m_DbgStress)
|
|
return -1;
|
|
#endif
|
|
|
|
// no need to load sound when we are running with no sound
|
|
if(!m_SoundEnabled)
|
|
return -1;
|
|
|
|
if(!m_pStorage)
|
|
return -1;
|
|
|
|
IOHANDLE File = m_pStorage->OpenFile(pFilename, IOFLAG_READ, IStorage::TYPE_ALL);
|
|
if(!File)
|
|
{
|
|
dbg_msg("sound/opus", "failed to open file. filename='%s'", pFilename);
|
|
return -1;
|
|
}
|
|
|
|
int SampleID = AllocID();
|
|
int DataSize = io_length(File);
|
|
if(SampleID < 0 || DataSize <= 0)
|
|
{
|
|
io_close(File);
|
|
File = NULL;
|
|
dbg_msg("sound/opus", "failed to open file. filename='%s'", pFilename);
|
|
return -1;
|
|
}
|
|
|
|
// read the whole file into memory
|
|
char *pData = new char[DataSize];
|
|
io_read(File, pData, DataSize);
|
|
|
|
SampleID = DecodeOpus(SampleID, pData, DataSize);
|
|
|
|
delete[] pData;
|
|
io_close(File);
|
|
File = NULL;
|
|
|
|
if(g_Config.m_Debug)
|
|
dbg_msg("sound/opus", "loaded %s", pFilename);
|
|
|
|
RateConvert(SampleID);
|
|
return SampleID;
|
|
}
|
|
|
|
|
|
int CSound::LoadWV(const char *pFilename)
|
|
{
|
|
// don't waste memory on sound when we are stress testing
|
|
#ifdef CONF_DEBUG
|
|
if(g_Config.m_DbgStress)
|
|
return -1;
|
|
#endif
|
|
|
|
// no need to load sound when we are running with no sound
|
|
if(!m_SoundEnabled)
|
|
return -1;
|
|
|
|
if(!m_pStorage)
|
|
return -1;
|
|
|
|
IOHANDLE File = m_pStorage->OpenFile(pFilename, IOFLAG_READ, IStorage::TYPE_ALL);
|
|
if(!File)
|
|
{
|
|
dbg_msg("sound/wv", "failed to open file. filename='%s'", pFilename);
|
|
return -1;
|
|
}
|
|
|
|
int SampleID = AllocID();
|
|
int DataSize = io_length(File);
|
|
if(SampleID < 0 || DataSize <= 0)
|
|
{
|
|
io_close(File);
|
|
File = NULL;
|
|
dbg_msg("sound/wv", "failed to open file. filename='%s'", pFilename);
|
|
return -1;
|
|
}
|
|
|
|
// read the whole file into memory
|
|
char *pData = new char[DataSize];
|
|
io_read(File, pData, DataSize);
|
|
|
|
SampleID = DecodeWV(SampleID, pData, DataSize);
|
|
|
|
delete[] pData;
|
|
io_close(File);
|
|
File = NULL;
|
|
|
|
if(g_Config.m_Debug)
|
|
dbg_msg("sound/wv", "loaded %s", pFilename);
|
|
|
|
RateConvert(SampleID);
|
|
return SampleID;
|
|
}
|
|
|
|
int CSound::LoadOpusFromMem(const void *pData, unsigned DataSize, bool FromEditor = false)
|
|
{
|
|
// don't waste memory on sound when we are stress testing
|
|
#ifdef CONF_DEBUG
|
|
if(g_Config.m_DbgStress)
|
|
return -1;
|
|
#endif
|
|
|
|
// no need to load sound when we are running with no sound
|
|
if(!m_SoundEnabled && !FromEditor)
|
|
return -1;
|
|
|
|
if(!pData)
|
|
return -1;
|
|
|
|
int SampleID = AllocID();
|
|
if(SampleID < 0)
|
|
return -1;
|
|
|
|
SampleID = DecodeOpus(SampleID, pData, DataSize);
|
|
|
|
RateConvert(SampleID);
|
|
return SampleID;
|
|
}
|
|
|
|
int CSound::LoadWVFromMem(const void *pData, unsigned DataSize, bool FromEditor = false)
|
|
{
|
|
// don't waste memory on sound when we are stress testing
|
|
#ifdef CONF_DEBUG
|
|
if(g_Config.m_DbgStress)
|
|
return -1;
|
|
#endif
|
|
|
|
// no need to load sound when we are running with no sound
|
|
if(!m_SoundEnabled && !FromEditor)
|
|
return -1;
|
|
|
|
if(!pData)
|
|
return -1;
|
|
|
|
int SampleID = AllocID();
|
|
if(SampleID < 0)
|
|
return -1;
|
|
|
|
SampleID = DecodeWV(SampleID, pData, DataSize);
|
|
|
|
RateConvert(SampleID);
|
|
return SampleID;
|
|
}
|
|
|
|
void CSound::UnloadSample(int SampleID)
|
|
{
|
|
if(SampleID == -1 || SampleID >= NUM_SAMPLES)
|
|
return;
|
|
|
|
Stop(SampleID);
|
|
free(m_aSamples[SampleID].m_pData);
|
|
|
|
m_aSamples[SampleID].m_pData = 0x0;
|
|
}
|
|
|
|
float CSound::GetSampleDuration(int SampleID)
|
|
{
|
|
if(SampleID == -1 || SampleID >= NUM_SAMPLES)
|
|
return 0.0f;
|
|
|
|
return (m_aSamples[SampleID].m_NumFrames/m_aSamples[SampleID].m_Rate);
|
|
}
|
|
|
|
void CSound::SetListenerPos(float x, float y)
|
|
{
|
|
m_CenterX = (int)x;
|
|
m_CenterY = (int)y;
|
|
}
|
|
|
|
void CSound::SetVoiceVolume(CVoiceHandle Voice, float Volume)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
Volume = clamp(Volume, 0.0f, 1.0f);
|
|
m_aVoices[VoiceID].m_Vol = (int)(Volume*255.0f);
|
|
}
|
|
|
|
void CSound::SetVoiceFalloff(CVoiceHandle Voice, float Falloff)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
Falloff = clamp(Falloff, 0.0f, 1.0f);
|
|
m_aVoices[VoiceID].m_Falloff = Falloff;
|
|
}
|
|
|
|
void CSound::SetVoiceLocation(CVoiceHandle Voice, float x, float y)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
m_aVoices[VoiceID].m_X = x;
|
|
m_aVoices[VoiceID].m_Y = y;
|
|
}
|
|
|
|
void CSound::SetVoiceTimeOffset(CVoiceHandle Voice, float offset)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
lock_wait(m_SoundLock);
|
|
{
|
|
if(m_aVoices[VoiceID].m_pSample)
|
|
{
|
|
int Tick = 0;
|
|
bool IsLooping = m_aVoices[VoiceID].m_Flags&ISound::FLAG_LOOP;
|
|
uint64_t TickOffset = m_aVoices[VoiceID].m_pSample->m_Rate * offset;
|
|
if(m_aVoices[VoiceID].m_pSample->m_NumFrames > 0 && IsLooping)
|
|
Tick = TickOffset % m_aVoices[VoiceID].m_pSample->m_NumFrames;
|
|
else
|
|
Tick = clamp(TickOffset, (uint64_t)0, (uint64_t)m_aVoices[VoiceID].m_pSample->m_NumFrames);
|
|
|
|
// at least 200msec off, else depend on buffer size
|
|
float Threshold = maximum(0.2f * m_aVoices[VoiceID].m_pSample->m_Rate, (float)m_MaxFrames);
|
|
if(abs(m_aVoices[VoiceID].m_Tick-Tick) > Threshold)
|
|
{
|
|
// take care of looping (modulo!)
|
|
if( !(IsLooping && (minimum(m_aVoices[VoiceID].m_Tick, Tick) + m_aVoices[VoiceID].m_pSample->m_NumFrames - maximum(m_aVoices[VoiceID].m_Tick, Tick)) <= Threshold))
|
|
{
|
|
m_aVoices[VoiceID].m_Tick = Tick;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
lock_unlock(m_SoundLock);
|
|
}
|
|
|
|
void CSound::SetVoiceCircle(CVoiceHandle Voice, float Radius)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
m_aVoices[VoiceID].m_Shape = ISound::SHAPE_CIRCLE;
|
|
m_aVoices[VoiceID].m_Circle.m_Radius = maximum(0.0f, Radius);
|
|
}
|
|
|
|
void CSound::SetVoiceRectangle(CVoiceHandle Voice, float Width, float Height)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
m_aVoices[VoiceID].m_Shape = ISound::SHAPE_RECTANGLE;
|
|
m_aVoices[VoiceID].m_Rectangle.m_Width = maximum(0.0f, Width);
|
|
m_aVoices[VoiceID].m_Rectangle.m_Height = maximum(0.0f, Height);
|
|
}
|
|
|
|
void CSound::SetChannel(int ChannelID, float Vol, float Pan)
|
|
{
|
|
m_aChannels[ChannelID].m_Vol = (int)(Vol*255.0f);
|
|
m_aChannels[ChannelID].m_Pan = (int)(Pan*255.0f); // TODO: this is only on and off right now
|
|
}
|
|
|
|
ISound::CVoiceHandle CSound::Play(int ChannelID, int SampleID, int Flags, float x, float y)
|
|
{
|
|
int VoiceID = -1;
|
|
int Age = -1;
|
|
int i;
|
|
|
|
lock_wait(m_SoundLock);
|
|
|
|
// search for voice
|
|
for(i = 0; i < NUM_VOICES; i++)
|
|
{
|
|
int id = (m_NextVoice + i) % NUM_VOICES;
|
|
if(!m_aVoices[id].m_pSample)
|
|
{
|
|
VoiceID = id;
|
|
m_NextVoice = id+1;
|
|
break;
|
|
}
|
|
}
|
|
|
|
// voice found, use it
|
|
if(VoiceID != -1)
|
|
{
|
|
m_aVoices[VoiceID].m_pSample = &m_aSamples[SampleID];
|
|
m_aVoices[VoiceID].m_pChannel = &m_aChannels[ChannelID];
|
|
if(Flags & FLAG_LOOP)
|
|
m_aVoices[VoiceID].m_Tick = m_aSamples[SampleID].m_PausedAt;
|
|
else
|
|
m_aVoices[VoiceID].m_Tick = 0;
|
|
m_aVoices[VoiceID].m_Vol = 255;
|
|
m_aVoices[VoiceID].m_Flags = Flags;
|
|
m_aVoices[VoiceID].m_X = (int)x;
|
|
m_aVoices[VoiceID].m_Y = (int)y;
|
|
m_aVoices[VoiceID].m_Falloff = 0.0f;
|
|
m_aVoices[VoiceID].m_Shape = ISound::SHAPE_CIRCLE;
|
|
m_aVoices[VoiceID].m_Circle.m_Radius = DefaultDistance;
|
|
Age = m_aVoices[VoiceID].m_Age;
|
|
}
|
|
|
|
lock_unlock(m_SoundLock);
|
|
return CreateVoiceHandle(VoiceID, Age);
|
|
}
|
|
|
|
ISound::CVoiceHandle CSound::PlayAt(int ChannelID, int SampleID, int Flags, float x, float y)
|
|
{
|
|
return Play(ChannelID, SampleID, Flags|ISound::FLAG_POS, x, y);
|
|
}
|
|
|
|
ISound::CVoiceHandle CSound::Play(int ChannelID, int SampleID, int Flags)
|
|
{
|
|
return Play(ChannelID, SampleID, Flags, 0, 0);
|
|
}
|
|
|
|
void CSound::Stop(int SampleID)
|
|
{
|
|
// TODO: a nice fade out
|
|
lock_wait(m_SoundLock);
|
|
CSample *pSample = &m_aSamples[SampleID];
|
|
for(int i = 0; i < NUM_VOICES; i++)
|
|
{
|
|
if(m_aVoices[i].m_pSample == pSample)
|
|
{
|
|
if(m_aVoices[i].m_Flags & FLAG_LOOP)
|
|
m_aVoices[i].m_pSample->m_PausedAt = m_aVoices[i].m_Tick;
|
|
else
|
|
m_aVoices[i].m_pSample->m_PausedAt = 0;
|
|
m_aVoices[i].m_pSample = 0;
|
|
}
|
|
}
|
|
lock_unlock(m_SoundLock);
|
|
}
|
|
|
|
void CSound::StopAll()
|
|
{
|
|
// TODO: a nice fade out
|
|
lock_wait(m_SoundLock);
|
|
for(int i = 0; i < NUM_VOICES; i++)
|
|
{
|
|
if(m_aVoices[i].m_pSample)
|
|
{
|
|
if(m_aVoices[i].m_Flags & FLAG_LOOP)
|
|
m_aVoices[i].m_pSample->m_PausedAt = m_aVoices[i].m_Tick;
|
|
else
|
|
m_aVoices[i].m_pSample->m_PausedAt = 0;
|
|
}
|
|
m_aVoices[i].m_pSample = 0;
|
|
}
|
|
lock_unlock(m_SoundLock);
|
|
}
|
|
|
|
void CSound::StopVoice(CVoiceHandle Voice)
|
|
{
|
|
if(!Voice.IsValid())
|
|
return;
|
|
|
|
int VoiceID = Voice.Id();
|
|
|
|
if(m_aVoices[VoiceID].m_Age != Voice.Age())
|
|
return;
|
|
|
|
lock_wait(m_SoundLock);
|
|
{
|
|
m_aVoices[VoiceID].m_pSample = 0;
|
|
m_aVoices[VoiceID].m_Age++;
|
|
}
|
|
lock_unlock(m_SoundLock);
|
|
}
|
|
|
|
|
|
IEngineSound *CreateEngineSound() { return new CSound; }
|