ddnet/src/engine/client/sound.cpp

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