/* (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
#include
#include
#include
#include "SDL.h"
#include "sound.h"
extern "C" { // wavpack
#include
}
#include
enum
{
NUM_SAMPLES = 512,
NUM_VOICES = 64,
NUM_CHANNELS = 16,
MAX_FRAMES = 1024
};
struct CSample
{
short *m_pData;
int m_NumFrames;
int m_Rate;
int m_Channels;
int m_LoopStart;
int m_LoopEnd;
};
struct CChannel
{
int m_Vol;
int m_Pan;
} ;
struct CVoice
{
CSample *m_pSample;
CChannel *m_pChannel;
int m_Tick;
int m_Vol; // 0 - 255
int m_Flags;
int m_X, m_Y;
} ;
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_SoundEnabled = 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;
// 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 aMixBuffer[MAX_FRAMES*2] = {0};
int MasterVol;
// aquire lock while we are mixing
lock_wait(m_SoundLock);
MasterVol = m_SoundVolume;
for(unsigned i = 0; i < NUM_VOICES; i++)
{
if(m_aVoices[i].m_pSample)
{
// mix voice
CVoice *v = &m_aVoices[i];
int *pOut = aMixBuffer;
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];
unsigned End = v->m_pSample->m_NumFrames-v->m_Tick;
int Rvol = v->m_pChannel->m_Vol;
int Lvol = v->m_pChannel->m_Vol;
// make sure that we don't go outside the sound data
if(Frames < End)
End = Frames;
// 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
const int Range = 1500; // magic value, remove
int dx = v->m_X - m_CenterX;
int dy = v->m_Y - m_CenterY;
int Dist = (int)sqrtf((float)dx*dx+dy*dy); // float here. nasty
int p = IntAbs(dx);
if(Dist < Range)
{
// panning
if(dx > 0)
Lvol = ((Range-p)*Lvol)/Range;
else
Rvol = ((Range-p)*Rvol)/Range;
// falloff
Lvol = (Lvol*(Range-Dist))/Range;
Rvol = (Rvol*(Range-Dist))/Range;
}
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++;
}
// free voice if not used any more
if(v->m_Tick == v->m_pSample->m_NumFrames)
v->m_pSample = 0;
}
}
// release the lock
lock_release(m_SoundLock);
{
// clamp accumulated values
// TODO: this seams slow
for(unsigned i = 0; i < Frames; i++)
{
int j = i<<1;
int vl = ((aMixBuffer[j]*MasterVol)/101)>>8;
int vr = ((aMixBuffer[j+1]*MasterVol)/101)>>8;
pFinalOut[j] = Int2Short(vl);
pFinalOut[j+1] = Int2Short(vr);
}
}
#if defined(CONF_ARCH_ENDIAN_BIG)
swap_endian(pFinalOut, sizeof(short), Frames * 2);
#endif
}
static void SdlCallback(void *pUnused, Uint8 *pStream, int Len)
{
(void)pUnused;
Mix((short *)pStream, Len/2/2);
}
int CSound::Init()
{
m_pGraphics = Kernel()->RequestInterface();
m_pStorage = Kernel()->RequestInterface();
SDL_AudioSpec Format;
m_SoundLock = lock_create();
if(!g_Config.m_SndEnable)
return 0;
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!
if(SDL_OpenAudio(&Format, NULL) < 0)
{
dbg_msg("client/sound", "unable to open audio: %s", SDL_GetError());
return -1;
}
else
dbg_msg("client/sound", "sound init successful");
SDL_PauseAudio(0);
m_SoundEnabled = 1;
Update(); // update the volume
return 0;
}
int CSound::Update()
{
// update volume
int WantedVolume = g_Config.m_SndVolume;
if(!m_pGraphics->WindowActive() && g_Config.m_SndNonactiveMute)
WantedVolume = 0;
if(WantedVolume != m_SoundVolume)
{
lock_wait(m_SoundLock);
m_SoundVolume = WantedVolume;
lock_release(m_SoundLock);
}
return 0;
}
int CSound::Shutdown()
{
SDL_CloseAudio();
lock_destroy(m_SoundLock);
return 0;
}
int CSound::AllocID()
{
// TODO: linear search, get rid of it
for(unsigned SampleID = 0; SampleID < NUM_SAMPLES; SampleID++)
{
if(m_aSamples[SampleID].m_pData == 0x0)
return SampleID;
}
return -1;
}
void CSound::RateConvert(int SampleID)
{
CSample *pSample = &m_aSamples[SampleID];
int NumFrames = 0;
short *pNewData = 0;
// 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 *)mem_alloc(NumFrames*pSample->m_Channels*sizeof(short), 1);
for(int i = 0; i < NumFrames; i++)
{
// resample TODO: this should be done better, like linear atleast
float a = i/(float)NumFrames;
int f = (int)(a*pSample->m_NumFrames);
if(f >= pSample->m_NumFrames)
f = pSample->m_NumFrames-1;
// 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];
}
}
// free old data and apply new
mem_free(pSample->m_pData);
pSample->m_pData = pNewData;
pSample->m_NumFrames = NumFrames;
}
int CSound::ReadData(void *pBuffer, int Size)
{
return io_read(ms_File, pBuffer, Size);
}
int CSound::LoadWV(const char *pFilename)
{
CSample *pSample;
int SampleID = -1;
char aError[100];
WavpackContext *pContext;
// don't waste memory on sound when we are stress testing
if(g_Config.m_DbgStress)
return -1;
// no need to load sound when we are running with no sound
if(!m_SoundEnabled)
return 1;
if(!m_pStorage)
return -1;
ms_File = m_pStorage->OpenFile(pFilename, IOFLAG_READ, IStorage::TYPE_ALL);
if(!ms_File)
{
dbg_msg("sound/wv", "failed to open file. filename='%s'", pFilename);
return -1;
}
SampleID = AllocID();
if(SampleID < 0)
return -1;
pSample = &m_aSamples[SampleID];
pContext = WavpackOpenFileInput(ReadData, aError);
if (pContext)
{
int m_aSamples = WavpackGetNumSamples(pContext);
int BitsPerSample = WavpackGetBitsPerSample(pContext);
unsigned int SampleRate = WavpackGetSampleRate(pContext);
int m_aChannels = WavpackGetNumChannels(pContext);
int *pData;
int *pSrc;
short *pDst;
int i;
pSample->m_Channels = m_aChannels;
pSample->m_Rate = SampleRate;
if(pSample->m_Channels > 2)
{
dbg_msg("sound/wv", "file is not mono or stereo. filename='%s'", pFilename);
return -1;
}
/*
if(snd->rate != 44100)
{
dbg_msg("sound/wv", "file is %d Hz, not 44100 Hz. filename='%s'", snd->rate, filename);
return -1;
}*/
if(BitsPerSample != 16)
{
dbg_msg("sound/wv", "bps is %d, not 16, filname='%s'", BitsPerSample, pFilename);
return -1;
}
pData = (int *)mem_alloc(4*m_aSamples*m_aChannels, 1);
WavpackUnpackSamples(pContext, pData, m_aSamples); // TODO: check return value
pSrc = pData;
pSample->m_pData = (short *)mem_alloc(2*m_aSamples*m_aChannels, 1);
pDst = pSample->m_pData;
for (i = 0; i < m_aSamples*m_aChannels; i++)
*pDst++ = (short)*pSrc++;
mem_free(pData);
pSample->m_NumFrames = m_aSamples;
pSample->m_LoopStart = -1;
pSample->m_LoopEnd = -1;
}
else
{
dbg_msg("sound/wv", "failed to open %s: %s", pFilename, aError);
}
io_close(ms_File);
ms_File = NULL;
if(g_Config.m_Debug)
dbg_msg("sound/wv", "loaded %s", pFilename);
RateConvert(SampleID);
return SampleID;
}
void CSound::SetListenerPos(float x, float y)
{
m_CenterX = (int)x;
m_CenterY = (int)y;
}
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
}
int CSound::Play(int ChannelID, int SampleID, int Flags, float x, float y)
{
int VoiceID = -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];
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;
}
lock_release(m_SoundLock);
return VoiceID;
}
int CSound::PlayAt(int ChannelID, int SampleID, int Flags, float x, float y)
{
return Play(ChannelID, SampleID, Flags|ISound::FLAG_POS, x, y);
}
int CSound::Play(int ChannelID, int SampleID, int Flags)
{
return Play(ChannelID, SampleID, Flags, 0, 0);
}
void CSound::Stop(int VoiceID)
{
// TODO: a nice fade out
lock_wait(m_SoundLock);
m_aVoices[VoiceID].m_pSample = 0;
lock_release(m_SoundLock);
}
void CSound::StopAll()
{
// TODO: a nice fade out
lock_wait(m_SoundLock);
for(int i = 0; i < NUM_VOICES; i++)
{
m_aVoices[i].m_pSample = 0;
}
lock_release(m_SoundLock);
}
IOHANDLE CSound::ms_File = 0;
IEngineSound *CreateEngineSound() { return new CSound; }