ddnet/src/engine/client/sound.cpp
Robert Müller 9cf3094934 Also check for incorrect sample index with assertion
Ensure the sample being allocated is not currently used also by checking its next free sample index.
2024-04-27 13:11:35 +02:00

983 lines
23 KiB
C++

/* (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 <SDL.h>
#include <base/math.h>
#include <base/system.h>
#include <engine/graphics.h>
#include <engine/shared/config.h>
#include <engine/storage.h>
#include "sound.h"
#if defined(CONF_VIDEORECORDER)
#include <engine/shared/video.h>
#endif
extern "C" {
#include <opusfile.h>
#include <wavpack.h>
}
#include <cmath>
static constexpr int SAMPLE_INDEX_USED = -2;
static constexpr int SAMPLE_INDEX_FULL = -1;
void CSound::Mix(short *pFinalOut, unsigned Frames)
{
Frames = minimum(Frames, m_MaxFrames);
mem_zero(m_pMixBuffer, Frames * 2 * sizeof(int));
// acquire lock while we are mixing
m_SoundLock.lock();
const int MasterVol = m_SoundVolume.load(std::memory_order_relaxed);
for(auto &Voice : m_aVoices)
{
if(!Voice.m_pSample)
continue;
// mix voice
int *pOut = m_pMixBuffer;
const int Step = Voice.m_pSample->m_Channels; // setup input sources
short *pInL = &Voice.m_pSample->m_pData[Voice.m_Tick * Step];
short *pInR = &Voice.m_pSample->m_pData[Voice.m_Tick * Step + 1];
unsigned End = Voice.m_pSample->m_NumFrames - Voice.m_Tick;
int VolumeR = round_truncate(Voice.m_pChannel->m_Vol * (Voice.m_Vol / 255.0f));
int VolumeL = VolumeR;
// make sure that we don't go outside the sound data
if(Frames < End)
End = Frames;
// check if we have a mono sound
if(Voice.m_pSample->m_Channels == 1)
pInR = pInL;
// volume calculation
if(Voice.m_Flags & ISound::FLAG_POS && Voice.m_pChannel->m_Pan)
{
// TODO: we should respect the channel panning value
const int dx = Voice.m_X - m_CenterX.load(std::memory_order_relaxed);
const int dy = Voice.m_Y - m_CenterY.load(std::memory_order_relaxed);
float FalloffX = 0.0f;
float FalloffY = 0.0f;
int RangeX = 0; // for panning
bool InVoiceField = false;
switch(Voice.m_Shape)
{
case ISound::SHAPE_CIRCLE:
{
const float Radius = Voice.m_Circle.m_Radius;
RangeX = Radius;
// dx and dy can be larger than 46341 and thus the calculation would go beyond the limits of a integer,
// therefore we cast them into float
const int Dist = (int)length(vec2(dx, dy));
if(Dist < Radius)
{
InVoiceField = true;
// falloff
int FalloffDistance = Radius * Voice.m_Falloff;
if(Dist > FalloffDistance)
FalloffX = FalloffY = (Radius - Dist) / (Radius - FalloffDistance);
else
FalloffX = FalloffY = 1.0f;
}
else
InVoiceField = false;
break;
}
case ISound::SHAPE_RECTANGLE:
{
RangeX = Voice.m_Rectangle.m_Width / 2.0f;
const int abs_dx = absolute(dx);
const int abs_dy = absolute(dy);
const int w = Voice.m_Rectangle.m_Width / 2.0f;
const int h = Voice.m_Rectangle.m_Height / 2.0f;
if(abs_dx < w && abs_dy < h)
{
InVoiceField = true;
// falloff
int fx = Voice.m_Falloff * w;
int fy = Voice.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)
{
// panning
if(!(Voice.m_Flags & ISound::FLAG_NO_PANNING))
{
if(dx > 0)
VolumeL = ((RangeX - absolute(dx)) * VolumeL) / RangeX;
else
VolumeR = ((RangeX - absolute(dx)) * VolumeR) / RangeX;
}
{
VolumeL *= FalloffX * FalloffY;
VolumeR *= FalloffX * FalloffY;
}
}
else
{
VolumeL = 0;
VolumeR = 0;
}
}
// process all frames
for(unsigned s = 0; s < End; s++)
{
*pOut++ += (*pInL) * VolumeL;
*pOut++ += (*pInR) * VolumeR;
pInL += Step;
pInR += Step;
Voice.m_Tick++;
}
// free voice if not used any more
if(Voice.m_Tick == Voice.m_pSample->m_NumFrames)
{
if(Voice.m_Flags & ISound::FLAG_LOOP)
Voice.m_Tick = 0;
else
{
Voice.m_pSample = nullptr;
Voice.m_Age++;
}
}
}
m_SoundLock.unlock();
// clamp accumulated values
for(unsigned i = 0; i < Frames * 2; i++)
pFinalOut[i] = clamp<int>(((m_pMixBuffer[i] * MasterVol) / 101) >> 8, std::numeric_limits<short>::min(), std::numeric_limits<short>::max());
#if defined(CONF_ARCH_ENDIAN_BIG)
swap_endian(pFinalOut, sizeof(short), Frames * 2);
#endif
}
static void SdlCallback(void *pUser, Uint8 *pStream, int Len)
{
CSound *pSound = static_cast<CSound *>(pUser);
#if defined(CONF_VIDEORECORDER)
if(!(IVideo::Current() && g_Config.m_ClVideoSndEnable))
{
pSound->Mix((short *)pStream, Len / sizeof(short) / 2);
}
else
{
mem_zero(pStream, Len);
}
#else
pSound->Mix((short *)pStream, Len / sizeof(short) / 2);
#endif
}
int CSound::Init()
{
m_SoundEnabled = false;
m_pGraphics = Kernel()->RequestInterface<IEngineGraphics>();
m_pStorage = Kernel()->RequestInterface<IStorage>();
if(!g_Config.m_SndEnable)
return 0;
if(SDL_InitSubSystem(SDL_INIT_AUDIO) < 0)
{
dbg_msg("sound", "unable to init SDL audio: %s", SDL_GetError());
return -1;
}
m_MixingRate = g_Config.m_SndRate;
SDL_AudioSpec Format, FormatOut;
Format.freq = m_MixingRate;
Format.format = AUDIO_S16;
Format.channels = 2;
Format.samples = g_Config.m_SndBufferSize;
Format.callback = SdlCallback;
Format.userdata = this;
// Open the audio device and start playing sound!
m_Device = SDL_OpenAudioDevice(nullptr, 0, &Format, &FormatOut, 0);
if(m_Device == 0)
{
dbg_msg("sound", "unable to open audio: %s", SDL_GetError());
return -1;
}
else
dbg_msg("sound", "sound init successful using audio driver '%s'", SDL_GetCurrentAudioDriver());
m_MaxFrames = FormatOut.samples * 2;
#if defined(CONF_VIDEORECORDER)
m_MaxFrames = maximum<uint32_t>(m_MaxFrames, 1024 * 2); // make the buffer bigger just in case
#endif
m_pMixBuffer = (int *)calloc(m_MaxFrames * 2, sizeof(int));
m_FirstFreeSampleIndex = 0;
for(size_t i = 0; i < std::size(m_aSamples) - 1; ++i)
{
m_aSamples[i].m_Index = i;
m_aSamples[i].m_NextFreeSampleIndex = i + 1;
}
m_aSamples[std::size(m_aSamples) - 1].m_Index = std::size(m_aSamples) - 1;
m_aSamples[std::size(m_aSamples) - 1].m_NextFreeSampleIndex = SAMPLE_INDEX_FULL;
SDL_PauseAudioDevice(m_Device, 0);
m_SoundEnabled = true;
Update();
return 0;
}
int CSound::Update()
{
UpdateVolume();
return 0;
}
void CSound::UpdateVolume()
{
int WantedVolume = g_Config.m_SndVolume;
if(!m_pGraphics->WindowActive() && g_Config.m_SndNonactiveMute)
WantedVolume = 0;
m_SoundVolume.store(WantedVolume, std::memory_order_relaxed);
}
void CSound::Shutdown()
{
for(unsigned SampleId = 0; SampleId < NUM_SAMPLES; SampleId++)
{
UnloadSample(SampleId);
}
SDL_CloseAudioDevice(m_Device);
SDL_QuitSubSystem(SDL_INIT_AUDIO);
free(m_pMixBuffer);
m_pMixBuffer = nullptr;
}
CSample *CSound::AllocSample()
{
if(m_FirstFreeSampleIndex == SAMPLE_INDEX_FULL)
return nullptr;
CSample *pSample = &m_aSamples[m_FirstFreeSampleIndex];
if(pSample->m_pData != nullptr || pSample->m_NextFreeSampleIndex == SAMPLE_INDEX_USED)
{
char aError[128];
str_format(aError, sizeof(aError), "Sample was not unloaded (index=%d, next=%d, duration=%f, data=%p)",
pSample->m_Index, pSample->m_NextFreeSampleIndex, pSample->TotalTime(), pSample->m_pData);
dbg_assert(false, aError);
}
m_FirstFreeSampleIndex = pSample->m_NextFreeSampleIndex;
pSample->m_NextFreeSampleIndex = SAMPLE_INDEX_USED;
return pSample;
}
void CSound::RateConvert(CSample &Sample) const
{
dbg_assert(Sample.m_pData != nullptr, "Sample is not loaded");
// make sure that we need to convert this sound
if(Sample.m_Rate == m_MixingRate)
return;
// allocate new data
const int NumFrames = (int)((Sample.m_NumFrames / (float)Sample.m_Rate) * m_MixingRate);
short *pNewData = (short *)calloc((size_t)NumFrames * Sample.m_Channels, sizeof(short));
for(int i = 0; i < NumFrames; i++)
{
// resample TODO: this should be done better, like linear at least
float a = i / (float)NumFrames;
int f = (int)(a * Sample.m_NumFrames);
if(f >= Sample.m_NumFrames)
f = Sample.m_NumFrames - 1;
// set new data
if(Sample.m_Channels == 1)
pNewData[i] = Sample.m_pData[f];
else if(Sample.m_Channels == 2)
{
pNewData[i * 2] = Sample.m_pData[f * 2];
pNewData[i * 2 + 1] = Sample.m_pData[f * 2 + 1];
}
}
// free old data and apply new
free(Sample.m_pData);
Sample.m_pData = pNewData;
Sample.m_NumFrames = NumFrames;
Sample.m_Rate = m_MixingRate;
}
bool CSound::DecodeOpus(CSample &Sample, const void *pData, unsigned DataSize) const
{
int OpusError = 0;
OggOpusFile *pOpusFile = op_open_memory((const unsigned char *)pData, DataSize, &OpusError);
if(pOpusFile)
{
const int NumChannels = op_channel_count(pOpusFile, -1);
if(NumChannels > 2)
{
op_free(pOpusFile);
dbg_msg("sound/opus", "file is not mono or stereo.");
return false;
}
const int NumSamples = op_pcm_total(pOpusFile, -1); // per channel!
if(NumSamples < 0)
{
op_free(pOpusFile);
dbg_msg("sound/opus", "failed to get number of samples, error %d", NumSamples);
return false;
}
short *pSampleData = (short *)calloc((size_t)NumSamples * NumChannels, sizeof(short));
int Pos = 0;
while(Pos < NumSamples)
{
const int Read = op_read(pOpusFile, pSampleData + Pos * NumChannels, (NumSamples - Pos) * NumChannels, nullptr);
if(Read < 0)
{
free(pSampleData);
op_free(pOpusFile);
dbg_msg("sound/opus", "op_read error %d at %d", Read, Pos);
return false;
}
else if(Read == 0) // EOF
break;
Pos += Read;
}
op_free(pOpusFile);
Sample.m_pData = pSampleData;
Sample.m_NumFrames = Pos;
Sample.m_Rate = 48000;
Sample.m_Channels = NumChannels;
Sample.m_LoopStart = -1;
Sample.m_LoopEnd = -1;
Sample.m_PausedAt = 0;
}
else
{
dbg_msg("sound/opus", "failed to decode sample, error %d", OpusError);
return false;
}
return true;
}
// TODO: Update WavPack to get rid of these global variables
static const void *s_pWVBuffer = nullptr;
static int s_WVBufferPosition = 0;
static int s_WVBufferSize = 0;
static int ReadDataOld(void *pBuffer, int Size)
{
int ChunkSize = minimum(Size, s_WVBufferSize - s_WVBufferPosition);
mem_copy(pBuffer, (const char *)s_pWVBuffer + s_WVBufferPosition, ChunkSize);
s_WVBufferPosition += ChunkSize;
return ChunkSize;
}
#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;
}
static unsigned int GetPos(void *pId)
{
(void)pId;
return s_WVBufferPosition;
}
static unsigned int GetLength(void *pId)
{
(void)pId;
return s_WVBufferSize;
}
static int PushBackByte(void *pId, int Char)
{
s_WVBufferPosition -= 1;
return 0;
}
#endif
bool CSound::DecodeWV(CSample &Sample, const void *pData, unsigned DataSize) const
{
char aError[100];
dbg_assert(s_pWVBuffer == nullptr, "DecodeWV already in use");
s_pWVBuffer = pData;
s_WVBufferSize = DataSize;
s_WVBufferPosition = 0;
#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;
Callback.read_bytes = ReadData;
WavpackContext *pContext = WavpackOpenFileInputEx(&Callback, (void *)1, 0, aError, 0, 0);
#else
WavpackContext *pContext = WavpackOpenFileInput(ReadDataOld, aError);
#endif
if(pContext)
{
const int NumSamples = WavpackGetNumSamples(pContext);
const int BitsPerSample = WavpackGetBitsPerSample(pContext);
const unsigned int SampleRate = WavpackGetSampleRate(pContext);
const int NumChannels = WavpackGetNumChannels(pContext);
if(NumChannels > 2)
{
dbg_msg("sound/wv", "file is not mono or stereo.");
s_pWVBuffer = nullptr;
return false;
}
if(BitsPerSample != 16)
{
dbg_msg("sound/wv", "bps is %d, not 16", BitsPerSample);
s_pWVBuffer = nullptr;
return false;
}
int *pBuffer = (int *)calloc((size_t)NumSamples * NumChannels, sizeof(int));
if(!WavpackUnpackSamples(pContext, pBuffer, NumSamples))
{
free(pBuffer);
dbg_msg("sound/wv", "WavpackUnpackSamples failed. NumSamples=%d, NumChannels=%d", NumSamples, NumChannels);
s_pWVBuffer = nullptr;
return false;
}
Sample.m_pData = (short *)calloc((size_t)NumSamples * NumChannels, sizeof(short));
int *pSrc = pBuffer;
short *pDst = Sample.m_pData;
for(int i = 0; i < NumSamples * NumChannels; i++)
*pDst++ = (short)*pSrc++;
free(pBuffer);
#ifdef CONF_WAVPACK_CLOSE_FILE
WavpackCloseFile(pContext);
#endif
Sample.m_NumFrames = NumSamples;
Sample.m_Rate = SampleRate;
Sample.m_Channels = NumChannels;
Sample.m_LoopStart = -1;
Sample.m_LoopEnd = -1;
Sample.m_PausedAt = 0;
s_pWVBuffer = nullptr;
}
else
{
dbg_msg("sound/wv", "failed to decode sample (%s)", aError);
s_pWVBuffer = nullptr;
return false;
}
return true;
}
int CSound::LoadOpus(const char *pFilename, int StorageType)
{
// no need to load sound when we are running with no sound
if(!m_SoundEnabled)
return -1;
if(!m_pStorage)
return -1;
CSample *pSample = AllocSample();
if(!pSample)
{
dbg_msg("sound/opus", "failed to allocate sample ID. filename='%s'", pFilename);
return -1;
}
void *pData;
unsigned DataSize;
if(!m_pStorage->ReadFile(pFilename, StorageType, &pData, &DataSize))
{
UnloadSample(pSample->m_Index);
dbg_msg("sound/opus", "failed to open file. filename='%s'", pFilename);
return -1;
}
const bool DecodeSuccess = DecodeOpus(*pSample, pData, DataSize);
free(pData);
if(!DecodeSuccess)
{
UnloadSample(pSample->m_Index);
return -1;
}
if(g_Config.m_Debug)
dbg_msg("sound/opus", "loaded %s", pFilename);
RateConvert(*pSample);
return pSample->m_Index;
}
int CSound::LoadWV(const char *pFilename, int StorageType)
{
// no need to load sound when we are running with no sound
if(!m_SoundEnabled)
return -1;
if(!m_pStorage)
return -1;
CSample *pSample = AllocSample();
if(!pSample)
{
dbg_msg("sound/wv", "failed to allocate sample ID. filename='%s'", pFilename);
return -1;
}
void *pData;
unsigned DataSize;
if(!m_pStorage->ReadFile(pFilename, StorageType, &pData, &DataSize))
{
UnloadSample(pSample->m_Index);
dbg_msg("sound/wv", "failed to open file. filename='%s'", pFilename);
return -1;
}
const bool DecodeSuccess = DecodeWV(*pSample, pData, DataSize);
free(pData);
if(!DecodeSuccess)
{
UnloadSample(pSample->m_Index);
return -1;
}
if(g_Config.m_Debug)
dbg_msg("sound/wv", "loaded %s", pFilename);
RateConvert(*pSample);
return pSample->m_Index;
}
int CSound::LoadOpusFromMem(const void *pData, unsigned DataSize, bool FromEditor = false)
{
// no need to load sound when we are running with no sound
if(!m_SoundEnabled && !FromEditor)
return -1;
if(!pData)
return -1;
CSample *pSample = AllocSample();
if(!pSample)
return -1;
if(!DecodeOpus(*pSample, pData, DataSize))
{
UnloadSample(pSample->m_Index);
return -1;
}
RateConvert(*pSample);
return pSample->m_Index;
}
int CSound::LoadWVFromMem(const void *pData, unsigned DataSize, bool FromEditor = false)
{
// no need to load sound when we are running with no sound
if(!m_SoundEnabled && !FromEditor)
return -1;
if(!pData)
return -1;
CSample *pSample = AllocSample();
if(!pSample)
return -1;
if(!DecodeWV(*pSample, pData, DataSize))
{
UnloadSample(pSample->m_Index);
return -1;
}
RateConvert(*pSample);
return pSample->m_Index;
}
void CSound::UnloadSample(int SampleId)
{
if(SampleId == -1 || SampleId >= NUM_SAMPLES)
return;
Stop(SampleId);
// Free data
CSample &Sample = m_aSamples[SampleId];
free(Sample.m_pData);
Sample.m_pData = nullptr;
// Free slot
if(Sample.m_NextFreeSampleIndex == SAMPLE_INDEX_USED)
{
Sample.m_NextFreeSampleIndex = m_FirstFreeSampleIndex;
m_FirstFreeSampleIndex = Sample.m_Index;
}
}
float CSound::GetSampleTotalTime(int SampleId)
{
if(SampleId == -1 || SampleId >= NUM_SAMPLES)
return 0.0f;
return m_aSamples[SampleId].TotalTime();
}
float CSound::GetSampleCurrentTime(int SampleId)
{
if(SampleId == -1 || SampleId >= NUM_SAMPLES)
return 0.0f;
const CLockScope LockScope(m_SoundLock);
CSample *pSample = &m_aSamples[SampleId];
for(auto &Voice : m_aVoices)
{
if(Voice.m_pSample == pSample)
{
return Voice.m_Tick / (float)pSample->m_Rate;
}
}
return pSample->m_PausedAt / (float)pSample->m_Rate;
}
void CSound::SetSampleCurrentTime(int SampleId, float Time)
{
if(SampleId == -1 || SampleId >= NUM_SAMPLES)
return;
const CLockScope LockScope(m_SoundLock);
CSample *pSample = &m_aSamples[SampleId];
for(auto &Voice : m_aVoices)
{
if(Voice.m_pSample == pSample)
{
Voice.m_Tick = pSample->m_NumFrames * Time;
return;
}
}
pSample->m_PausedAt = pSample->m_NumFrames * Time;
}
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
}
void CSound::SetListenerPos(float x, float y)
{
m_CenterX.store((int)x, std::memory_order_relaxed);
m_CenterY.store((int)y, std::memory_order_relaxed);
}
void CSound::SetVoiceVolume(CVoiceHandle Voice, float Volume)
{
if(!Voice.IsValid())
return;
int VoiceId = Voice.Id();
const CLockScope LockScope(m_SoundLock);
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();
const CLockScope LockScope(m_SoundLock);
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();
const CLockScope LockScope(m_SoundLock);
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 TimeOffset)
{
if(!Voice.IsValid())
return;
int VoiceId = Voice.Id();
const CLockScope LockScope(m_SoundLock);
if(m_aVoices[VoiceId].m_Age != Voice.Age())
return;
if(!m_aVoices[VoiceId].m_pSample)
return;
int Tick = 0;
bool IsLooping = m_aVoices[VoiceId].m_Flags & ISound::FLAG_LOOP;
uint64_t TickOffset = m_aVoices[VoiceId].m_pSample->m_Rate * TimeOffset;
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(absolute(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;
}
}
}
void CSound::SetVoiceCircle(CVoiceHandle Voice, float Radius)
{
if(!Voice.IsValid())
return;
int VoiceId = Voice.Id();
const CLockScope LockScope(m_SoundLock);
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();
const CLockScope LockScope(m_SoundLock);
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);
}
ISound::CVoiceHandle CSound::Play(int ChannelId, int SampleId, int Flags, float x, float y)
{
const CLockScope LockScope(m_SoundLock);
// search for voice
int VoiceId = -1;
for(int i = 0; i < NUM_VOICES; i++)
{
int NextId = (m_NextVoice + i) % NUM_VOICES;
if(!m_aVoices[NextId].m_pSample)
{
VoiceId = NextId;
m_NextVoice = NextId + 1;
break;
}
}
// voice found, use it
int Age = -1;
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 if(Flags & FLAG_PREVIEW)
{
m_aVoices[VoiceId].m_Tick = m_aSamples[SampleId].m_PausedAt;
m_aSamples[SampleId].m_PausedAt = 0;
}
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 = 1500;
Age = m_aVoices[VoiceId].m_Age;
}
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::Pause(int SampleId)
{
// TODO: a nice fade out
const CLockScope LockScope(m_SoundLock);
CSample *pSample = &m_aSamples[SampleId];
for(auto &Voice : m_aVoices)
{
if(Voice.m_pSample == pSample)
{
Voice.m_pSample->m_PausedAt = Voice.m_Tick;
Voice.m_pSample = nullptr;
}
}
}
void CSound::Stop(int SampleId)
{
// TODO: a nice fade out
const CLockScope LockScope(m_SoundLock);
CSample *pSample = &m_aSamples[SampleId];
for(auto &Voice : m_aVoices)
{
if(Voice.m_pSample == pSample)
{
if(Voice.m_Flags & FLAG_LOOP)
Voice.m_pSample->m_PausedAt = Voice.m_Tick;
else
Voice.m_pSample->m_PausedAt = 0;
Voice.m_pSample = nullptr;
}
}
}
void CSound::StopAll()
{
// TODO: a nice fade out
const CLockScope LockScope(m_SoundLock);
for(auto &Voice : m_aVoices)
{
if(Voice.m_pSample)
{
if(Voice.m_Flags & FLAG_LOOP)
Voice.m_pSample->m_PausedAt = Voice.m_Tick;
else
Voice.m_pSample->m_PausedAt = 0;
}
Voice.m_pSample = nullptr;
}
}
void CSound::StopVoice(CVoiceHandle Voice)
{
if(!Voice.IsValid())
return;
int VoiceId = Voice.Id();
const CLockScope LockScope(m_SoundLock);
if(m_aVoices[VoiceId].m_Age != Voice.Age())
return;
m_aVoices[VoiceId].m_pSample = nullptr;
m_aVoices[VoiceId].m_Age++;
}
bool CSound::IsPlaying(int SampleId)
{
const CLockScope LockScope(m_SoundLock);
const CSample *pSample = &m_aSamples[SampleId];
return std::any_of(std::begin(m_aVoices), std::end(m_aVoices), [pSample](const auto &Voice) { return Voice.m_pSample == pSample; });
}
void CSound::PauseAudioDevice()
{
SDL_PauseAudioDevice(m_Device, 1);
}
void CSound::UnpauseAudioDevice()
{
SDL_PauseAudioDevice(m_Device, 0);
}
IEngineSound *CreateEngineSound() { return new CSound; }