| 1 | /* (c) Magnus Auvinen. See licence.txt in the root of the distribution for more information. */ |
| 2 | /* If you are missing that file, acquire a complete release at teeworlds.com. */ |
| 3 | #include "sound.h" |
| 4 | |
| 5 | #include <base/bytes.h> |
| 6 | #include <base/dbg.h> |
| 7 | #include <base/log.h> |
| 8 | #include <base/math.h> |
| 9 | #include <base/mem.h> |
| 10 | #include <base/str.h> |
| 11 | |
| 12 | #include <engine/graphics.h> |
| 13 | #include <engine/shared/config.h> |
| 14 | #include <engine/storage.h> |
| 15 | |
| 16 | #include <SDL.h> |
| 17 | |
| 18 | #if defined(CONF_VIDEORECORDER) |
| 19 | #include <engine/shared/video.h> |
| 20 | #endif |
| 21 | extern "C" { |
| 22 | #include <opusfile.h> |
| 23 | #include <wavpack.h> |
| 24 | } |
| 25 | |
| 26 | #include <cmath> |
| 27 | |
| 28 | static constexpr int SAMPLE_INDEX_USED = -2; |
| 29 | static constexpr int SAMPLE_INDEX_FULL = -1; |
| 30 | |
| 31 | void CSound::Mix(short *pFinalOut, unsigned Frames) |
| 32 | { |
| 33 | Frames = minimum(a: Frames, b: m_MaxFrames); |
| 34 | mem_zero(block: m_pMixBuffer, size: Frames * 2 * sizeof(int)); |
| 35 | |
| 36 | // acquire lock while we are mixing |
| 37 | m_SoundLock.lock(); |
| 38 | |
| 39 | const int MasterVol = m_SoundVolume.load(m: std::memory_order_relaxed); |
| 40 | |
| 41 | for(auto &Voice : m_aVoices) |
| 42 | { |
| 43 | if(!Voice.m_pSample) |
| 44 | continue; |
| 45 | |
| 46 | // mix voice |
| 47 | int *pOut = m_pMixBuffer; |
| 48 | |
| 49 | const int Step = Voice.m_pSample->m_Channels; // setup input sources |
| 50 | short *pInL = &Voice.m_pSample->m_pData[Voice.m_Tick * Step]; |
| 51 | short *pInR = &Voice.m_pSample->m_pData[Voice.m_Tick * Step + 1]; |
| 52 | |
| 53 | unsigned End = Voice.m_pSample->m_NumFrames - Voice.m_Tick; |
| 54 | |
| 55 | int VolumeR = round_truncate(f: Voice.m_pChannel->m_Vol * (Voice.m_Vol / 255.0f)); |
| 56 | int VolumeL = VolumeR; |
| 57 | |
| 58 | // make sure that we don't go outside the sound data |
| 59 | if(Frames < End) |
| 60 | End = Frames; |
| 61 | |
| 62 | // check if we have a mono sound |
| 63 | if(Voice.m_pSample->m_Channels == 1) |
| 64 | pInR = pInL; |
| 65 | |
| 66 | // volume calculation |
| 67 | if(Voice.m_Flags & ISound::FLAG_POS && Voice.m_pChannel->m_Pan) |
| 68 | { |
| 69 | // TODO: we should respect the channel panning value |
| 70 | const vec2 Delta = Voice.m_Position - vec2(m_ListenerPositionX.load(m: std::memory_order_relaxed), m_ListenerPositionY.load(m: std::memory_order_relaxed)); |
| 71 | vec2 Falloff = vec2(0.0f, 0.0f); |
| 72 | |
| 73 | float RangeX = 0.0f; // for panning |
| 74 | bool InVoiceField = false; |
| 75 | |
| 76 | switch(Voice.m_Shape) |
| 77 | { |
| 78 | case ISound::SHAPE_CIRCLE: |
| 79 | { |
| 80 | const float Radius = Voice.m_Circle.m_Radius; |
| 81 | RangeX = Radius; |
| 82 | |
| 83 | const float Dist = length(a: Delta); |
| 84 | if(Dist < Radius) |
| 85 | { |
| 86 | InVoiceField = true; |
| 87 | |
| 88 | // falloff |
| 89 | const float FalloffDistance = Radius * Voice.m_Falloff; |
| 90 | Falloff.x = Falloff.y = Dist > FalloffDistance ? (Radius - Dist) / (Radius - FalloffDistance) : 1.0f; |
| 91 | } |
| 92 | break; |
| 93 | } |
| 94 | |
| 95 | case ISound::SHAPE_RECTANGLE: |
| 96 | { |
| 97 | const vec2 AbsoluteDelta = vec2(absolute(a: Delta.x), absolute(a: Delta.y)); |
| 98 | const float w = Voice.m_Rectangle.m_Width / 2.0f; |
| 99 | const float h = Voice.m_Rectangle.m_Height / 2.0f; |
| 100 | RangeX = w; |
| 101 | |
| 102 | if(AbsoluteDelta.x < w && AbsoluteDelta.y < h) |
| 103 | { |
| 104 | InVoiceField = true; |
| 105 | |
| 106 | // falloff |
| 107 | const vec2 FalloffDistance = vec2(w, h) * Voice.m_Falloff; |
| 108 | Falloff.x = AbsoluteDelta.x > FalloffDistance.x ? (w - AbsoluteDelta.x) / (w - FalloffDistance.x) : 1.0f; |
| 109 | Falloff.y = AbsoluteDelta.y > FalloffDistance.y ? (h - AbsoluteDelta.y) / (h - FalloffDistance.y) : 1.0f; |
| 110 | } |
| 111 | break; |
| 112 | } |
| 113 | }; |
| 114 | |
| 115 | if(InVoiceField) |
| 116 | { |
| 117 | // panning |
| 118 | if(!(Voice.m_Flags & ISound::FLAG_NO_PANNING)) |
| 119 | { |
| 120 | if(Delta.x > 0) |
| 121 | VolumeL = ((RangeX - absolute(a: Delta.x)) * VolumeL) / RangeX; |
| 122 | else |
| 123 | VolumeR = ((RangeX - absolute(a: Delta.x)) * VolumeR) / RangeX; |
| 124 | } |
| 125 | |
| 126 | { |
| 127 | VolumeL *= Falloff.x * Falloff.y; |
| 128 | VolumeR *= Falloff.x * Falloff.y; |
| 129 | } |
| 130 | } |
| 131 | else |
| 132 | { |
| 133 | VolumeL = 0; |
| 134 | VolumeR = 0; |
| 135 | } |
| 136 | } |
| 137 | |
| 138 | // process all frames |
| 139 | for(unsigned s = 0; s < End; s++) |
| 140 | { |
| 141 | *pOut++ += (*pInL) * VolumeL; |
| 142 | *pOut++ += (*pInR) * VolumeR; |
| 143 | pInL += Step; |
| 144 | pInR += Step; |
| 145 | Voice.m_Tick++; |
| 146 | } |
| 147 | |
| 148 | // free voice if not used any more |
| 149 | if(Voice.m_Tick == Voice.m_pSample->m_NumFrames) |
| 150 | { |
| 151 | if(Voice.m_Flags & ISound::FLAG_LOOP) |
| 152 | { |
| 153 | Voice.m_Tick = Voice.m_pSample->m_LoopStart; |
| 154 | } |
| 155 | else |
| 156 | { |
| 157 | Voice.m_pSample = nullptr; |
| 158 | Voice.m_Age++; |
| 159 | } |
| 160 | } |
| 161 | } |
| 162 | |
| 163 | m_SoundLock.unlock(); |
| 164 | |
| 165 | // clamp accumulated values |
| 166 | for(unsigned i = 0; i < Frames * 2; i++) |
| 167 | pFinalOut[i] = std::clamp<int>(val: ((m_pMixBuffer[i] * MasterVol) / 101) >> 8, lo: std::numeric_limits<short>::min(), hi: std::numeric_limits<short>::max()); |
| 168 | |
| 169 | #if defined(CONF_ARCH_ENDIAN_BIG) |
| 170 | swap_endian(pFinalOut, sizeof(short), Frames * 2); |
| 171 | #endif |
| 172 | } |
| 173 | |
| 174 | static void SdlCallback(void *pUser, Uint8 *pStream, int Len) |
| 175 | { |
| 176 | CSound *pSound = static_cast<CSound *>(pUser); |
| 177 | |
| 178 | #if defined(CONF_VIDEORECORDER) |
| 179 | if(!(IVideo::Current() && g_Config.m_ClVideoSndEnable)) |
| 180 | { |
| 181 | pSound->Mix(pFinalOut: (short *)pStream, Frames: Len / sizeof(short) / 2); |
| 182 | } |
| 183 | else |
| 184 | { |
| 185 | mem_zero(block: pStream, size: Len); |
| 186 | } |
| 187 | #else |
| 188 | pSound->Mix((short *)pStream, Len / sizeof(short) / 2); |
| 189 | #endif |
| 190 | } |
| 191 | |
| 192 | int CSound::Init() |
| 193 | { |
| 194 | m_SoundEnabled = false; |
| 195 | m_pGraphics = Kernel()->RequestInterface<IEngineGraphics>(); |
| 196 | m_pStorage = Kernel()->RequestInterface<IStorage>(); |
| 197 | |
| 198 | // Initialize sample indices. We always need them to load sounds in |
| 199 | // the editor even if sound is disabled or failed to be enabled. |
| 200 | const CLockScope LockScope(m_SoundLock); |
| 201 | m_FirstFreeSampleIndex = 0; |
| 202 | for(size_t i = 0; i < std::size(m_aSamples) - 1; ++i) |
| 203 | { |
| 204 | m_aSamples[i].m_Index = i; |
| 205 | m_aSamples[i].m_NextFreeSampleIndex = i + 1; |
| 206 | m_aSamples[i].m_pData = nullptr; |
| 207 | } |
| 208 | m_aSamples[std::size(m_aSamples) - 1].m_Index = std::size(m_aSamples) - 1; |
| 209 | m_aSamples[std::size(m_aSamples) - 1].m_NextFreeSampleIndex = SAMPLE_INDEX_FULL; |
| 210 | |
| 211 | if(!g_Config.m_SndEnable) |
| 212 | return 0; |
| 213 | |
| 214 | if(SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) |
| 215 | { |
| 216 | log_error("sound" , "Unable to init SDL audio: %s" , SDL_GetError()); |
| 217 | return -1; |
| 218 | } |
| 219 | |
| 220 | SDL_AudioSpec Format, FormatOut; |
| 221 | Format.freq = g_Config.m_SndRate; |
| 222 | Format.format = AUDIO_S16; |
| 223 | Format.channels = 2; |
| 224 | Format.samples = g_Config.m_SndBufferSize; |
| 225 | Format.callback = SdlCallback; |
| 226 | Format.userdata = this; |
| 227 | |
| 228 | // Open the audio device and start playing sound! |
| 229 | m_Device = SDL_OpenAudioDevice(device: nullptr, iscapture: 0, desired: &Format, obtained: &FormatOut, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE); |
| 230 | if(m_Device == 0) |
| 231 | { |
| 232 | log_error("sound" , "Unable to open audio device: %s" , SDL_GetError()); |
| 233 | return -1; |
| 234 | } |
| 235 | else |
| 236 | { |
| 237 | log_info("sound" , "Sound init successful using audio driver '%s'" , SDL_GetCurrentAudioDriver()); |
| 238 | } |
| 239 | |
| 240 | m_MixingRate = FormatOut.freq; |
| 241 | m_MaxFrames = FormatOut.samples * 2; |
| 242 | #if defined(CONF_VIDEORECORDER) |
| 243 | m_MaxFrames = maximum<uint32_t>(a: m_MaxFrames, b: 1024 * 2); // make the buffer bigger just in case |
| 244 | #endif |
| 245 | m_pMixBuffer = (int *)calloc(nmemb: m_MaxFrames * 2, size: sizeof(int)); |
| 246 | |
| 247 | m_SoundEnabled = true; |
| 248 | Update(); |
| 249 | |
| 250 | SDL_PauseAudioDevice(dev: m_Device, pause_on: 0); |
| 251 | return 0; |
| 252 | } |
| 253 | |
| 254 | int CSound::Update() |
| 255 | { |
| 256 | UpdateVolume(); |
| 257 | return 0; |
| 258 | } |
| 259 | |
| 260 | void CSound::UpdateVolume() |
| 261 | { |
| 262 | int WantedVolume = g_Config.m_SndVolume; |
| 263 | if(!m_pGraphics->WindowActive() && g_Config.m_SndNonactiveMute) |
| 264 | WantedVolume = 0; |
| 265 | m_SoundVolume.store(i: WantedVolume, m: std::memory_order_relaxed); |
| 266 | } |
| 267 | |
| 268 | void CSound::Shutdown() |
| 269 | { |
| 270 | StopAll(); |
| 271 | |
| 272 | // Stop sound callback before freeing sample data |
| 273 | SDL_CloseAudioDevice(dev: m_Device); |
| 274 | SDL_QuitSubSystem(SDL_INIT_AUDIO); |
| 275 | m_Device = 0; |
| 276 | |
| 277 | const CLockScope LockScope(m_SoundLock); |
| 278 | for(auto &Sample : m_aSamples) |
| 279 | { |
| 280 | free(ptr: Sample.m_pData); |
| 281 | Sample.m_pData = nullptr; |
| 282 | } |
| 283 | |
| 284 | free(ptr: m_pMixBuffer); |
| 285 | m_pMixBuffer = nullptr; |
| 286 | m_SoundEnabled = false; |
| 287 | } |
| 288 | |
| 289 | CSample *CSound::AllocSample() |
| 290 | { |
| 291 | const CLockScope LockScope(m_SoundLock); |
| 292 | if(m_FirstFreeSampleIndex == SAMPLE_INDEX_FULL) |
| 293 | return nullptr; |
| 294 | |
| 295 | CSample *pSample = &m_aSamples[m_FirstFreeSampleIndex]; |
| 296 | dbg_assert( |
| 297 | pSample->m_pData == nullptr && pSample->m_NextFreeSampleIndex != SAMPLE_INDEX_USED, |
| 298 | "Sample was not unloaded (index=%d, next=%d, duration=%f, data=%p)" , |
| 299 | pSample->m_Index, pSample->m_NextFreeSampleIndex, pSample->TotalTime(), pSample->m_pData); |
| 300 | m_FirstFreeSampleIndex = pSample->m_NextFreeSampleIndex; |
| 301 | pSample->m_NextFreeSampleIndex = SAMPLE_INDEX_USED; |
| 302 | return pSample; |
| 303 | } |
| 304 | |
| 305 | void CSound::RateConvert(CSample &Sample) const |
| 306 | { |
| 307 | dbg_assert(Sample.IsLoaded(), "Sample not loaded" ); |
| 308 | // make sure that we need to convert this sound |
| 309 | if(Sample.m_Rate == m_MixingRate) |
| 310 | return; |
| 311 | |
| 312 | // allocate new data |
| 313 | const int NumFrames = (int)((Sample.m_NumFrames / (float)Sample.m_Rate) * m_MixingRate); |
| 314 | short *pNewData = (short *)calloc(nmemb: (size_t)NumFrames * Sample.m_Channels, size: sizeof(short)); |
| 315 | |
| 316 | for(int i = 0; i < NumFrames; i++) |
| 317 | { |
| 318 | // resample TODO: this should be done better, like linear at least |
| 319 | float a = i / (float)NumFrames; |
| 320 | int f = (int)(a * Sample.m_NumFrames); |
| 321 | if(f >= Sample.m_NumFrames) |
| 322 | f = Sample.m_NumFrames - 1; |
| 323 | |
| 324 | // set new data |
| 325 | if(Sample.m_Channels == 1) |
| 326 | pNewData[i] = Sample.m_pData[f]; |
| 327 | else if(Sample.m_Channels == 2) |
| 328 | { |
| 329 | pNewData[i * 2] = Sample.m_pData[f * 2]; |
| 330 | pNewData[i * 2 + 1] = Sample.m_pData[f * 2 + 1]; |
| 331 | } |
| 332 | } |
| 333 | |
| 334 | // adjust looping position, note that this is not precise |
| 335 | const double Factor = (double)m_MixingRate / (double)Sample.m_Rate; |
| 336 | Sample.m_LoopStart = std::round(x: Sample.m_LoopStart * Factor); |
| 337 | |
| 338 | // free old data and apply new |
| 339 | free(ptr: Sample.m_pData); |
| 340 | Sample.m_pData = pNewData; |
| 341 | Sample.m_NumFrames = NumFrames; |
| 342 | Sample.m_Rate = m_MixingRate; |
| 343 | } |
| 344 | |
| 345 | bool CSound::DecodeOpus(CSample &Sample, const void *pData, unsigned DataSize, const char *pContextName) const |
| 346 | { |
| 347 | int OpusError = 0; |
| 348 | OggOpusFile *pOpusFile = op_open_memory(data: (const unsigned char *)pData, size: DataSize, error: &OpusError); |
| 349 | if(pOpusFile) |
| 350 | { |
| 351 | const int NumChannels = op_channel_count(of: pOpusFile, li: -1); |
| 352 | if(NumChannels > 2) |
| 353 | { |
| 354 | op_free(of: pOpusFile); |
| 355 | log_error("sound/opus" , "File is not mono or stereo. Filename='%s'" , pContextName); |
| 356 | return false; |
| 357 | } |
| 358 | |
| 359 | const int NumSamples = op_pcm_total(of: pOpusFile, li: -1); // per channel! |
| 360 | if(NumSamples < 0) |
| 361 | { |
| 362 | op_free(of: pOpusFile); |
| 363 | log_error("sound/opus" , "Failed to get number of samples, error %d. Filename='%s'" , NumSamples, pContextName); |
| 364 | return false; |
| 365 | } |
| 366 | |
| 367 | short *pSampleData = (short *)calloc(nmemb: (size_t)NumSamples * NumChannels, size: sizeof(short)); |
| 368 | |
| 369 | int Pos = 0; |
| 370 | while(Pos < NumSamples) |
| 371 | { |
| 372 | const int Read = op_read(of: pOpusFile, pcm: pSampleData + Pos * NumChannels, buf_size: (NumSamples - Pos) * NumChannels, li: nullptr); |
| 373 | if(Read < 0) |
| 374 | { |
| 375 | free(ptr: pSampleData); |
| 376 | op_free(of: pOpusFile); |
| 377 | log_error("sound/opus" , "op_read error %d at %d. Filename='%s'" , Read, Pos, pContextName); |
| 378 | return false; |
| 379 | } |
| 380 | else if(Read == 0) // EOF |
| 381 | break; |
| 382 | Pos += Read; |
| 383 | } |
| 384 | |
| 385 | Sample.m_pData = pSampleData; |
| 386 | Sample.m_NumFrames = Pos; |
| 387 | Sample.m_Rate = 48000; |
| 388 | Sample.m_Channels = NumChannels; |
| 389 | Sample.m_LoopStart = 0; |
| 390 | Sample.m_PausedAt = 0; |
| 391 | |
| 392 | const OpusTags *pTags = op_tags(of: pOpusFile, li: -1); |
| 393 | if(pTags) |
| 394 | { |
| 395 | for(int i = 0; i < pTags->comments; ++i) |
| 396 | { |
| 397 | const char * = pTags->user_comments[i]; |
| 398 | if(!pComment) |
| 399 | continue; |
| 400 | if(!str_startswith(str: pComment, prefix: "LOOP_START=" )) |
| 401 | continue; |
| 402 | int LoopStart = -1; |
| 403 | if(!str_toint(str: pComment + str_length(str: "LOOP_START=" ), out: &LoopStart)) |
| 404 | { |
| 405 | log_error("sound/opus" , "Invalid LOOP_START tag. Value='%s' Filename='%s'" , pComment + str_length("LOOP_START=" ), pContextName); |
| 406 | break; |
| 407 | } |
| 408 | if(LoopStart < 0 || LoopStart >= Sample.m_NumFrames) |
| 409 | { |
| 410 | log_error("sound/opus" , "Tag LOOP_START out of range. Value=%d Min=0 Max=%d Filename='%s'" , LoopStart, Sample.m_NumFrames - 1, pContextName); |
| 411 | break; |
| 412 | } |
| 413 | Sample.m_LoopStart = LoopStart; |
| 414 | break; |
| 415 | } |
| 416 | } |
| 417 | |
| 418 | op_free(of: pOpusFile); |
| 419 | } |
| 420 | else |
| 421 | { |
| 422 | log_error("sound/opus" , "Failed to decode sample, error %d. Filename='%s'" , OpusError, pContextName); |
| 423 | return false; |
| 424 | } |
| 425 | |
| 426 | return true; |
| 427 | } |
| 428 | |
| 429 | // TODO: Update WavPack to get rid of these global variables |
| 430 | static const void *s_pWVBuffer = nullptr; |
| 431 | static int s_WVBufferPosition = 0; |
| 432 | static int s_WVBufferSize = 0; |
| 433 | |
| 434 | static int ReadDataOld(void *pBuffer, int Size) |
| 435 | { |
| 436 | int ChunkSize = minimum(a: Size, b: s_WVBufferSize - s_WVBufferPosition); |
| 437 | mem_copy(dest: pBuffer, source: (const char *)s_pWVBuffer + s_WVBufferPosition, size: ChunkSize); |
| 438 | s_WVBufferPosition += ChunkSize; |
| 439 | return ChunkSize; |
| 440 | } |
| 441 | |
| 442 | #if defined(CONF_WAVPACK_OPEN_FILE_INPUT_EX) |
| 443 | static int ReadData(void *pId, void *pBuffer, int Size) |
| 444 | { |
| 445 | (void)pId; |
| 446 | return ReadDataOld(pBuffer, Size); |
| 447 | } |
| 448 | |
| 449 | static int ReturnFalse(void *pId) |
| 450 | { |
| 451 | (void)pId; |
| 452 | return 0; |
| 453 | } |
| 454 | |
| 455 | static unsigned int GetPos(void *pId) |
| 456 | { |
| 457 | (void)pId; |
| 458 | return s_WVBufferPosition; |
| 459 | } |
| 460 | |
| 461 | static unsigned int GetLength(void *pId) |
| 462 | { |
| 463 | (void)pId; |
| 464 | return s_WVBufferSize; |
| 465 | } |
| 466 | |
| 467 | static int PushBackByte(void *pId, int Char) |
| 468 | { |
| 469 | s_WVBufferPosition -= 1; |
| 470 | return 0; |
| 471 | } |
| 472 | #endif |
| 473 | |
| 474 | bool CSound::DecodeWV(CSample &Sample, const void *pData, unsigned DataSize, const char *pContextName) const |
| 475 | { |
| 476 | // no need to load sound when we are running with no sound |
| 477 | if(!m_SoundEnabled) |
| 478 | return false; |
| 479 | |
| 480 | dbg_assert(s_pWVBuffer == nullptr, "DecodeWV already in use" ); |
| 481 | s_pWVBuffer = pData; |
| 482 | s_WVBufferSize = DataSize; |
| 483 | s_WVBufferPosition = 0; |
| 484 | |
| 485 | char aError[100]; |
| 486 | |
| 487 | #if defined(CONF_WAVPACK_OPEN_FILE_INPUT_EX) |
| 488 | WavpackStreamReader Callback = {.read_bytes: 0}; |
| 489 | Callback.can_seek = ReturnFalse; |
| 490 | Callback.get_length = GetLength; |
| 491 | Callback.get_pos = GetPos; |
| 492 | Callback.push_back_byte = PushBackByte; |
| 493 | Callback.read_bytes = ReadData; |
| 494 | WavpackContext *pContext = WavpackOpenFileInputEx(reader: &Callback, wv_id: (void *)1, wvc_id: 0, error: aError, flags: 0, norm_offset: 0); |
| 495 | #else |
| 496 | WavpackContext *pContext = WavpackOpenFileInput(ReadDataOld, aError); |
| 497 | #endif |
| 498 | if(pContext) |
| 499 | { |
| 500 | const int NumSamples = WavpackGetNumSamples(wpc: pContext); |
| 501 | const int BitsPerSample = WavpackGetBitsPerSample(wpc: pContext); |
| 502 | const unsigned int SampleRate = WavpackGetSampleRate(wpc: pContext); |
| 503 | const int NumChannels = WavpackGetNumChannels(wpc: pContext); |
| 504 | |
| 505 | if(NumChannels > 2) |
| 506 | { |
| 507 | log_error("sound/wv" , "File is not mono or stereo. Filename='%s'" , pContextName); |
| 508 | s_pWVBuffer = nullptr; |
| 509 | return false; |
| 510 | } |
| 511 | |
| 512 | if(BitsPerSample != 16) |
| 513 | { |
| 514 | log_error("sound/wv" , "Bits per sample is %d, not 16. Filename='%s'" , BitsPerSample, pContextName); |
| 515 | s_pWVBuffer = nullptr; |
| 516 | return false; |
| 517 | } |
| 518 | |
| 519 | int *pBuffer = (int *)calloc(nmemb: (size_t)NumSamples * NumChannels, size: sizeof(int)); |
| 520 | if(!WavpackUnpackSamples(wpc: pContext, buffer: pBuffer, samples: NumSamples)) |
| 521 | { |
| 522 | free(ptr: pBuffer); |
| 523 | log_error("sound/wv" , "WavpackUnpackSamples failed. NumSamples=%d NumChannels=%d Filename='%s'" , NumSamples, NumChannels, pContextName); |
| 524 | s_pWVBuffer = nullptr; |
| 525 | return false; |
| 526 | } |
| 527 | |
| 528 | Sample.m_pData = (short *)calloc(nmemb: (size_t)NumSamples * NumChannels, size: sizeof(short)); |
| 529 | |
| 530 | int *pSrc = pBuffer; |
| 531 | short *pDst = Sample.m_pData; |
| 532 | for(int i = 0; i < NumSamples * NumChannels; i++) |
| 533 | *pDst++ = (short)*pSrc++; |
| 534 | |
| 535 | free(ptr: pBuffer); |
| 536 | #ifdef CONF_WAVPACK_CLOSE_FILE |
| 537 | WavpackCloseFile(wpc: pContext); |
| 538 | #endif |
| 539 | |
| 540 | Sample.m_NumFrames = NumSamples; |
| 541 | Sample.m_Rate = SampleRate; |
| 542 | Sample.m_Channels = NumChannels; |
| 543 | Sample.m_LoopStart = 0; |
| 544 | Sample.m_PausedAt = 0; |
| 545 | |
| 546 | s_pWVBuffer = nullptr; |
| 547 | } |
| 548 | else |
| 549 | { |
| 550 | log_error("sound/wv" , "Failed to decode sample (%s). Filename='%s'" , aError, pContextName); |
| 551 | s_pWVBuffer = nullptr; |
| 552 | return false; |
| 553 | } |
| 554 | |
| 555 | return true; |
| 556 | } |
| 557 | |
| 558 | int CSound::LoadOpus(const char *pFilename, int StorageType) |
| 559 | { |
| 560 | // no need to load sound when we are running with no sound |
| 561 | if(!m_SoundEnabled) |
| 562 | return -1; |
| 563 | |
| 564 | CSample *pSample = AllocSample(); |
| 565 | if(!pSample) |
| 566 | { |
| 567 | log_error("sound/opus" , "Failed to allocate sample ID. Filename='%s'" , pFilename); |
| 568 | return -1; |
| 569 | } |
| 570 | |
| 571 | void *pData; |
| 572 | unsigned DataSize; |
| 573 | if(!m_pStorage->ReadFile(pFilename, Type: StorageType, ppResult: &pData, pResultLen: &DataSize)) |
| 574 | { |
| 575 | UnloadSample(SampleId: pSample->m_Index); |
| 576 | log_error("sound/opus" , "Failed to open file. Filename='%s'" , pFilename); |
| 577 | return -1; |
| 578 | } |
| 579 | |
| 580 | const bool DecodeSuccess = DecodeOpus(Sample&: *pSample, pData, DataSize, pContextName: pFilename); |
| 581 | free(ptr: pData); |
| 582 | if(!DecodeSuccess) |
| 583 | { |
| 584 | UnloadSample(SampleId: pSample->m_Index); |
| 585 | return -1; |
| 586 | } |
| 587 | |
| 588 | if(g_Config.m_Debug) |
| 589 | log_trace("sound/opus" , "Loaded '%s' (index %d)" , pFilename, pSample->m_Index); |
| 590 | |
| 591 | RateConvert(Sample&: *pSample); |
| 592 | return pSample->m_Index; |
| 593 | } |
| 594 | |
| 595 | int CSound::LoadWV(const char *pFilename, int StorageType) |
| 596 | { |
| 597 | // no need to load sound when we are running with no sound |
| 598 | if(!m_SoundEnabled) |
| 599 | return -1; |
| 600 | |
| 601 | CSample *pSample = AllocSample(); |
| 602 | if(!pSample) |
| 603 | { |
| 604 | log_error("sound/wv" , "Failed to allocate sample ID. Filename='%s'" , pFilename); |
| 605 | return -1; |
| 606 | } |
| 607 | |
| 608 | void *pData; |
| 609 | unsigned DataSize; |
| 610 | if(!m_pStorage->ReadFile(pFilename, Type: StorageType, ppResult: &pData, pResultLen: &DataSize)) |
| 611 | { |
| 612 | UnloadSample(SampleId: pSample->m_Index); |
| 613 | log_error("sound/wv" , "Failed to open file. Filename='%s'" , pFilename); |
| 614 | return -1; |
| 615 | } |
| 616 | |
| 617 | const bool DecodeSuccess = DecodeWV(Sample&: *pSample, pData, DataSize, pContextName: pFilename); |
| 618 | free(ptr: pData); |
| 619 | if(!DecodeSuccess) |
| 620 | { |
| 621 | UnloadSample(SampleId: pSample->m_Index); |
| 622 | return -1; |
| 623 | } |
| 624 | |
| 625 | if(g_Config.m_Debug) |
| 626 | log_trace("sound/wv" , "Loaded '%s' (index %d)" , pFilename, pSample->m_Index); |
| 627 | |
| 628 | RateConvert(Sample&: *pSample); |
| 629 | return pSample->m_Index; |
| 630 | } |
| 631 | |
| 632 | int CSound::LoadOpusFromMem(const void *pData, unsigned DataSize, bool ForceLoad, const char *pContextName) |
| 633 | { |
| 634 | // no need to load sound when we are running with no sound |
| 635 | if(!m_SoundEnabled && !ForceLoad) |
| 636 | return -1; |
| 637 | |
| 638 | CSample *pSample = AllocSample(); |
| 639 | if(!pSample) |
| 640 | return -1; |
| 641 | |
| 642 | if(!DecodeOpus(Sample&: *pSample, pData, DataSize, pContextName)) |
| 643 | { |
| 644 | UnloadSample(SampleId: pSample->m_Index); |
| 645 | return -1; |
| 646 | } |
| 647 | |
| 648 | RateConvert(Sample&: *pSample); |
| 649 | return pSample->m_Index; |
| 650 | } |
| 651 | |
| 652 | int CSound::LoadWVFromMem(const void *pData, unsigned DataSize, bool ForceLoad, const char *pContextName) |
| 653 | { |
| 654 | // no need to load sound when we are running with no sound |
| 655 | if(!m_SoundEnabled && !ForceLoad) |
| 656 | return -1; |
| 657 | |
| 658 | CSample *pSample = AllocSample(); |
| 659 | if(!pSample) |
| 660 | return -1; |
| 661 | |
| 662 | if(!DecodeWV(Sample&: *pSample, pData, DataSize, pContextName)) |
| 663 | { |
| 664 | UnloadSample(SampleId: pSample->m_Index); |
| 665 | return -1; |
| 666 | } |
| 667 | |
| 668 | RateConvert(Sample&: *pSample); |
| 669 | return pSample->m_Index; |
| 670 | } |
| 671 | |
| 672 | void CSound::UnloadSample(int SampleId) |
| 673 | { |
| 674 | if(SampleId == -1) |
| 675 | return; |
| 676 | |
| 677 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 678 | const CLockScope LockScope(m_SoundLock); |
| 679 | CSample &Sample = m_aSamples[SampleId]; |
| 680 | |
| 681 | if(Sample.IsLoaded()) |
| 682 | { |
| 683 | // Stop voices using this sample |
| 684 | for(auto &Voice : m_aVoices) |
| 685 | { |
| 686 | if(Voice.m_pSample == &Sample) |
| 687 | { |
| 688 | Voice.m_pSample = nullptr; |
| 689 | } |
| 690 | } |
| 691 | |
| 692 | // Free data |
| 693 | free(ptr: Sample.m_pData); |
| 694 | Sample.m_pData = nullptr; |
| 695 | } |
| 696 | |
| 697 | // Free slot |
| 698 | if(Sample.m_NextFreeSampleIndex == SAMPLE_INDEX_USED) |
| 699 | { |
| 700 | Sample.m_NextFreeSampleIndex = m_FirstFreeSampleIndex; |
| 701 | m_FirstFreeSampleIndex = Sample.m_Index; |
| 702 | } |
| 703 | } |
| 704 | |
| 705 | float CSound::GetSampleTotalTime(int SampleId) |
| 706 | { |
| 707 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 708 | |
| 709 | const CLockScope LockScope(m_SoundLock); |
| 710 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 711 | return m_aSamples[SampleId].TotalTime(); |
| 712 | } |
| 713 | |
| 714 | float CSound::GetSampleCurrentTime(int SampleId) |
| 715 | { |
| 716 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 717 | |
| 718 | const CLockScope LockScope(m_SoundLock); |
| 719 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 720 | CSample *pSample = &m_aSamples[SampleId]; |
| 721 | for(auto &Voice : m_aVoices) |
| 722 | { |
| 723 | if(Voice.m_pSample == pSample) |
| 724 | { |
| 725 | return Voice.m_Tick / (float)pSample->m_Rate; |
| 726 | } |
| 727 | } |
| 728 | |
| 729 | return pSample->m_PausedAt / (float)pSample->m_Rate; |
| 730 | } |
| 731 | |
| 732 | void CSound::SetSampleCurrentTime(int SampleId, float Time) |
| 733 | { |
| 734 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 735 | |
| 736 | const CLockScope LockScope(m_SoundLock); |
| 737 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 738 | CSample *pSample = &m_aSamples[SampleId]; |
| 739 | for(auto &Voice : m_aVoices) |
| 740 | { |
| 741 | if(Voice.m_pSample == pSample) |
| 742 | { |
| 743 | Voice.m_Tick = pSample->m_NumFrames * Time; |
| 744 | return; |
| 745 | } |
| 746 | } |
| 747 | |
| 748 | pSample->m_PausedAt = pSample->m_NumFrames * Time; |
| 749 | } |
| 750 | |
| 751 | void CSound::SetChannel(int ChannelId, float Vol, float Pan) |
| 752 | { |
| 753 | dbg_assert(ChannelId >= 0 && ChannelId < NUM_CHANNELS, "ChannelId invalid" ); |
| 754 | |
| 755 | const CLockScope LockScope(m_SoundLock); |
| 756 | m_aChannels[ChannelId].m_Vol = (int)(Vol * 255.0f); |
| 757 | m_aChannels[ChannelId].m_Pan = (int)(Pan * 255.0f); // TODO: this is only on and off right now |
| 758 | } |
| 759 | |
| 760 | void CSound::SetListenerPosition(vec2 Position) |
| 761 | { |
| 762 | m_ListenerPositionX.store(t: Position.x, m: std::memory_order_relaxed); |
| 763 | m_ListenerPositionY.store(t: Position.y, m: std::memory_order_relaxed); |
| 764 | } |
| 765 | |
| 766 | void CSound::SetVoiceVolume(CVoiceHandle Voice, float Volume) |
| 767 | { |
| 768 | if(!Voice.IsValid()) |
| 769 | return; |
| 770 | |
| 771 | int VoiceId = Voice.Id(); |
| 772 | |
| 773 | const CLockScope LockScope(m_SoundLock); |
| 774 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 775 | return; |
| 776 | |
| 777 | Volume = std::clamp(val: Volume, lo: 0.0f, hi: 1.0f); |
| 778 | m_aVoices[VoiceId].m_Vol = (int)(Volume * 255.0f); |
| 779 | } |
| 780 | |
| 781 | void CSound::SetVoiceFalloff(CVoiceHandle Voice, float Falloff) |
| 782 | { |
| 783 | if(!Voice.IsValid()) |
| 784 | return; |
| 785 | |
| 786 | int VoiceId = Voice.Id(); |
| 787 | |
| 788 | const CLockScope LockScope(m_SoundLock); |
| 789 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 790 | return; |
| 791 | |
| 792 | Falloff = std::clamp(val: Falloff, lo: 0.0f, hi: 1.0f); |
| 793 | m_aVoices[VoiceId].m_Falloff = Falloff; |
| 794 | } |
| 795 | |
| 796 | void CSound::SetVoicePosition(CVoiceHandle Voice, vec2 Position) |
| 797 | { |
| 798 | if(!Voice.IsValid()) |
| 799 | return; |
| 800 | |
| 801 | int VoiceId = Voice.Id(); |
| 802 | |
| 803 | const CLockScope LockScope(m_SoundLock); |
| 804 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 805 | return; |
| 806 | |
| 807 | m_aVoices[VoiceId].m_Position = Position; |
| 808 | } |
| 809 | |
| 810 | void CSound::SetVoiceTimeOffset(CVoiceHandle Voice, float TimeOffset) |
| 811 | { |
| 812 | if(!Voice.IsValid()) |
| 813 | return; |
| 814 | |
| 815 | int VoiceId = Voice.Id(); |
| 816 | |
| 817 | const CLockScope LockScope(m_SoundLock); |
| 818 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 819 | return; |
| 820 | |
| 821 | if(!m_aVoices[VoiceId].m_pSample) |
| 822 | return; |
| 823 | |
| 824 | int Tick = 0; |
| 825 | bool IsLooping = m_aVoices[VoiceId].m_Flags & ISound::FLAG_LOOP; |
| 826 | uint64_t TickOffset = m_aVoices[VoiceId].m_pSample->m_Rate * TimeOffset; |
| 827 | if(m_aVoices[VoiceId].m_pSample->m_NumFrames > 0 && IsLooping) |
| 828 | { |
| 829 | const int LoopStart = m_aVoices[VoiceId].m_pSample->m_LoopStart; |
| 830 | const int NumFrames = m_aVoices[VoiceId].m_pSample->m_NumFrames; |
| 831 | if(TickOffset < static_cast<uint64_t>(NumFrames)) |
| 832 | { |
| 833 | // Still in first playthrough |
| 834 | Tick = TickOffset; |
| 835 | } |
| 836 | else |
| 837 | { |
| 838 | // Past first playthrough, wrap within loop section only |
| 839 | const int LoopLength = NumFrames - LoopStart; |
| 840 | if(LoopLength > 0) |
| 841 | Tick = LoopStart + ((TickOffset - NumFrames) % LoopLength); |
| 842 | else |
| 843 | Tick = LoopStart; |
| 844 | } |
| 845 | } |
| 846 | else |
| 847 | { |
| 848 | Tick = std::clamp<uint64_t>(val: TickOffset, lo: 0, hi: m_aVoices[VoiceId].m_pSample->m_NumFrames); |
| 849 | } |
| 850 | |
| 851 | // at least 200msec off, else depend on buffer size |
| 852 | float Threshold = maximum(a: 0.2f * m_aVoices[VoiceId].m_pSample->m_Rate, b: (float)m_MaxFrames); |
| 853 | if(absolute(a: m_aVoices[VoiceId].m_Tick - Tick) > Threshold) |
| 854 | { |
| 855 | // take care of looping (modulo!) |
| 856 | if(!(IsLooping && (minimum(a: m_aVoices[VoiceId].m_Tick, b: Tick) + m_aVoices[VoiceId].m_pSample->m_NumFrames - maximum(a: m_aVoices[VoiceId].m_Tick, b: Tick)) <= Threshold)) |
| 857 | { |
| 858 | m_aVoices[VoiceId].m_Tick = Tick; |
| 859 | } |
| 860 | } |
| 861 | } |
| 862 | |
| 863 | void CSound::SetVoiceCircle(CVoiceHandle Voice, float Radius) |
| 864 | { |
| 865 | if(!Voice.IsValid()) |
| 866 | return; |
| 867 | |
| 868 | int VoiceId = Voice.Id(); |
| 869 | |
| 870 | const CLockScope LockScope(m_SoundLock); |
| 871 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 872 | return; |
| 873 | |
| 874 | m_aVoices[VoiceId].m_Shape = ISound::SHAPE_CIRCLE; |
| 875 | m_aVoices[VoiceId].m_Circle.m_Radius = maximum(a: 0.0f, b: Radius); |
| 876 | } |
| 877 | |
| 878 | void CSound::SetVoiceRectangle(CVoiceHandle Voice, float Width, float Height) |
| 879 | { |
| 880 | if(!Voice.IsValid()) |
| 881 | return; |
| 882 | |
| 883 | int VoiceId = Voice.Id(); |
| 884 | |
| 885 | const CLockScope LockScope(m_SoundLock); |
| 886 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 887 | return; |
| 888 | |
| 889 | m_aVoices[VoiceId].m_Shape = ISound::SHAPE_RECTANGLE; |
| 890 | m_aVoices[VoiceId].m_Rectangle.m_Width = maximum(a: 0.0f, b: Width); |
| 891 | m_aVoices[VoiceId].m_Rectangle.m_Height = maximum(a: 0.0f, b: Height); |
| 892 | } |
| 893 | |
| 894 | ISound::CVoiceHandle CSound::Play(int ChannelId, int SampleId, int Flags, float Volume, vec2 Position) |
| 895 | { |
| 896 | const CLockScope LockScope(m_SoundLock); |
| 897 | |
| 898 | // search for voice |
| 899 | int VoiceId = -1; |
| 900 | for(int i = 0; i < NUM_VOICES; i++) |
| 901 | { |
| 902 | int NextId = (m_NextVoice + i) % NUM_VOICES; |
| 903 | if(!m_aVoices[NextId].m_pSample) |
| 904 | { |
| 905 | VoiceId = NextId; |
| 906 | m_NextVoice = NextId + 1; |
| 907 | break; |
| 908 | } |
| 909 | } |
| 910 | if(VoiceId == -1) |
| 911 | { |
| 912 | return CreateVoiceHandle(Index: -1, Age: -1); |
| 913 | } |
| 914 | |
| 915 | // voice found, use it |
| 916 | m_aVoices[VoiceId].m_pSample = &m_aSamples[SampleId]; |
| 917 | m_aVoices[VoiceId].m_pChannel = &m_aChannels[ChannelId]; |
| 918 | if(Flags & FLAG_LOOP) |
| 919 | { |
| 920 | m_aVoices[VoiceId].m_Tick = m_aSamples[SampleId].m_PausedAt; |
| 921 | } |
| 922 | else if(Flags & FLAG_PREVIEW) |
| 923 | { |
| 924 | m_aVoices[VoiceId].m_Tick = m_aSamples[SampleId].m_PausedAt; |
| 925 | m_aSamples[SampleId].m_PausedAt = 0; |
| 926 | } |
| 927 | else |
| 928 | { |
| 929 | m_aVoices[VoiceId].m_Tick = 0; |
| 930 | } |
| 931 | m_aVoices[VoiceId].m_Vol = (int)(std::clamp(val: Volume, lo: 0.0f, hi: 1.0f) * 255.0f); |
| 932 | m_aVoices[VoiceId].m_Flags = Flags; |
| 933 | m_aVoices[VoiceId].m_Position = Position; |
| 934 | m_aVoices[VoiceId].m_Falloff = 0.0f; |
| 935 | m_aVoices[VoiceId].m_Shape = ISound::SHAPE_CIRCLE; |
| 936 | m_aVoices[VoiceId].m_Circle.m_Radius = 1500; |
| 937 | return CreateVoiceHandle(Index: VoiceId, Age: m_aVoices[VoiceId].m_Age); |
| 938 | } |
| 939 | |
| 940 | ISound::CVoiceHandle CSound::PlayAt(int ChannelId, int SampleId, int Flags, float Volume, vec2 Position) |
| 941 | { |
| 942 | return Play(ChannelId, SampleId, Flags: Flags | ISound::FLAG_POS, Volume, Position); |
| 943 | } |
| 944 | |
| 945 | ISound::CVoiceHandle CSound::Play(int ChannelId, int SampleId, int Flags, float Volume) |
| 946 | { |
| 947 | return Play(ChannelId, SampleId, Flags, Volume, Position: vec2(0.0f, 0.0f)); |
| 948 | } |
| 949 | |
| 950 | void CSound::Pause(int SampleId) |
| 951 | { |
| 952 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 953 | |
| 954 | // TODO: a nice fade out |
| 955 | const CLockScope LockScope(m_SoundLock); |
| 956 | CSample *pSample = &m_aSamples[SampleId]; |
| 957 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 958 | for(auto &Voice : m_aVoices) |
| 959 | { |
| 960 | if(Voice.m_pSample == pSample) |
| 961 | { |
| 962 | Voice.m_pSample->m_PausedAt = Voice.m_Tick; |
| 963 | Voice.m_pSample = nullptr; |
| 964 | } |
| 965 | } |
| 966 | } |
| 967 | |
| 968 | void CSound::Stop(int SampleId) |
| 969 | { |
| 970 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 971 | |
| 972 | // TODO: a nice fade out |
| 973 | const CLockScope LockScope(m_SoundLock); |
| 974 | CSample *pSample = &m_aSamples[SampleId]; |
| 975 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 976 | for(auto &Voice : m_aVoices) |
| 977 | { |
| 978 | if(Voice.m_pSample == pSample) |
| 979 | { |
| 980 | if(Voice.m_Flags & FLAG_LOOP) |
| 981 | Voice.m_pSample->m_PausedAt = Voice.m_Tick; |
| 982 | else |
| 983 | Voice.m_pSample->m_PausedAt = 0; |
| 984 | Voice.m_pSample = nullptr; |
| 985 | } |
| 986 | } |
| 987 | } |
| 988 | |
| 989 | void CSound::StopAll() |
| 990 | { |
| 991 | // TODO: a nice fade out |
| 992 | const CLockScope LockScope(m_SoundLock); |
| 993 | for(auto &Voice : m_aVoices) |
| 994 | { |
| 995 | if(Voice.m_pSample) |
| 996 | { |
| 997 | if(Voice.m_Flags & FLAG_LOOP) |
| 998 | Voice.m_pSample->m_PausedAt = Voice.m_Tick; |
| 999 | else |
| 1000 | Voice.m_pSample->m_PausedAt = 0; |
| 1001 | } |
| 1002 | Voice.m_pSample = nullptr; |
| 1003 | } |
| 1004 | } |
| 1005 | |
| 1006 | void CSound::StopVoice(CVoiceHandle Voice) |
| 1007 | { |
| 1008 | if(!Voice.IsValid()) |
| 1009 | return; |
| 1010 | |
| 1011 | int VoiceId = Voice.Id(); |
| 1012 | |
| 1013 | const CLockScope LockScope(m_SoundLock); |
| 1014 | if(m_aVoices[VoiceId].m_Age != Voice.Age()) |
| 1015 | return; |
| 1016 | |
| 1017 | m_aVoices[VoiceId].m_pSample = nullptr; |
| 1018 | m_aVoices[VoiceId].m_Age++; |
| 1019 | } |
| 1020 | |
| 1021 | bool CSound::IsPlaying(int SampleId) |
| 1022 | { |
| 1023 | dbg_assert(SampleId >= 0 && SampleId < NUM_SAMPLES, "SampleId invalid" ); |
| 1024 | const CLockScope LockScope(m_SoundLock); |
| 1025 | const CSample *pSample = &m_aSamples[SampleId]; |
| 1026 | dbg_assert(m_aSamples[SampleId].IsLoaded(), "Sample not loaded" ); |
| 1027 | return std::any_of(first: std::begin(arr&: m_aVoices), last: std::end(arr&: m_aVoices), pred: [pSample](const auto &Voice) { return Voice.m_pSample == pSample; }); |
| 1028 | } |
| 1029 | |
| 1030 | void CSound::PauseAudioDevice() |
| 1031 | { |
| 1032 | SDL_PauseAudioDevice(dev: m_Device, pause_on: 1); |
| 1033 | } |
| 1034 | |
| 1035 | void CSound::UnpauseAudioDevice() |
| 1036 | { |
| 1037 | SDL_PauseAudioDevice(dev: m_Device, pause_on: 0); |
| 1038 | } |
| 1039 | |
| 1040 | IEngineSound *CreateEngineSound() { return new CSound; } |
| 1041 | |