| 1 | /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * |
| 2 | * Mupen64plus-sdl-audio - main.c * |
| 3 | * Mupen64Plus homepage: http://code.google.com/p/mupen64plus/ * |
| 4 | * Copyright (C) 2007-2009 Richard Goedeken * |
| 5 | * Copyright (C) 2007-2008 Ebenblues * |
| 6 | * Copyright (C) 2003 JttL * |
| 7 | * Copyright (C) 2002 Hacktarux * |
| 8 | * * |
| 9 | * This program is free software; you can redistribute it and/or modify * |
| 10 | * it under the terms of the GNU General Public License as published by * |
| 11 | * the Free Software Foundation; either version 2 of the License, or * |
| 12 | * (at your option) any later version. * |
| 13 | * * |
| 14 | * This program is distributed in the hope that it will be useful, * |
| 15 | * but WITHOUT ANY WARRANTY; without even the implied warranty of * |
| 16 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * |
| 17 | * GNU General Public License for more details. * |
| 18 | * * |
| 19 | * You should have received a copy of the GNU General Public License * |
| 20 | * along with this program; if not, write to the * |
| 21 | * Free Software Foundation, Inc., * |
| 22 | * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. * |
| 23 | * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ |
| 24 | |
| 25 | #include <stdio.h> |
| 26 | #include <stdlib.h> |
| 27 | #include <string.h> |
| 28 | |
| 29 | #include <SDL.h> |
| 30 | #include <SDL_audio.h> |
| 31 | |
| 32 | #ifdef USE_SRC |
| 33 | #include <samplerate.h> |
| 34 | #endif |
| 35 | #ifdef USE_SPEEX |
| 36 | #include <speex/speex_resampler.h> |
| 37 | #endif |
| 38 | |
| 39 | #define M64P_PLUGIN_PROTOTYPES 1 |
| 40 | #include "m64p_types.h" |
| 41 | #include "m64p_plugin.h" |
| 42 | #include "m64p_common.h" |
| 43 | #include "m64p_config.h" |
| 44 | |
| 45 | #include "main.h" |
| 46 | #include "volume.h" |
| 47 | #include "osal_dynamiclib.h" |
| 48 | |
| 49 | /* Default start-time size of primary buffer (in equivalent output samples). |
| 50 | This is the buffer where audio is loaded after it's extracted from n64's memory. |
| 51 | This value must be larger than PRIMARY_BUFFER_TARGET */ |
| 52 | #define PRIMARY_BUFFER_SIZE 16384 |
| 53 | |
| 54 | /* this is the buffer fullness level (in equivalent output samples) which is targeted |
| 55 | for the primary audio buffer (by inserting delays) each time data is received from |
| 56 | the running N64 program. This value must be larger than the SECONDARY_BUFFER_SIZE. |
| 57 | Decreasing this value will reduce audio latency but requires a faster PC to avoid |
| 58 | choppiness. Increasing this will increase audio latency but reduce the chance of |
| 59 | drop-outs. The default value 10240 gives a 232ms maximum A/V delay at 44.1khz */ |
| 60 | #define PRIMARY_BUFFER_TARGET 10240 |
| 61 | |
| 62 | /* Size of secondary buffer, in output samples. This is the requested size of SDL's |
| 63 | hardware buffer, and the size of the mix buffer for doing SDL volume control. The |
| 64 | SDL documentation states that this should be a power of two between 512 and 8192. */ |
| 65 | #define SECONDARY_BUFFER_SIZE 8192 |
| 66 | /*SEB 2048 before*/ |
| 67 | |
| 68 | /* This sets default frequency what is used if rom doesn't want to change it. |
| 69 | Probably only game that needs this is Zelda: Ocarina Of Time Master Quest |
| 70 | *NOTICE* We should try to find out why Demos' frequencies are always wrong |
| 71 | They tend to rely on a default frequency, apparently, never the same one ;)*/ |
| 72 | #define DEFAULT_FREQUENCY 33600 |
| 73 | |
| 74 | /* number of bytes per sample */ |
| 75 | #define N64_SAMPLE_BYTES 4 |
| 76 | #define SDL_SAMPLE_BYTES 4 |
| 77 | |
| 78 | /* volume mixer types */ |
| 79 | #define VOLUME_TYPE_SDL 1 |
| 80 | #define VOLUME_TYPE_OSS 2 |
| 81 | |
| 82 | /* local variables */ |
| 83 | static void (*l_DebugCallback)(void *, int, const char *) = NULL; |
| 84 | static void *l_DebugCallContext = NULL; |
| 85 | static int l_PluginInit = 0; |
| 86 | static int l_PausedForSync = 1; /* Audio is started in paused state after SDL initialization */ |
| 87 | static m64p_handle l_ConfigAudio; |
| 88 | |
| 89 | enum resampler_type { |
| 90 | RESAMPLER_TRIVIAL, |
| 91 | #ifdef USE_SRC |
| 92 | RESAMPLER_SRC, |
| 93 | #endif |
| 94 | #ifdef USE_SPEEX |
| 95 | RESAMPLER_SPEEX, |
| 96 | #endif |
| 97 | }; |
| 98 | |
| 99 | /* Read header for type definition */ |
| 100 | static AUDIO_INFO AudioInfo; |
| 101 | /* The hardware specifications we are using */ |
| 102 | static SDL_AudioSpec *hardware_spec; |
| 103 | /* Pointer to the primary audio buffer */ |
| 104 | static unsigned char *primaryBuffer = NULL; |
| 105 | static unsigned int primaryBufferBytes = 0; |
| 106 | /* Pointer to the mixing buffer for voume control*/ |
| 107 | static unsigned char *mixBuffer = NULL; |
| 108 | /* Position in buffer array where next audio chunk should be placed */ |
| 109 | static unsigned int buffer_pos = 0; |
| 110 | /* Audio frequency, this is usually obtained from the game, but for compatibility we set default value */ |
| 111 | static int GameFreq = DEFAULT_FREQUENCY; |
| 112 | /* timestamp for the last time that our audio callback was called */ |
| 113 | static unsigned int last_callback_ticks = 0; |
| 114 | /* SpeedFactor is used to increase/decrease game playback speed */ |
| 115 | static unsigned int speed_factor = 100; |
| 116 | // If this is true then left and right channels are swapped */ |
| 117 | static int SwapChannels = 0; |
| 118 | // Size of Primary audio buffer in equivalent output samples |
| 119 | static unsigned int PrimaryBufferSize = PRIMARY_BUFFER_SIZE; |
| 120 | // Fullness level target for Primary audio buffer, in equivalent output samples |
| 121 | static unsigned int PrimaryBufferTarget = PRIMARY_BUFFER_TARGET; |
| 122 | // Size of Secondary audio buffer in output samples |
| 123 | static unsigned int SecondaryBufferSize = SECONDARY_BUFFER_SIZE; |
| 124 | // Resample type |
| 125 | static enum resampler_type Resample = RESAMPLER_TRIVIAL; |
| 126 | // Resampler specific quality |
| 127 | static int ResampleQuality = 3; |
| 128 | // volume to scale the audio by, range of 0..100 |
| 129 | // if muted, this holds the volume when not muted |
| 130 | static int VolPercent = 80; |
| 131 | // how much percent to increment/decrement volume by |
| 132 | static int VolDelta = 5; |
| 133 | // the actual volume passed into SDL, range of 0..SDL_MIX_MAXVOLUME |
| 134 | static int VolSDL = SDL_MIX_MAXVOLUME; |
| 135 | // Muted or not |
| 136 | static int VolIsMuted = 0; |
| 137 | //which type of volume control to use |
| 138 | static int VolumeControlType = VOLUME_TYPE_SDL; |
| 139 | |
| 140 | static int OutputFreq; |
| 141 | |
| 142 | // Prototype of local functions |
| 143 | static void my_audio_callback(void *userdata, unsigned char *stream, int len); |
| 144 | static void InitializeAudio(int freq); |
| 145 | static void ReadConfig(void); |
| 146 | static void InitializeSDL(void); |
| 147 | |
| 148 | static int critical_failure = 0; |
| 149 | |
| 150 | /* definitions of pointers to Core config functions */ |
| 151 | ptr_ConfigOpenSection ConfigOpenSection = NULL; |
| 152 | ptr_ConfigDeleteSection ConfigDeleteSection = NULL; |
| 153 | ptr_ConfigSaveSection ConfigSaveSection = NULL; |
| 154 | ptr_ConfigSetParameter ConfigSetParameter = NULL; |
| 155 | ptr_ConfigGetParameter ConfigGetParameter = NULL; |
| 156 | ptr_ConfigGetParameterHelp ConfigGetParameterHelp = NULL; |
| 157 | ptr_ConfigSetDefaultInt ConfigSetDefaultInt = NULL; |
| 158 | ptr_ConfigSetDefaultFloat ConfigSetDefaultFloat = NULL; |
| 159 | ptr_ConfigSetDefaultBool ConfigSetDefaultBool = NULL; |
| 160 | ptr_ConfigSetDefaultString ConfigSetDefaultString = NULL; |
| 161 | ptr_ConfigGetParamInt ConfigGetParamInt = NULL; |
| 162 | ptr_ConfigGetParamFloat ConfigGetParamFloat = NULL; |
| 163 | ptr_ConfigGetParamBool ConfigGetParamBool = NULL; |
| 164 | ptr_ConfigGetParamString ConfigGetParamString = NULL; |
| 165 | |
| 166 | /* Global functions */ |
| 167 | static void DebugMessage(int level, const char *message, ...) |
| 168 | { |
| 169 | char msgbuf[1024]; |
| 170 | va_list args; |
| 171 | |
| 172 | if (l_DebugCallback == NULL) |
| 173 | return; |
| 174 | |
| 175 | va_start(args, message); |
| 176 | vsprintf(msgbuf, message, args); |
| 177 | |
| 178 | (*l_DebugCallback)(l_DebugCallContext, level, msgbuf); |
| 179 | |
| 180 | va_end(args); |
| 181 | } |
| 182 | |
| 183 | /* Mupen64Plus plugin functions */ |
| 184 | EXPORT m64p_error CALL PluginStartup(m64p_dynlib_handle CoreLibHandle, void *Context, |
| 185 | void (*DebugCallback)(void *, int, const char *)) |
| 186 | { |
| 187 | ptr_CoreGetAPIVersions CoreAPIVersionFunc; |
| 188 | |
| 189 | int ConfigAPIVersion, DebugAPIVersion, VidextAPIVersion, bSaveConfig; |
| 190 | float fConfigParamsVersion = 0.0f; |
| 191 | |
| 192 | if (l_PluginInit) |
| 193 | return M64ERR_ALREADY_INIT; |
| 194 | |
| 195 | /* first thing is to set the callback function for debug info */ |
| 196 | l_DebugCallback = DebugCallback; |
| 197 | l_DebugCallContext = Context; |
| 198 | |
| 199 | /* attach and call the CoreGetAPIVersions function, check Config API version for compatibility */ |
| 200 | CoreAPIVersionFunc = (ptr_CoreGetAPIVersions) osal_dynlib_getproc(CoreLibHandle, "CoreGetAPIVersions"); |
| 201 | if (CoreAPIVersionFunc == NULL) |
| 202 | { |
| 203 | DebugMessage(M64MSG_ERROR, "Core emulator broken; no CoreAPIVersionFunc() function found."); |
| 204 | return M64ERR_INCOMPATIBLE; |
| 205 | } |
| 206 | |
| 207 | (*CoreAPIVersionFunc)(&ConfigAPIVersion, &DebugAPIVersion, &VidextAPIVersion, NULL); |
| 208 | if ((ConfigAPIVersion & 0xffff0000) != (CONFIG_API_VERSION & 0xffff0000)) |
| 209 | { |
| 210 | DebugMessage(M64MSG_ERROR, "Emulator core Config API (v%i.%i.%i) incompatible with plugin (v%i.%i.%i)", |
| 211 | VERSION_PRINTF_SPLIT(ConfigAPIVersion), VERSION_PRINTF_SPLIT(CONFIG_API_VERSION)); |
| 212 | return M64ERR_INCOMPATIBLE; |
| 213 | } |
| 214 | |
| 215 | /* Get the core config function pointers from the library handle */ |
| 216 | ConfigOpenSection = (ptr_ConfigOpenSection) osal_dynlib_getproc(CoreLibHandle, "ConfigOpenSection"); |
| 217 | ConfigDeleteSection = (ptr_ConfigDeleteSection) osal_dynlib_getproc(CoreLibHandle, "ConfigDeleteSection"); |
| 218 | ConfigSaveSection = (ptr_ConfigSaveSection) osal_dynlib_getproc(CoreLibHandle, "ConfigSaveSection"); |
| 219 | ConfigSetParameter = (ptr_ConfigSetParameter) osal_dynlib_getproc(CoreLibHandle, "ConfigSetParameter"); |
| 220 | ConfigGetParameter = (ptr_ConfigGetParameter) osal_dynlib_getproc(CoreLibHandle, "ConfigGetParameter"); |
| 221 | ConfigSetDefaultInt = (ptr_ConfigSetDefaultInt) osal_dynlib_getproc(CoreLibHandle, "ConfigSetDefaultInt"); |
| 222 | ConfigSetDefaultFloat = (ptr_ConfigSetDefaultFloat) osal_dynlib_getproc(CoreLibHandle, "ConfigSetDefaultFloat"); |
| 223 | ConfigSetDefaultBool = (ptr_ConfigSetDefaultBool) osal_dynlib_getproc(CoreLibHandle, "ConfigSetDefaultBool"); |
| 224 | ConfigSetDefaultString = (ptr_ConfigSetDefaultString) osal_dynlib_getproc(CoreLibHandle, "ConfigSetDefaultString"); |
| 225 | ConfigGetParamInt = (ptr_ConfigGetParamInt) osal_dynlib_getproc(CoreLibHandle, "ConfigGetParamInt"); |
| 226 | ConfigGetParamFloat = (ptr_ConfigGetParamFloat) osal_dynlib_getproc(CoreLibHandle, "ConfigGetParamFloat"); |
| 227 | ConfigGetParamBool = (ptr_ConfigGetParamBool) osal_dynlib_getproc(CoreLibHandle, "ConfigGetParamBool"); |
| 228 | ConfigGetParamString = (ptr_ConfigGetParamString) osal_dynlib_getproc(CoreLibHandle, "ConfigGetParamString"); |
| 229 | |
| 230 | if (!ConfigOpenSection || !ConfigDeleteSection || !ConfigSetParameter || !ConfigGetParameter || |
| 231 | !ConfigSetDefaultInt || !ConfigSetDefaultFloat || !ConfigSetDefaultBool || !ConfigSetDefaultString || |
| 232 | !ConfigGetParamInt || !ConfigGetParamFloat || !ConfigGetParamBool || !ConfigGetParamString) |
| 233 | return M64ERR_INCOMPATIBLE; |
| 234 | |
| 235 | /* ConfigSaveSection was added in Config API v2.1.0 */ |
| 236 | if (ConfigAPIVersion >= 0x020100 && !ConfigSaveSection) |
| 237 | return M64ERR_INCOMPATIBLE; |
| 238 | |
| 239 | /* get a configuration section handle */ |
| 240 | if (ConfigOpenSection("Audio-SDL", &l_ConfigAudio) != M64ERR_SUCCESS) |
| 241 | { |
| 242 | DebugMessage(M64MSG_ERROR, "Couldn't open config section 'Audio-SDL'"); |
| 243 | return M64ERR_INPUT_NOT_FOUND; |
| 244 | } |
| 245 | |
| 246 | /* check the section version number */ |
| 247 | bSaveConfig = 0; |
| 248 | if (ConfigGetParameter(l_ConfigAudio, "Version", M64TYPE_FLOAT, &fConfigParamsVersion, sizeof(float)) != M64ERR_SUCCESS) |
| 249 | { |
| 250 | DebugMessage(M64MSG_WARNING, "No version number in 'Audio-SDL' config section. Setting defaults."); |
| 251 | ConfigDeleteSection("Audio-SDL"); |
| 252 | ConfigOpenSection("Audio-SDL", &l_ConfigAudio); |
| 253 | bSaveConfig = 1; |
| 254 | } |
| 255 | else if (((int) fConfigParamsVersion) != ((int) CONFIG_PARAM_VERSION)) |
| 256 | { |
| 257 | DebugMessage(M64MSG_WARNING, "Incompatible version %.2f in 'Audio-SDL' config section: current is %.2f. Setting defaults.", fConfigParamsVersion, (float) CONFIG_PARAM_VERSION); |
| 258 | ConfigDeleteSection("Audio-SDL"); |
| 259 | ConfigOpenSection("Audio-SDL", &l_ConfigAudio); |
| 260 | bSaveConfig = 1; |
| 261 | } |
| 262 | else if ((CONFIG_PARAM_VERSION - fConfigParamsVersion) >= 0.0001f) |
| 263 | { |
| 264 | /* handle upgrades */ |
| 265 | float fVersion = CONFIG_PARAM_VERSION; |
| 266 | ConfigSetParameter(l_ConfigAudio, "Version", M64TYPE_FLOAT, &fVersion); |
| 267 | DebugMessage(M64MSG_INFO, "Updating parameter set version in 'Audio-SDL' config section to %.2f", fVersion); |
| 268 | bSaveConfig = 1; |
| 269 | } |
| 270 | |
| 271 | /* set the default values for this plugin */ |
| 272 | ConfigSetDefaultFloat(l_ConfigAudio, "Version", CONFIG_PARAM_VERSION, "Mupen64Plus SDL Audio Plugin config parameter version number"); |
| 273 | ConfigSetDefaultInt(l_ConfigAudio, "DEFAULT_FREQUENCY", DEFAULT_FREQUENCY, "Frequency which is used if rom doesn't want to change it"); |
| 274 | ConfigSetDefaultBool(l_ConfigAudio, "SWAP_CHANNELS", 0, "Swaps left and right channels"); |
| 275 | ConfigSetDefaultInt(l_ConfigAudio, "PRIMARY_BUFFER_SIZE", PRIMARY_BUFFER_SIZE, "Size of primary buffer in output samples. This is where audio is loaded after it's extracted from n64's memory."); |
| 276 | ConfigSetDefaultInt(l_ConfigAudio, "PRIMARY_BUFFER_TARGET", PRIMARY_BUFFER_TARGET, "Fullness level target for Primary audio buffer, in equivalent output samples"); |
| 277 | ConfigSetDefaultInt(l_ConfigAudio, "SECONDARY_BUFFER_SIZE", SECONDARY_BUFFER_SIZE, "Size of secondary buffer in output samples. This is SDL's hardware buffer."); |
| 278 | ConfigSetDefaultString(l_ConfigAudio, "RESAMPLE", "trivial", "Audio resampling algorithm. src-sinc-best-quality, src-sinc-medium-quality, src-sinc-fastest, src-zero-order-hold, src-linear, speex-fixed-{10-0}, trivial"); |
| 279 | ConfigSetDefaultInt(l_ConfigAudio, "VOLUME_CONTROL_TYPE", VOLUME_TYPE_SDL, "Volume control type: 1 = SDL (only affects Mupen64Plus output) 2 = OSS mixer (adjusts master PC volume)"); |
| 280 | ConfigSetDefaultInt(l_ConfigAudio, "VOLUME_ADJUST", 5, "Percentage change each time the volume is increased or decreased"); |
| 281 | ConfigSetDefaultInt(l_ConfigAudio, "VOLUME_DEFAULT", 80, "Default volume when a game is started. Only used if VOLUME_CONTROL_TYPE is 1"); |
| 282 | |
| 283 | if (bSaveConfig && ConfigAPIVersion >= 0x020100) |
| 284 | ConfigSaveSection("Audio-SDL"); |
| 285 | |
| 286 | l_PluginInit = 1; |
| 287 | return M64ERR_SUCCESS; |
| 288 | } |
| 289 | |
| 290 | EXPORT m64p_error CALL PluginShutdown(void) |
| 291 | { |
| 292 | if (!l_PluginInit) |
| 293 | return M64ERR_NOT_INIT; |
| 294 | |
| 295 | /* reset some local variables */ |
| 296 | l_DebugCallback = NULL; |
| 297 | l_DebugCallContext = NULL; |
| 298 | |
| 299 | /* make sure our buffer is freed */ |
| 300 | if (mixBuffer != NULL) |
| 301 | { |
| 302 | free(mixBuffer); |
| 303 | mixBuffer = NULL; |
| 304 | } |
| 305 | |
| 306 | l_PluginInit = 0; |
| 307 | return M64ERR_SUCCESS; |
| 308 | } |
| 309 | |
| 310 | EXPORT m64p_error CALL PluginGetVersion(m64p_plugin_type *PluginType, int *PluginVersion, int *APIVersion, const char **PluginNamePtr, int *Capabilities) |
| 311 | { |
| 312 | /* set version info */ |
| 313 | if (PluginType != NULL) |
| 314 | *PluginType = M64PLUGIN_AUDIO; |
| 315 | |
| 316 | if (PluginVersion != NULL) |
| 317 | *PluginVersion = SDL_AUDIO_PLUGIN_VERSION; |
| 318 | |
| 319 | if (APIVersion != NULL) |
| 320 | *APIVersion = AUDIO_PLUGIN_API_VERSION; |
| 321 | |
| 322 | if (PluginNamePtr != NULL) |
| 323 | *PluginNamePtr = "Mupen64Plus SDL Audio Plugin"; |
| 324 | |
| 325 | if (Capabilities != NULL) |
| 326 | { |
| 327 | *Capabilities = 0; |
| 328 | } |
| 329 | |
| 330 | return M64ERR_SUCCESS; |
| 331 | } |
| 332 | |
| 333 | /* ----------- Audio Functions ------------- */ |
| 334 | EXPORT void CALL AiDacrateChanged( int SystemType ) |
| 335 | { |
| 336 | int f = GameFreq; |
| 337 | |
| 338 | if (!l_PluginInit) |
| 339 | return; |
| 340 | |
| 341 | switch (SystemType) |
| 342 | { |
| 343 | case SYSTEM_NTSC: |
| 344 | f = 48681812 / (*AudioInfo.AI_DACRATE_REG + 1); |
| 345 | break; |
| 346 | case SYSTEM_PAL: |
| 347 | f = 49656530 / (*AudioInfo.AI_DACRATE_REG + 1); |
| 348 | break; |
| 349 | case SYSTEM_MPAL: |
| 350 | f = 48628316 / (*AudioInfo.AI_DACRATE_REG + 1); |
| 351 | break; |
| 352 | } |
| 353 | InitializeAudio(f); |
| 354 | } |
| 355 | |
| 356 | |
| 357 | EXPORT void CALL AiLenChanged( void ) |
| 358 | { |
| 359 | unsigned int LenReg; |
| 360 | unsigned char *p; |
| 361 | unsigned int CurrLevel, CurrTime, ExpectedLevel, ExpectedTime; |
| 362 | |
| 363 | if (critical_failure == 1) |
| 364 | return; |
| 365 | if (!l_PluginInit) |
| 366 | return; |
| 367 | |
| 368 | LenReg = *AudioInfo.AI_LEN_REG; |
| 369 | p = AudioInfo.RDRAM + (*AudioInfo.AI_DRAM_ADDR_REG & 0xFFFFFF); |
| 370 | |
| 371 | if (buffer_pos + LenReg < primaryBufferBytes) |
| 372 | { |
| 373 | unsigned int i; |
| 374 | |
| 375 | SDL_LockAudio(); |
| 376 | for ( i = 0 ; i < LenReg ; i += 4 ) |
| 377 | { |
| 378 | |
| 379 | if(SwapChannels == 0) |
| 380 | { |
| 381 | // Left channel |
| 382 | primaryBuffer[ buffer_pos + i ] = p[ i + 2 ]; |
| 383 | primaryBuffer[ buffer_pos + i + 1 ] = p[ i + 3 ]; |
| 384 | |
| 385 | // Right channel |
| 386 | primaryBuffer[ buffer_pos + i + 2 ] = p[ i ]; |
| 387 | primaryBuffer[ buffer_pos + i + 3 ] = p[ i + 1 ]; |
| 388 | } else { |
| 389 | // Left channel |
| 390 | primaryBuffer[ buffer_pos + i ] = p[ i ]; |
| 391 | primaryBuffer[ buffer_pos + i + 1 ] = p[ i + 1 ]; |
| 392 | |
| 393 | // Right channel |
| 394 | primaryBuffer[ buffer_pos + i + 2 ] = p[ i + 2]; |
| 395 | primaryBuffer[ buffer_pos + i + 3 ] = p[ i + 3 ]; |
| 396 | } |
| 397 | } |
| 398 | buffer_pos += i; |
| 399 | SDL_UnlockAudio(); |
| 400 | } |
| 401 | else |
| 402 | { |
| 403 | DebugMessage(M64MSG_WARNING, "AiLenChanged(): Audio buffer overflow."); |
| 404 | } |
| 405 | |
| 406 | /* Now we need to handle synchronization, by inserting time delay to keep the emulator running at the correct speed */ |
| 407 | /* Start by calculating the current Primary buffer fullness in terms of output samples */ |
| 408 | CurrLevel = (unsigned int) (((long long) (buffer_pos/N64_SAMPLE_BYTES) * OutputFreq * 100) / (GameFreq * speed_factor)); |
| 409 | /* Next, extrapolate to the buffer level at the expected time of the next audio callback, assuming that the |
| 410 | buffer is filled at the same rate as the output frequency */ |
| 411 | CurrTime = SDL_GetTicks(); |
| 412 | ExpectedTime = last_callback_ticks + ((1000 * SecondaryBufferSize) / OutputFreq); |
| 413 | ExpectedLevel = CurrLevel; |
| 414 | if (CurrTime < ExpectedTime) |
| 415 | ExpectedLevel += (ExpectedTime - CurrTime) * OutputFreq / 1000; |
| 416 | /* If the expected value of the Primary Buffer Fullness at the time of the next audio callback is more than 10 |
| 417 | milliseconds ahead of our target buffer fullness level, then insert a delay now */ |
| 418 | DebugMessage(M64MSG_VERBOSE, "%03i New audio bytes: %i Time to next callback: %i Current/Expected buffer level: %i/%i", |
| 419 | CurrTime % 1000, LenReg, (int) (ExpectedTime - CurrTime), CurrLevel, ExpectedLevel); |
| 420 | if (ExpectedLevel >= PrimaryBufferTarget + OutputFreq / 100) |
| 421 | { |
| 422 | unsigned int WaitTime = (ExpectedLevel - PrimaryBufferTarget) * 1000 / OutputFreq; |
| 423 | DebugMessage(M64MSG_VERBOSE, " AiLenChanged(): Waiting %ims", WaitTime); |
| 424 | if (l_PausedForSync) |
| 425 | SDL_PauseAudio(0); |
| 426 | l_PausedForSync = 0; |
| 427 | SDL_Delay(WaitTime); |
| 428 | } |
| 429 | /* Or if the expected level of the primary buffer is less than the secondary buffer size |
| 430 | (ie, predicting an underflow), then pause the audio to let the emulator catch up to speed */ |
| 431 | else if (ExpectedLevel < SecondaryBufferSize) |
| 432 | { |
| 433 | DebugMessage(M64MSG_VERBOSE, " AiLenChanged(): Possible underflow at next audio callback; pausing playback"); |
| 434 | if (!l_PausedForSync) |
| 435 | SDL_PauseAudio(1); |
| 436 | l_PausedForSync = 1; |
| 437 | } |
| 438 | /* otherwise the predicted buffer level is within our tolerance, so everything is okay */ |
| 439 | else |
| 440 | { |
| 441 | if (l_PausedForSync) |
| 442 | SDL_PauseAudio(0); |
| 443 | l_PausedForSync = 0; |
| 444 | } |
| 445 | } |
| 446 | |
| 447 | EXPORT int CALL InitiateAudio( AUDIO_INFO Audio_Info ) |
| 448 | { |
| 449 | if (!l_PluginInit) |
| 450 | return 0; |
| 451 | |
| 452 | AudioInfo = Audio_Info; |
| 453 | return 1; |
| 454 | } |
| 455 | |
| 456 | static int underrun_count = 0; |
| 457 | |
| 458 | #ifdef USE_SRC |
| 459 | static float *_src = NULL; |
| 460 | static unsigned int _src_len = 0; |
| 461 | static float *_dest = NULL; |
| 462 | static unsigned int _dest_len = 0; |
| 463 | static int error; |
| 464 | static SRC_STATE *src_state; |
| 465 | static SRC_DATA src_data; |
| 466 | #endif |
| 467 | #ifdef USE_SPEEX |
| 468 | SpeexResamplerState* spx_state = NULL; |
| 469 | static int error; |
| 470 | #endif |
| 471 | |
| 472 | static int resample(unsigned char *input, int input_avail, int oldsamplerate, unsigned char *output, int output_needed, int newsamplerate) |
| 473 | { |
| 474 | int *psrc = (int*)input; |
| 475 | int *pdest = (int*)output; |
| 476 | int i = 0, j = 0; |
| 477 | |
| 478 | #ifdef USE_SPEEX |
| 479 | spx_uint32_t in_len, out_len; |
| 480 | if(Resample == RESAMPLER_SPEEX) |
| 481 | { |
| 482 | if(spx_state == NULL) |
| 483 | { |
| 484 | spx_state = speex_resampler_init(2, oldsamplerate, newsamplerate, ResampleQuality, &error); |
| 485 | if(spx_state == NULL) |
| 486 | { |
| 487 | memset(output, 0, output_needed); |
| 488 | return 0; |
| 489 | } |
| 490 | } |
| 491 | speex_resampler_set_rate(spx_state, oldsamplerate, newsamplerate); |
| 492 | in_len = input_avail / 4; |
| 493 | out_len = output_needed / 4; |
| 494 | |
| 495 | if ((error = speex_resampler_process_interleaved_int(spx_state, (const spx_int16_t *)input, &in_len, (spx_int16_t *)output, &out_len))) |
| 496 | { |
| 497 | memset(output, 0, output_needed); |
| 498 | return input_avail; // number of bytes consumed |
| 499 | } |
| 500 | return in_len * 4; |
| 501 | } |
| 502 | #endif |
| 503 | #ifdef USE_SRC |
| 504 | if(Resample == RESAMPLER_SRC) |
| 505 | { |
| 506 | // the high quality resampler needs more input than the samplerate ratio would indicate to work properly |
| 507 | if (input_avail > output_needed * 3 / 2) |
| 508 | input_avail = output_needed * 3 / 2; // just to avoid too much short-float-short conversion time |
| 509 | if (_src_len < input_avail*2 && input_avail > 0) |
| 510 | { |
| 511 | if(_src) free(_src); |
| 512 | _src_len = input_avail*2; |
| 513 | _src = malloc(_src_len); |
| 514 | } |
| 515 | if (_dest_len < output_needed*2 && output_needed > 0) |
| 516 | { |
| 517 | if(_dest) free(_dest); |
| 518 | _dest_len = output_needed*2; |
| 519 | _dest = malloc(_dest_len); |
| 520 | } |
| 521 | memset(_src,0,_src_len); |
| 522 | memset(_dest,0,_dest_len); |
| 523 | if(src_state == NULL) |
| 524 | { |
| 525 | src_state = src_new (ResampleQuality, 2, &error); |
| 526 | if(src_state == NULL) |
| 527 | { |
| 528 | memset(output, 0, output_needed); |
| 529 | return 0; |
| 530 | } |
| 531 | } |
| 532 | src_short_to_float_array ((short *) input, _src, input_avail/2); |
| 533 | src_data.end_of_input = 0; |
| 534 | src_data.data_in = _src; |
| 535 | src_data.input_frames = input_avail/4; |
| 536 | src_data.src_ratio = (float) newsamplerate / oldsamplerate; |
| 537 | src_data.data_out = _dest; |
| 538 | src_data.output_frames = output_needed/4; |
| 539 | if ((error = src_process (src_state, &src_data))) |
| 540 | { |
| 541 | memset(output, 0, output_needed); |
| 542 | return input_avail; // number of bytes consumed |
| 543 | } |
| 544 | src_float_to_short_array (_dest, (short *) output, output_needed/2); |
| 545 | return src_data.input_frames_used * 4; |
| 546 | } |
| 547 | #endif |
| 548 | // RESAMPLE == TRIVIAL |
| 549 | if (newsamplerate >= oldsamplerate) |
| 550 | { |
| 551 | int sldf = oldsamplerate; |
| 552 | int const2 = 2*sldf; |
| 553 | int dldf = newsamplerate; |
| 554 | int const1 = const2 - 2*dldf; |
| 555 | int criteria = const2 - dldf; |
| 556 | for (i = 0; i < output_needed/4; i++) |
| 557 | { |
| 558 | pdest[i] = psrc[j]; |
| 559 | if(criteria >= 0) |
| 560 | { |
| 561 | ++j; |
| 562 | criteria += const1; |
| 563 | } |
| 564 | else criteria += const2; |
| 565 | } |
| 566 | return j * 4; //number of bytes consumed |
| 567 | } |
| 568 | // newsamplerate < oldsamplerate, this only happens when speed_factor > 1 |
| 569 | for (i = 0; i < output_needed/4; i++) |
| 570 | { |
| 571 | j = i * oldsamplerate / newsamplerate; |
| 572 | pdest[i] = psrc[j]; |
| 573 | } |
| 574 | return j * 4; //number of bytes consumed |
| 575 | } |
| 576 | |
| 577 | static void my_audio_callback(void *userdata, unsigned char *stream, int len) |
| 578 | { |
| 579 | int oldsamplerate, newsamplerate; |
| 580 | |
| 581 | if (!l_PluginInit) |
| 582 | return; |
| 583 | |
| 584 | /* mark the time, for synchronization on the input side */ |
| 585 | last_callback_ticks = SDL_GetTicks(); |
| 586 | |
| 587 | newsamplerate = OutputFreq * 100 / speed_factor; |
| 588 | oldsamplerate = GameFreq; |
| 589 | |
| 590 | if (buffer_pos > (unsigned int) (len * oldsamplerate) / newsamplerate) |
| 591 | { |
| 592 | int input_used; |
| 593 | #if defined(HAS_OSS_SUPPORT) |
| 594 | if (VolumeControlType == VOLUME_TYPE_OSS) |
| 595 | { |
| 596 | input_used = resample(primaryBuffer, buffer_pos, oldsamplerate, stream, len, newsamplerate); |
| 597 | } |
| 598 | else |
| 599 | #endif |
| 600 | { |
| 601 | input_used = resample(primaryBuffer, buffer_pos, oldsamplerate, mixBuffer, len, newsamplerate); |
| 602 | memset(stream, 0, len); |
| 603 | SDL_MixAudio(stream, mixBuffer, len, VolSDL); |
| 604 | } |
| 605 | memmove(primaryBuffer, &primaryBuffer[input_used], buffer_pos - input_used); |
| 606 | buffer_pos -= input_used; |
| 607 | DebugMessage(M64MSG_VERBOSE, "%03i my_audio_callback: used %i samples", |
| 608 | last_callback_ticks % 1000, len / SDL_SAMPLE_BYTES); |
| 609 | } |
| 610 | else |
| 611 | { |
| 612 | unsigned int SamplesNeeded = (len * oldsamplerate) / (newsamplerate * SDL_SAMPLE_BYTES); |
| 613 | unsigned int SamplesPresent = buffer_pos / N64_SAMPLE_BYTES; |
| 614 | underrun_count++; |
| 615 | DebugMessage(M64MSG_VERBOSE, "%03i Buffer underflow (%i). %i samples present, %i needed", |
| 616 | last_callback_ticks % 1000, underrun_count, SamplesPresent, SamplesNeeded); |
| 617 | memset(stream , 0, len); |
| 618 | } |
| 619 | } |
| 620 | EXPORT int CALL RomOpen(void) |
| 621 | { |
| 622 | if (!l_PluginInit) |
| 623 | return 0; |
| 624 | |
| 625 | ReadConfig(); |
| 626 | InitializeAudio(GameFreq); |
| 627 | return 1; |
| 628 | } |
| 629 | |
| 630 | static void InitializeSDL(void) |
| 631 | { |
| 632 | DebugMessage(M64MSG_INFO, "Initializing SDL audio subsystem..."); |
| 633 | |
| 634 | if(SDL_Init(SDL_INIT_AUDIO | SDL_INIT_TIMER) < 0) |
| 635 | { |
| 636 | DebugMessage(M64MSG_ERROR, "Failed to initialize SDL audio subsystem; forcing exit.\n"); |
| 637 | critical_failure = 1; |
| 638 | return; |
| 639 | } |
| 640 | critical_failure = 0; |
| 641 | |
| 642 | } |
| 643 | |
| 644 | static void CreatePrimaryBuffer(void) |
| 645 | { |
| 646 | unsigned int newPrimaryBytes = (unsigned int) ((long long) PrimaryBufferSize * GameFreq * speed_factor / |
| 647 | (OutputFreq * 100)) * N64_SAMPLE_BYTES; |
| 648 | if (primaryBuffer == NULL) |
| 649 | { |
| 650 | DebugMessage(M64MSG_VERBOSE, "Allocating memory for audio buffer: %i bytes.", newPrimaryBytes); |
| 651 | primaryBuffer = (unsigned char*) malloc(newPrimaryBytes); |
| 652 | memset(primaryBuffer, 0, newPrimaryBytes); |
| 653 | primaryBufferBytes = newPrimaryBytes; |
| 654 | } |
| 655 | else if (newPrimaryBytes > primaryBufferBytes) /* primary buffer only grows; there's no point in shrinking it */ |
| 656 | { |
| 657 | unsigned char *newPrimaryBuffer = (unsigned char*) malloc(newPrimaryBytes); |
| 658 | unsigned char *oldPrimaryBuffer = primaryBuffer; |
| 659 | SDL_LockAudio(); |
| 660 | memcpy(newPrimaryBuffer, oldPrimaryBuffer, primaryBufferBytes); |
| 661 | memset(newPrimaryBuffer + primaryBufferBytes, 0, newPrimaryBytes - primaryBufferBytes); |
| 662 | primaryBuffer = newPrimaryBuffer; |
| 663 | primaryBufferBytes = newPrimaryBytes; |
| 664 | SDL_UnlockAudio(); |
| 665 | free(oldPrimaryBuffer); |
| 666 | } |
| 667 | } |
| 668 | |
| 669 | static void InitializeAudio(int freq) |
| 670 | { |
| 671 | SDL_AudioSpec *desired, *obtained; |
| 672 | |
| 673 | if(SDL_WasInit(SDL_INIT_AUDIO|SDL_INIT_TIMER) == (SDL_INIT_AUDIO|SDL_INIT_TIMER) ) |
| 674 | { |
| 675 | DebugMessage(M64MSG_VERBOSE, "InitializeAudio(): SDL Audio sub-system already initialized."); |
| 676 | SDL_PauseAudio(1); |
| 677 | SDL_CloseAudio(); |
| 678 | } |
| 679 | else |
| 680 | { |
| 681 | DebugMessage(M64MSG_VERBOSE, "InitializeAudio(): Initializing SDL Audio"); |
| 682 | DebugMessage(M64MSG_VERBOSE, "Primary buffer: %i output samples.", PrimaryBufferSize); |
| 683 | DebugMessage(M64MSG_VERBOSE, "Primary target fullness: %i output samples.", PrimaryBufferTarget); |
| 684 | DebugMessage(M64MSG_VERBOSE, "Secondary buffer: %i output samples.", SecondaryBufferSize); |
| 685 | InitializeSDL(); |
| 686 | } |
| 687 | if (critical_failure == 1) |
| 688 | return; |
| 689 | GameFreq = freq; // This is important for the sync |
| 690 | if(hardware_spec != NULL) free(hardware_spec); |
| 691 | |
| 692 | // Allocate space for SDL_AudioSpec |
| 693 | desired = malloc(sizeof(SDL_AudioSpec)); |
| 694 | obtained = malloc(sizeof(SDL_AudioSpec)); |
| 695 | |
| 696 | if(freq < 11025) OutputFreq = 11025; |
| 697 | else if(freq < 22050) OutputFreq = 22050; |
| 698 | else OutputFreq = 44100; |
| 699 | |
| 700 | desired->freq = OutputFreq; |
| 701 | |
| 702 | DebugMessage(M64MSG_VERBOSE, "Requesting frequency: %iHz.", desired->freq); |
| 703 | /* 16-bit signed audio */ |
| 704 | desired->format=AUDIO_S16SYS; |
| 705 | DebugMessage(M64MSG_VERBOSE, "Requesting format: %i.", desired->format); |
| 706 | /* Stereo */ |
| 707 | desired->channels=2; |
| 708 | /* reload these because they gets re-assigned from SDL data below, and InitializeAudio can be called more than once */ |
| 709 | PrimaryBufferSize = ConfigGetParamInt(l_ConfigAudio, "PRIMARY_BUFFER_SIZE"); |
| 710 | PrimaryBufferTarget = ConfigGetParamInt(l_ConfigAudio, "PRIMARY_BUFFER_TARGET"); |
| 711 | SecondaryBufferSize = ConfigGetParamInt(l_ConfigAudio, "SECONDARY_BUFFER_SIZE"); |
| 712 | desired->samples = SecondaryBufferSize; |
| 713 | /* Our callback function */ |
| 714 | desired->callback = my_audio_callback; |
| 715 | desired->userdata = NULL; |
| 716 | |
| 717 | /* Open the audio device */ |
| 718 | l_PausedForSync = 1; |
| 719 | if (SDL_OpenAudio(desired, obtained) < 0) |
| 720 | { |
| 721 | DebugMessage(M64MSG_ERROR, "Couldn't open audio: %s", SDL_GetError()); |
| 722 | critical_failure = 1; |
| 723 | return; |
| 724 | } |
| 725 | if (desired->format != obtained->format) |
| 726 | { |
| 727 | DebugMessage(M64MSG_WARNING, "Obtained audio format differs from requested."); |
| 728 | } |
| 729 | if (desired->freq != obtained->freq) |
| 730 | { |
| 731 | DebugMessage(M64MSG_WARNING, "Obtained frequency differs from requested."); |
| 732 | } |
| 733 | |
| 734 | /* desired spec is no longer needed */ |
| 735 | free(desired); |
| 736 | hardware_spec=obtained; |
| 737 | |
| 738 | /* allocate memory for audio buffers */ |
| 739 | OutputFreq = hardware_spec->freq; |
| 740 | SecondaryBufferSize = hardware_spec->samples; |
| 741 | if (PrimaryBufferTarget < SecondaryBufferSize) |
| 742 | PrimaryBufferTarget = SecondaryBufferSize; |
| 743 | if (PrimaryBufferSize < PrimaryBufferTarget) |
| 744 | PrimaryBufferSize = PrimaryBufferTarget; |
| 745 | if (PrimaryBufferSize < SecondaryBufferSize * 2) |
| 746 | PrimaryBufferSize = SecondaryBufferSize * 2; |
| 747 | CreatePrimaryBuffer(); |
| 748 | if (mixBuffer != NULL) |
| 749 | free(mixBuffer); |
| 750 | mixBuffer = (unsigned char*) malloc(SecondaryBufferSize * SDL_SAMPLE_BYTES); |
| 751 | |
| 752 | /* preset the last callback time */ |
| 753 | if (last_callback_ticks == 0) |
| 754 | last_callback_ticks = SDL_GetTicks(); |
| 755 | |
| 756 | DebugMessage(M64MSG_VERBOSE, "Frequency: %i", hardware_spec->freq); |
| 757 | DebugMessage(M64MSG_VERBOSE, "Format: %i", hardware_spec->format); |
| 758 | DebugMessage(M64MSG_VERBOSE, "Channels: %i", hardware_spec->channels); |
| 759 | DebugMessage(M64MSG_VERBOSE, "Silence: %i", hardware_spec->silence); |
| 760 | DebugMessage(M64MSG_VERBOSE, "Samples: %i", hardware_spec->samples); |
| 761 | DebugMessage(M64MSG_VERBOSE, "Size: %i", hardware_spec->size); |
| 762 | |
| 763 | /* set playback volume */ |
| 764 | #if defined(HAS_OSS_SUPPORT) |
| 765 | if (VolumeControlType == VOLUME_TYPE_OSS) |
| 766 | { |
| 767 | VolPercent = volGet(); |
| 768 | } |
| 769 | else |
| 770 | #endif |
| 771 | { |
| 772 | VolSDL = SDL_MIX_MAXVOLUME * VolPercent / 100; |
| 773 | } |
| 774 | |
| 775 | } |
| 776 | EXPORT void CALL RomClosed( void ) |
| 777 | { |
| 778 | if (!l_PluginInit) |
| 779 | return; |
| 780 | if (critical_failure == 1) |
| 781 | return; |
| 782 | DebugMessage(M64MSG_VERBOSE, "Cleaning up SDL sound plugin..."); |
| 783 | |
| 784 | // Shut down SDL Audio output |
| 785 | SDL_PauseAudio(1); |
| 786 | SDL_CloseAudio(); |
| 787 | |
| 788 | // Delete the buffer, as we are done producing sound |
| 789 | if (primaryBuffer != NULL) |
| 790 | { |
| 791 | primaryBufferBytes = 0; |
| 792 | free(primaryBuffer); |
| 793 | primaryBuffer = NULL; |
| 794 | } |
| 795 | if (mixBuffer != NULL) |
| 796 | { |
| 797 | free(mixBuffer); |
| 798 | mixBuffer = NULL; |
| 799 | } |
| 800 | |
| 801 | // Delete the hardware spec struct |
| 802 | if(hardware_spec != NULL) free(hardware_spec); |
| 803 | hardware_spec = NULL; |
| 804 | |
| 805 | // Shutdown the respective subsystems |
| 806 | if(SDL_WasInit(SDL_INIT_AUDIO) != 0) SDL_QuitSubSystem(SDL_INIT_AUDIO); |
| 807 | if(SDL_WasInit(SDL_INIT_TIMER) != 0) SDL_QuitSubSystem(SDL_INIT_TIMER); |
| 808 | } |
| 809 | |
| 810 | EXPORT void CALL ProcessAList(void) |
| 811 | { |
| 812 | } |
| 813 | |
| 814 | EXPORT void CALL SetSpeedFactor(int percentage) |
| 815 | { |
| 816 | if (!l_PluginInit) |
| 817 | return; |
| 818 | if (percentage >= 10 && percentage <= 300) |
| 819 | speed_factor = percentage; |
| 820 | // we need a different size primary buffer to store the N64 samples when the speed changes |
| 821 | CreatePrimaryBuffer(); |
| 822 | } |
| 823 | |
| 824 | static void ReadConfig(void) |
| 825 | { |
| 826 | const char *resampler_id; |
| 827 | |
| 828 | /* read the configuration values into our static variables */ |
| 829 | GameFreq = ConfigGetParamInt(l_ConfigAudio, "DEFAULT_FREQUENCY"); |
| 830 | SwapChannels = ConfigGetParamBool(l_ConfigAudio, "SWAP_CHANNELS"); |
| 831 | PrimaryBufferSize = ConfigGetParamInt(l_ConfigAudio, "PRIMARY_BUFFER_SIZE"); |
| 832 | PrimaryBufferTarget = ConfigGetParamInt(l_ConfigAudio, "PRIMARY_BUFFER_TARGET"); |
| 833 | SecondaryBufferSize = ConfigGetParamInt(l_ConfigAudio, "SECONDARY_BUFFER_SIZE"); |
| 834 | resampler_id = ConfigGetParamString(l_ConfigAudio, "RESAMPLE"); |
| 835 | VolumeControlType = ConfigGetParamInt(l_ConfigAudio, "VOLUME_CONTROL_TYPE"); |
| 836 | VolDelta = ConfigGetParamInt(l_ConfigAudio, "VOLUME_ADJUST"); |
| 837 | VolPercent = ConfigGetParamInt(l_ConfigAudio, "VOLUME_DEFAULT"); |
| 838 | |
| 839 | if (!resampler_id) { |
| 840 | Resample = RESAMPLER_TRIVIAL; |
| 841 | DebugMessage(M64MSG_WARNING, "Could not find RESAMPLE configuration; use trivial resampler"); |
| 842 | return; |
| 843 | } |
| 844 | if (strcmp(resampler_id, "trivial") == 0) { |
| 845 | Resample = RESAMPLER_TRIVIAL; |
| 846 | return; |
| 847 | } |
| 848 | #ifdef USE_SPEEX |
| 849 | if (strncmp(resampler_id, "speex-fixed-", strlen("speex-fixed-")) == 0) { |
| 850 | int i; |
| 851 | static const char *speex_quality[] = { |
| 852 | "speex-fixed-0", |
| 853 | "speex-fixed-1", |
| 854 | "speex-fixed-2", |
| 855 | "speex-fixed-3", |
| 856 | "speex-fixed-4", |
| 857 | "speex-fixed-5", |
| 858 | "speex-fixed-6", |
| 859 | "speex-fixed-7", |
| 860 | "speex-fixed-8", |
| 861 | "speex-fixed-9", |
| 862 | "speex-fixed-10", |
| 863 | }; |
| 864 | Resample = RESAMPLER_SPEEX; |
| 865 | for (i = 0; i < sizeof(speex_quality) / sizeof(*speex_quality); i++) { |
| 866 | if (strcmp(speex_quality[i], resampler_id) == 0) { |
| 867 | ResampleQuality = i; |
| 868 | return; |
| 869 | } |
| 870 | } |
| 871 | DebugMessage(M64MSG_WARNING, "Unknown RESAMPLE configuration %s; use speex-fixed-4 resampler", resampler_id); |
| 872 | ResampleQuality = 4; |
| 873 | return; |
| 874 | } |
| 875 | #endif |
| 876 | #ifdef USE_SRC |
| 877 | if (strncmp(resampler_id, "src-", strlen("src-")) == 0) { |
| 878 | Resample = RESAMPLER_SRC; |
| 879 | if (strcmp(resampler_id, "src-sinc-best-quality") == 0) { |
| 880 | ResampleQuality = SRC_SINC_BEST_QUALITY; |
| 881 | return; |
| 882 | } |
| 883 | if (strcmp(resampler_id, "src-sinc-medium-quality") == 0) { |
| 884 | ResampleQuality = SRC_SINC_MEDIUM_QUALITY; |
| 885 | return; |
| 886 | } |
| 887 | if (strcmp(resampler_id, "src-sinc-fastest") == 0) { |
| 888 | ResampleQuality = SRC_SINC_FASTEST; |
| 889 | return; |
| 890 | } |
| 891 | if (strcmp(resampler_id, "src-zero-order-hold") == 0) { |
| 892 | ResampleQuality = SRC_ZERO_ORDER_HOLD; |
| 893 | return; |
| 894 | } |
| 895 | if (strcmp(resampler_id, "src-linear") == 0) { |
| 896 | ResampleQuality = SRC_LINEAR; |
| 897 | return; |
| 898 | } |
| 899 | DebugMessage(M64MSG_WARNING, "Unknown RESAMPLE configuration %s; use src-sinc-medium-quality resampler", resampler_id); |
| 900 | ResampleQuality = SRC_SINC_MEDIUM_QUALITY; |
| 901 | return; |
| 902 | } |
| 903 | #endif |
| 904 | DebugMessage(M64MSG_WARNING, "Unknown RESAMPLE configuration %s; use trivial resampler", resampler_id); |
| 905 | Resample = RESAMPLER_TRIVIAL; |
| 906 | } |
| 907 | |
| 908 | // Returns the most recent ummuted volume level. |
| 909 | static int VolumeGetUnmutedLevel(void) |
| 910 | { |
| 911 | #if defined(HAS_OSS_SUPPORT) |
| 912 | // reload volume if we're using OSS |
| 913 | if (!VolIsMuted && VolumeControlType == VOLUME_TYPE_OSS) |
| 914 | { |
| 915 | return volGet(); |
| 916 | } |
| 917 | #endif |
| 918 | |
| 919 | return VolPercent; |
| 920 | } |
| 921 | |
| 922 | // Sets the volume level based on the contents of VolPercent and VolIsMuted |
| 923 | static void VolumeCommit(void) |
| 924 | { |
| 925 | int levelToCommit = VolIsMuted ? 0 : VolPercent; |
| 926 | |
| 927 | #if defined(HAS_OSS_SUPPORT) |
| 928 | if (VolumeControlType == VOLUME_TYPE_OSS) |
| 929 | { |
| 930 | //OSS mixer volume |
| 931 | volSet(levelToCommit); |
| 932 | } |
| 933 | else |
| 934 | #endif |
| 935 | { |
| 936 | VolSDL = SDL_MIX_MAXVOLUME * levelToCommit / 100; |
| 937 | } |
| 938 | } |
| 939 | |
| 940 | EXPORT void CALL VolumeMute(void) |
| 941 | { |
| 942 | if (!l_PluginInit) |
| 943 | return; |
| 944 | |
| 945 | // Store the volume level in order to restore it later |
| 946 | if (!VolIsMuted) |
| 947 | VolPercent = VolumeGetUnmutedLevel(); |
| 948 | |
| 949 | // Toogle mute |
| 950 | VolIsMuted = !VolIsMuted; |
| 951 | VolumeCommit(); |
| 952 | } |
| 953 | |
| 954 | EXPORT void CALL VolumeUp(void) |
| 955 | { |
| 956 | if (!l_PluginInit) |
| 957 | return; |
| 958 | |
| 959 | VolumeSetLevel(VolumeGetUnmutedLevel() + VolDelta); |
| 960 | } |
| 961 | |
| 962 | EXPORT void CALL VolumeDown(void) |
| 963 | { |
| 964 | if (!l_PluginInit) |
| 965 | return; |
| 966 | |
| 967 | VolumeSetLevel(VolumeGetUnmutedLevel() - VolDelta); |
| 968 | } |
| 969 | |
| 970 | EXPORT int CALL VolumeGetLevel(void) |
| 971 | { |
| 972 | return VolIsMuted ? 0 : VolumeGetUnmutedLevel(); |
| 973 | } |
| 974 | |
| 975 | EXPORT void CALL VolumeSetLevel(int level) |
| 976 | { |
| 977 | if (!l_PluginInit) |
| 978 | return; |
| 979 | |
| 980 | //if muted, unmute first |
| 981 | VolIsMuted = 0; |
| 982 | |
| 983 | // adjust volume |
| 984 | VolPercent = level; |
| 985 | if (VolPercent < 0) |
| 986 | VolPercent = 0; |
| 987 | else if (VolPercent > 100) |
| 988 | VolPercent = 100; |
| 989 | |
| 990 | VolumeCommit(); |
| 991 | } |
| 992 | |
| 993 | EXPORT const char * CALL VolumeGetString(void) |
| 994 | { |
| 995 | static char VolumeString[32]; |
| 996 | |
| 997 | if (VolIsMuted) |
| 998 | { |
| 999 | strcpy(VolumeString, "Mute"); |
| 1000 | } |
| 1001 | else |
| 1002 | { |
| 1003 | sprintf(VolumeString, "%i%%", VolPercent); |
| 1004 | } |
| 1005 | |
| 1006 | return VolumeString; |
| 1007 | } |
| 1008 | |