Mercurial > almixer_isolated
annotate ALmixer.c @ 10:c808684660a7
Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
Behavior change: Decided to make finishedNaturally return false if ending was due to expired ticks.
| author | Eric Wing <ewing . public |-at-| gmail . com> |
|---|---|
| date | Sun, 31 Oct 2010 05:49:44 -0700 |
| parents | 4b1048af7e55 |
| children | bfe90b4f3d87 |
| rev | line source |
|---|---|
| 0 | 1 |
| 2 /* Here's an OpenAL implementation modeled after | |
| 3 * the SDL_SoundMixer which was built ontop of SDL_Mixer | |
| 4 * and SDL_Sound. | |
| 5 * Eric Wing | |
| 6 */ | |
| 7 | |
|
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8 #include "ALmixer.h" |
| 0 | 9 |
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10 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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11 #include "ALmixer_rwops.h" |
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12 #include "SoundDecoder.h" |
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13 #else |
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14 #include "SDL_sound.h" |
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15 #endif |
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16 |
| 0 | 17 #include "al.h" /* OpenAL */ |
| 18 #include "alc.h" /* For creating OpenAL contexts */ | |
| 19 | |
| 1 | 20 #ifdef __APPLE__ |
| 21 /* For performance things like ALC_CONVERT_DATA_UPON_LOADING */ | |
| 22 /* Note: ALC_CONVERT_DATA_UPON_LOADING used to be in the alc.h header. | |
| 23 * But in the Tiger OpenAL 1.1 release (10.4.7 and Xcode 2.4), the | |
| 24 * define was moved to a new header file and renamed to | |
| 25 * ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING. | |
| 26 */ | |
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27 /* |
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28 #include <TargetConditionals.h> |
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29 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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30 |
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31 #else |
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32 #include <OpenAL/MacOSX_OALExtensions.h> |
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33 #endif |
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34 */ |
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35 |
| 1 | 36 #endif |
| 37 | |
| 0 | 38 /* For malloc, bsearch, qsort */ |
| 39 #include <stdlib.h> | |
| 40 | |
| 41 /* For memcpy */ | |
| 42 #include <string.h> | |
| 43 | |
| 44 #if 0 | |
| 45 /* for toupper */ | |
| 46 #include <ctype.h> | |
| 47 /* for strrchr */ | |
| 48 #include <string.h> | |
| 49 #endif | |
| 50 | |
| 51 /* Currently used in the output debug functions */ | |
| 52 #include <stdio.h> | |
| 53 | |
| 54 /* My own CircularQueue implementation needed | |
| 55 * to work around the Nvidia problem of the | |
| 56 * lack of a buffer query. | |
| 57 */ | |
| 58 #include "CircularQueue.h" | |
| 59 | |
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60 #ifdef ENABLE_ALMIXER_THREADS |
| 0 | 61 /* Needed for the Mutex locks (and threads if enabled) */ |
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62 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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63 #include "SimpleMutex.h" |
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64 #include "SimpleThread.h" |
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65 typedef struct SimpleMutex SDL_mutex; |
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66 typedef struct SimpleThread SDL_Thread; |
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67 #define SDL_CreateMutex SimpleMutex_CreateMutex |
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68 #define SDL_DestroyMutex SimpleMutex_DestroyMutex |
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69 #define SDL_LockMutex SimpleMutex_LockMutex |
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70 #define SDL_UnlockMutex SimpleMutex_UnlockMutex |
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71 #define SDL_CreateThread SimpleThread_CreateThread |
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72 #define SDL_WaitThread SimpleThread_WaitThread |
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73 |
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74 #else |
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75 #include "SDL_thread.h" |
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76 #endif |
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77 #endif |
| 0 | 78 |
| 79 /* Because of the API differences between the Loki | |
| 80 * and Creative distributions, we need to know which | |
| 81 * version to use. The LOKI distribution currently | |
| 82 * has AL_BYTE_LOKI defined in altypes.h which | |
| 83 * I will use as a flag to identify the distributions. | |
| 84 * If this is ever removed, I might revert back to the | |
| 85 * if defined(_WIN32) or defined(__APPLE__) test to | |
| 86 * identify the Creative dist. | |
| 87 * I'm not sure if or how the Nvidia distribution differs | |
| 88 * from the Creative distribution. So for | |
| 89 * now, the Nvidia distribution gets lumped with the | |
| 90 * Creative dist and I hope nothing will break. | |
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91 * My alGetString may be the most vulnerable. |
| 0 | 92 */ |
| 93 #ifdef AL_BYTE_LOKI | |
| 94 #define USING_LOKI_AL_DIST | |
| 95 /* This is a short term fix to get around the | |
| 96 * queuing problem with non-power of two buffer sizes. | |
| 97 * Hopefully the maintainers will fix this before | |
| 98 * we're ready to ship. | |
| 99 */ | |
| 100 #define ENABLE_LOKI_QUEUE_FIX_HACK | |
| 101 | |
| 102 /* The AL_GAIN in the Loki dist doesn't seem to do | |
| 103 * what I want/expect it to do. I want to use it for | |
| 104 * Fading, but it seems to work like an off/on switch. | |
| 105 * 0 = off, >0 = on. | |
| 106 * The AL_GAIN_LINEAR_LOKI switch seems to do what | |
| 107 * I want, so I'll redefine it here so the code is consistent | |
| 108 */ | |
| 109 /* Update: I've changed the source volume implementations | |
| 110 * to use AL_MAX_GAIN, so I don't think I need this block | |
| 111 * of code anymore. The listener uses AL_GAIN, but I | |
| 112 * hope they got this one right since there isn't a AL_MAX_GAIN | |
| 113 * for the listener. | |
| 114 */ | |
| 115 /* | |
| 116 #undef AL_GAIN | |
| 117 #include "alexttypes.h" | |
| 118 #define AL_GAIN AL_GAIN_LINEAR_LOKI | |
| 119 */ | |
| 120 #else | |
| 121 /* Might need to run other tests to figure out the DIST */ | |
| 122 /* I've been told that Nvidia doesn't define constants | |
| 123 * in the headers like Creative. Instead of | |
| 124 * #define AL_REFERENCE_DISTANCE 0x1020, | |
| 125 * Nvidia prefers you query OpenAL for a value. | |
| 126 * int AL_REFERENCE_DISTANCE = alGetEnumValue(ALubyte*)"AL_REFERNECE_DISTANCE"); | |
| 127 * So I'm assuming this means the Nvidia lacks this value. | |
| 128 * If this is the case, | |
| 129 * I guess we can use it to identify the Nvidia dist | |
| 130 */ | |
| 131 #ifdef AL_REFERENCE_DISTANCE | |
| 132 #define USING_CREATIVE_AL_DIST | |
| 133 #else | |
| 134 #define USING_NVIDIA_AL_DIST | |
| 135 #endif | |
| 136 #endif | |
| 137 | |
| 138 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 139 /* Need memset to zero out data */ | |
| 140 #include <string.h> | |
| 141 #endif | |
| 142 | |
| 143 | |
| 144 /* Seek issues for predecoded samples: | |
| 145 * The problem is that OpenAL makes us copy an | |
| 146 * entire buffer if we want to use it. This | |
| 147 * means we potentially have two copies of the | |
| 148 * same data. For predecoded data, this can be a | |
| 149 * large amount of memory. However, for seek | |
| 150 * support, I need to be able to get access to | |
| 151 * the original data so I can set byte positions. | |
| 152 * The following flags let you disable seek support | |
| 153 * if you don't want the memory hit, keep everything, | |
| 154 * or let you try to minimize the memory wasted by | |
| 155 * fetching it from the OpenAL buffer if needed | |
| 156 * and making a copy of it. | |
| 157 * Update: I don't think I need this flag anymore. I've made the | |
| 158 * effects of this user customizable by the access_data flag on load. | |
| 159 * If set to true, then seek and data callbacks work, with the | |
| 160 * cost of more memory and possibly CPU for copying the data through | |
| 161 * the callbacks. If false, then the extra memory is freed, but | |
| 162 * you don't get the features. | |
| 163 */ | |
| 164 /* | |
| 165 #define DISABLE_PREDECODED_SEEK | |
| 166 */ | |
| 167 /* Problem: Even though alGetBufferi(., AL_DATA, .) | |
| 168 * is in the Creative Programmer's reference, | |
| 169 * it actually isn't in the dist. (Invalid enum | |
| 170 * in Creative, can't compile in Loki.) | |
| 171 * So we have to keep it disabled | |
| 172 */ | |
| 173 #define DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 174 | |
| 175 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 176 /* Needed for memcpy */ | |
| 177 #include <string.h> | |
| 178 #endif | |
| 179 | |
| 180 /* Old way of doing things: | |
| 181 #if defined(_WIN32) || defined(__APPLE__) | |
| 182 #define USING_CREATIVE_AL_DIST | |
| 183 #else | |
| 184 #define USING_LOKI_AL_DIST | |
| 185 #endif | |
| 186 */ | |
| 187 | |
| 188 /************ REMOVE ME (Don't need anymore) ********/ | |
| 189 #if 0 | |
| 190 /* Let's get fancy and see if triple buffering | |
| 191 * does anything good for us | |
| 192 * Must be 2 or more or things will probably break | |
| 193 */ | |
| 194 #define NUMBER_OF_QUEUE_BUFFERS 5 | |
| 195 /* This is the number of buffers that are queued up | |
| 196 * when play first starts up. This should be at least 1 | |
| 197 * and no more than NUMBER_OF_QUEUE_BUFFERS | |
| 198 */ | |
| 199 #define NUMBER_OF_START_UP_BUFFERS 2 | |
| 200 #endif | |
| 201 /************ END REMOVE ME (Don't need anymore) ********/ | |
| 202 | |
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203 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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204 #include "tErrorLib.h" |
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205 static TErrorPool* s_ALmixerErrorPool = NULL; |
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206 #endif |
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207 |
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208 static ALboolean ALmixer_Initialized = 0; |
| 0 | 209 /* This should be set correctly by Init */ |
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210 static ALuint ALmixer_Frequency_global = ALMIXER_DEFAULT_FREQUENCY; |
| 0 | 211 |
| 212 /* Will be initialized in Init */ | |
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213 static ALint Number_of_Channels_global = 0; |
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214 static ALint Number_of_Reserve_Channels_global = 0; |
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215 static ALuint Is_Playing_global = 0; |
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216 |
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217 #ifdef ENABLE_ALMIXER_THREADS |
| 0 | 218 /* This is for a simple lock system. It is not meant to be good, |
| 219 * but just sufficient to minimize/avoid threading issues | |
| 220 */ | |
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221 static SDL_mutex* s_simpleLock; |
| 0 | 222 static SDL_Thread* Stream_Thread_global = NULL; |
| 223 #endif | |
| 224 | |
| 225 | |
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226 #ifdef __APPLE__ |
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227 static ALvoid Internal_alcMacOSXMixerOutputRate(const ALdouble sample_rate) |
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228 { |
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229 static void (*alcMacOSXMixerOutputRateProcPtr)(const ALdouble) = NULL; |
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230 |
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231 if(NULL == alcMacOSXMixerOutputRateProcPtr) |
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232 { |
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233 alcMacOSXMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXMixerOutputRate"); |
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234 } |
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235 |
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236 if(NULL != alcMacOSXMixerOutputRateProcPtr) |
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237 { |
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238 alcMacOSXMixerOutputRateProcPtr(sample_rate); |
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239 } |
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240 |
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241 return; |
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242 } |
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243 |
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244 ALdouble Internal_alcMacOSXGetMixerOutputRate() |
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245 { |
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246 static ALdouble (*alcMacOSXGetMixerOutputRateProcPtr)(void) = NULL; |
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247 |
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248 if(NULL == alcMacOSXGetMixerOutputRateProcPtr) |
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249 { |
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250 alcMacOSXGetMixerOutputRateProcPtr = alGetProcAddress((const ALCchar*) "alcMacOSXGetMixerOutputRate"); |
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251 } |
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252 |
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253 if(NULL != alcMacOSXGetMixerOutputRateProcPtr) |
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254 { |
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255 return alcMacOSXGetMixerOutputRateProcPtr(); |
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256 } |
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257 |
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258 return 0.0; |
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259 } |
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260 #endif |
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261 |
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262 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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263 |
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264 #if defined(__APPLE__) |
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265 #include <QuartzCore/QuartzCore.h> |
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266 #include <unistd.h> |
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267 static CFTimeInterval s_ticksBaseTime = 0.0; |
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268 |
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269 #elif defined(_WIN32) |
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270 #define WIN32_LEAN_AND_MEAN |
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271 #include <windows.h> |
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272 #include <winbase.h> |
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273 LARGE_INTEGER s_hiResTicksPerSecond; |
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274 double s_hiResSecondsPerTick; |
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275 LARGE_INTEGER s_ticksBaseTime; |
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276 #else |
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277 #include <unistd.h> |
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278 #include <time.h> |
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279 static struct timespec s_ticksBaseTime; |
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280 #endif |
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281 static void ALmixer_InitTime() |
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282 { |
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283 #if defined(__APPLE__) |
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284 s_ticksBaseTime = CACurrentMediaTime(); |
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285 |
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286 #elif defined(_WIN32) |
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287 LARGE_INTEGER hi_res_ticks_per_second; |
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288 if(TRUE == QueryPerformanceFrequency(&hi_res_ticks_per_second)) |
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289 { |
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290 QueryPerformanceCounter(&s_ticksBaseTime); |
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291 s_hiResSecondsPerTick = 1.0 / hi_res_ticks_per_second; |
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292 } |
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293 else |
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294 { |
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295 ALMixer_SetError("Windows error: High resolution clock failed."); |
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296 fprintf(stderr, "Windows error: High resolution clock failed. Audio will not work correctly.\n"); |
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297 } |
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298 #else |
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299 /* clock_gettime is POSIX.1-2001 */ |
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300 clock_gettime(CLOCK_MONOTONIC, &s_ticksBaseTime); |
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301 #endif |
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302 |
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303 } |
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304 static ALuint ALmixer_GetTicks() |
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305 { |
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306 #if defined(__APPLE__) |
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307 return (ALuint)((CACurrentMediaTime()-s_ticksBaseTime)*1000.0); |
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308 #elif defined(_WIN32) |
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309 LARGE_INTEGER current_time; |
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310 QueryPerformanceCounter(¤t_time); |
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311 return (ALuint)((current_time.QuadPart - s_ticksBaseTime.QuadPart) * 1000 * s_hiResSecondsPerTick); |
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312 |
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313 #else /* assuming POSIX */ |
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314 /* clock_gettime is POSIX.1-2001 */ |
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315 struct timespec current_time; |
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316 clock_gettime(CLOCK_MONOTONIC, ¤t_time); |
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317 return (ALuint)((current_time.tv_sec - s_ticksBaseTime.tv_sec)*1000.0 + (current_time.tv_nec - s_ticksBaseTime.tv_nsec) / 1000000); |
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318 #endif |
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319 } |
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320 static void ALmixer_Delay(ALuint milliseconds_delay) |
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321 { |
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322 #if defined(_WIN32) |
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323 Sleep(milliseconds_delay); |
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324 #else |
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325 usleep(milliseconds_delay); |
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326 #endif |
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327 } |
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328 #else |
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329 #include "SDL.h" /* For SDL_GetTicks(), SDL_Delay */ |
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330 #define ALmixer_GetTicks SDL_GetTicks |
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331 #define ALmixer_Delay SDL_Delay |
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332 #endif |
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333 |
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334 |
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335 |
| 0 | 336 /* If ENABLE_PARANOID_SIGNEDNESS_CHECK is used, |
| 337 * these values will be reset on Init() | |
| 338 * Consider these values Read-Only. | |
| 339 */ | |
| 340 | |
| 341 #define ALMIXER_SIGNED_VALUE 127 | |
| 342 #define ALMIXER_UNSIGNED_VALUE 255 | |
| 343 | |
| 344 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
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345 static ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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346 static ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
| 0 | 347 #else |
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348 static const ALushort SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; |
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349 static const ALushort SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; |
| 0 | 350 #endif |
| 351 | |
| 1 | 352 |
| 353 /* This can be private instead of being in the header now that I moved | |
| 354 * ALmixer_Data inside here. | |
| 355 */ | |
| 356 typedef struct ALmixer_Buffer_Map ALmixer_Buffer_Map; | |
| 357 | |
| 358 | |
| 359 struct ALmixer_Data | |
| 360 { | |
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361 ALboolean decoded_all; /* dictates different behaviors */ |
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362 ALint total_time; /* total playing time of sample (msec) */ |
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363 |
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364 ALuint in_use; /* needed to prevent sharing for streams */ |
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365 ALboolean eof; /* flag for eof, only used for streams */ |
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366 |
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367 ALuint total_bytes; /* For predecoded */ |
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368 ALuint loaded_bytes; /* For predecoded (for seek) */ |
| 1 | 369 |
| 370 Sound_Sample* sample; /* SDL_Sound provides the data */ | |
| 371 ALuint* buffer; /* array of OpenAL buffers (at least 1 for predecoded) */ | |
| 372 | |
| 373 /* Needed for streamed buffers */ | |
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374 ALuint max_queue_buffers; /* Max number of queue buffers */ |
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375 ALuint num_startup_buffers; /* Number of ramp-up buffers */ |
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376 ALuint num_buffers_in_use; /* number of buffers in use */ |
| 1 | 377 |
| 378 /* This stuff is for streamed buffers that require data access */ | |
| 379 ALmixer_Buffer_Map* buffer_map_list; /* translate ALbuffer to index | |
| 380 and holds pointer to copy of data for | |
| 381 data access */ | |
| 382 ALuint current_buffer; /* The current playing buffer */ | |
| 383 | |
| 384 /* Nvidia distribution refuses to recognize a simple buffer query command | |
| 385 * unlike all other distributions. It's forcing me to redo the code | |
| 386 * to accomodate this Nvidia flaw by making me maintain a "best guess" | |
| 387 * copy of what I think the buffer queue state looks like. | |
| 388 * A circular queue would a helpful data structure for this task, | |
| 389 * but I wanted to avoid making an additional header requirement, | |
| 390 * so I'm making it a void* | |
| 391 */ | |
| 392 void* circular_buffer_queue; | |
| 393 | |
| 394 | |
| 395 }; | |
| 396 | |
| 0 | 397 static struct ALmixer_Channel |
| 398 { | |
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399 ALboolean channel_in_use; |
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400 ALboolean callback_update; /* For streaming determination */ |
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401 ALboolean needs_stream; /* For streaming determination */ |
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402 ALboolean halted; |
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403 ALboolean paused; |
| 0 | 404 ALuint alsource; |
| 405 ALmixer_Data* almixer_data; | |
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406 ALint loops; |
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407 ALint expire_ticks; |
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408 ALuint start_time; |
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409 |
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410 ALboolean fade_enabled; |
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411 ALuint fade_expire_ticks; |
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412 ALuint fade_start_time; |
| 0 | 413 ALfloat fade_inv_time; |
| 414 ALfloat fade_start_volume; | |
| 415 ALfloat fade_end_volume; | |
| 416 ALfloat max_volume; | |
| 417 ALfloat min_volume; | |
| 418 | |
| 419 /* Do we need other flags? | |
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420 ALbyte *samples; |
| 0 | 421 int volume; |
| 422 int looping; | |
| 423 int tag; | |
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424 ALuint expire; |
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425 ALuint start_time; |
| 0 | 426 Mix_Fading fading; |
| 427 int fade_volume; | |
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428 ALuint fade_length; |
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429 ALuint ticks_fade; |
| 0 | 430 effect_info *effects; |
| 431 */ | |
| 432 } *ALmixer_Channel_List = NULL; | |
| 433 | |
| 1 | 434 struct ALmixer_Buffer_Map |
| 435 { | |
| 436 ALuint albuffer; | |
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437 ALint index; /* might not need */ |
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438 ALbyte* data; |
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439 ALuint num_bytes; |
| 1 | 440 }; |
| 441 | |
| 0 | 442 /* This will be used to find a channel if the user supplies a source */ |
| 443 typedef struct Source_Map | |
| 444 { | |
| 445 ALuint source; | |
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446 ALint channel; |
| 0 | 447 } Source_Map; |
| 448 /* Keep an array of all sources with their associated channel */ | |
| 449 static Source_Map* Source_Map_List; | |
| 450 | |
| 451 static int Compare_Source_Map(const void* a, const void* b) | |
| 452 { | |
| 453 return ( ((Source_Map*)a)->source - ((Source_Map*)b)->source ); | |
| 454 } | |
| 455 | |
| 456 /* Sort by channel instead of source */ | |
| 457 static int Compare_Source_Map_by_channel(const void* a, const void* b) | |
| 458 { | |
| 459 return ( ((Source_Map*)a)->channel - ((Source_Map*)b)->channel ); | |
| 460 } | |
| 461 | |
| 462 /* Compare by albuffer */ | |
| 463 static int Compare_Buffer_Map(const void* a, const void* b) | |
| 464 { | |
| 1 | 465 return ( ((ALmixer_Buffer_Map*)a)->albuffer - ((ALmixer_Buffer_Map*)b)->albuffer ); |
| 0 | 466 } |
| 467 | |
| 468 /* This is for the user defined callback via | |
| 469 * ALmixer_ChannelFinished() | |
| 470 */ | |
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471 static void (*Channel_Done_Callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data) = NULL; |
| 0 | 472 static void* Channel_Done_Callback_Userdata = NULL; |
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473 static void (*Channel_Data_Callback)(ALint which_channel, ALuint al_source, ALbyte* data, ALuint num_bytes, ALuint frequency, ALubyte channels, ALubyte bit_depth, ALboolean is_unsigned, ALboolean decode_mode_is_predecoded, ALuint length_in_msec, void* user_data) = NULL; |
| 1 | 474 static void* Channel_Data_Callback_Userdata = NULL; |
| 0 | 475 |
| 476 | |
| 477 static void PrintQueueStatus(ALuint source) | |
| 478 { | |
| 479 ALint buffers_queued = 0; | |
| 480 ALint buffers_processed = 0; | |
| 481 ALenum error; | |
| 482 | |
| 483 /* Get the number of buffers still queued */ | |
| 484 alGetSourcei( | |
| 485 source, | |
| 486 AL_BUFFERS_QUEUED, | |
| 487 &buffers_queued | |
| 488 ); | |
| 489 | |
| 490 if((error = alGetError()) != AL_NO_ERROR) | |
| 491 { | |
| 492 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", | |
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493 alGetString(error)); |
| 0 | 494 } |
| 495 /* Get the number of buffers processed | |
| 496 * so we know if we need to refill | |
| 497 */ | |
| 498 alGetSourcei( | |
| 499 source, | |
| 500 AL_BUFFERS_PROCESSED, | |
| 501 &buffers_processed | |
| 502 ); | |
| 503 if((error = alGetError()) != AL_NO_ERROR) | |
| 504 { | |
| 505 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_processed: %s\n", | |
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506 alGetString(error)); |
| 0 | 507 } |
| 508 | |
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509 /* |
| 0 | 510 fprintf(stderr, "For source: %d, buffers_queued=%d, buffers_processed=%d\n", |
| 511 source, | |
| 512 buffers_queued, | |
| 513 buffers_processed); | |
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514 */ |
| 0 | 515 } |
| 516 | |
| 517 | |
| 518 | |
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519 static void Init_Channel(ALint channel) |
| 0 | 520 { |
| 521 ALmixer_Channel_List[channel].channel_in_use = 0; | |
| 522 ALmixer_Channel_List[channel].callback_update = 0; | |
| 523 ALmixer_Channel_List[channel].needs_stream = 0; | |
| 524 ALmixer_Channel_List[channel].paused = 0; | |
| 525 ALmixer_Channel_List[channel].halted = 0; | |
| 526 ALmixer_Channel_List[channel].loops = 0; | |
| 527 | |
| 528 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 529 ALmixer_Channel_List[channel].start_time = 0; | |
| 530 | |
| 531 ALmixer_Channel_List[channel].fade_enabled = 0; | |
| 532 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
| 533 ALmixer_Channel_List[channel].fade_start_time = 0; | |
| 534 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
| 535 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
| 536 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
| 537 ALmixer_Channel_List[channel].max_volume = 1.0f; | |
| 538 ALmixer_Channel_List[channel].min_volume = 0.0f; | |
| 539 | |
| 540 ALmixer_Channel_List[channel].almixer_data = NULL; | |
| 541 } | |
| 542 /* Quick helper function to clean up a channel | |
| 543 * after it's done playing */ | |
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544 static void Clean_Channel(ALint channel) |
| 0 | 545 { |
| 546 ALenum error; | |
| 547 ALmixer_Channel_List[channel].channel_in_use = 0; | |
| 548 ALmixer_Channel_List[channel].callback_update = 0; | |
| 549 ALmixer_Channel_List[channel].needs_stream = 0; | |
| 550 ALmixer_Channel_List[channel].paused = 0; | |
| 551 ALmixer_Channel_List[channel].halted = 0; | |
| 552 ALmixer_Channel_List[channel].loops = 0; | |
| 553 | |
| 554 | |
| 555 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 556 ALmixer_Channel_List[channel].start_time = 0; | |
| 557 | |
| 558 ALmixer_Channel_List[channel].fade_enabled = 0; | |
| 559 ALmixer_Channel_List[channel].fade_expire_ticks = 0; | |
| 560 ALmixer_Channel_List[channel].fade_start_time = 0; | |
| 561 ALmixer_Channel_List[channel].fade_inv_time = 0.0f; | |
| 562 ALmixer_Channel_List[channel].fade_start_volume = 0.0f; | |
| 563 ALmixer_Channel_List[channel].fade_end_volume = 0.0f; | |
| 564 | |
| 565 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MAX_GAIN, | |
| 566 ALmixer_Channel_List[channel].max_volume); | |
| 567 | |
| 568 if((error = alGetError()) != AL_NO_ERROR) | |
| 569 { | |
| 570 fprintf(stderr, "10Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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571 alGetString(error)); |
| 0 | 572 } |
| 573 | |
| 574 alSourcef(ALmixer_Channel_List[channel].alsource, AL_MIN_GAIN, | |
| 575 ALmixer_Channel_List[channel].min_volume); | |
| 576 if((error = alGetError()) != AL_NO_ERROR) | |
| 577 { | |
| 578 fprintf(stderr, "11Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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579 alGetString(error)); |
| 0 | 580 } |
| 581 | |
| 582 if(ALmixer_Channel_List[channel].almixer_data != NULL) | |
| 583 { | |
| 584 if(ALmixer_Channel_List[channel].almixer_data->in_use > 0) | |
| 585 { | |
| 586 ALmixer_Channel_List[channel].almixer_data->in_use--; | |
| 587 } | |
| 588 } | |
| 589 /* Needed to determine if rewind is needed, can't reset */ | |
| 590 /* | |
| 591 ALmixer_Channel_List[channel].almixer_data->eof = 0; | |
| 592 */ | |
| 593 | |
| 594 ALmixer_Channel_List[channel].almixer_data = NULL; | |
| 595 } | |
| 596 | |
| 597 | |
| 598 #if 0 | |
| 599 /* Not needed anymore because not doing any fileext checks. | |
| 600 * | |
| 601 * Unfortunately, strcasecmp isn't portable so here's a | |
| 602 * reimplementation of it (taken from SDL_sound) | |
| 603 */ | |
| 604 static int ALmixer_strcasecmp(const char* x, const char* y) | |
| 605 { | |
| 606 int ux, uy; | |
| 607 | |
| 608 if (x == y) /* same pointer? Both NULL? */ | |
| 609 return(0); | |
| 610 | |
| 611 if (x == NULL) | |
| 612 return(-1); | |
| 613 | |
| 614 if (y == NULL) | |
| 615 return(1); | |
| 616 | |
| 617 do | |
| 618 { | |
| 619 ux = toupper((int) *x); | |
| 620 uy = toupper((int) *y); | |
| 621 if (ux > uy) | |
| 622 return(1); | |
| 623 else if (ux < uy) | |
| 624 return(-1); | |
| 625 x++; | |
| 626 y++; | |
| 627 } while ((ux) && (uy)); | |
| 628 | |
| 629 return(0); | |
| 630 } | |
| 631 #endif | |
| 632 | |
| 633 | |
| 634 /* What shoud this return? | |
| 635 * 127 for signed, 255 for unsigned | |
| 636 */ | |
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637 static ALubyte GetSignednessValue(ALushort format) |
| 0 | 638 { |
| 639 switch(format) | |
| 640 { | |
| 641 case AUDIO_U8: | |
| 642 case AUDIO_U16LSB: | |
| 643 case AUDIO_U16MSB: | |
| 644 return ALMIXER_UNSIGNED_VALUE; | |
| 645 break; | |
| 646 case AUDIO_S8: | |
| 647 case AUDIO_S16LSB: | |
| 648 case AUDIO_S16MSB: | |
| 649 return ALMIXER_SIGNED_VALUE; | |
| 650 break; | |
| 651 default: | |
| 652 return 0; | |
| 653 } | |
| 654 return 0; | |
| 655 } | |
| 656 | |
| 657 | |
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658 static ALubyte GetBitDepth(ALushort format) |
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659 { |
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660 ALubyte bit_depth = 16; |
| 0 | 661 |
| 662 switch(format) | |
| 663 { | |
| 664 case AUDIO_U8: | |
| 665 case AUDIO_S8: | |
| 666 bit_depth = 8; | |
| 667 break; | |
| 668 | |
| 669 case AUDIO_U16LSB: | |
| 670 /* | |
| 671 case AUDIO_U16: | |
| 672 */ | |
| 673 case AUDIO_S16LSB: | |
| 674 /* | |
| 675 case AUDIO_S16: | |
| 676 */ | |
| 677 case AUDIO_U16MSB: | |
| 678 case AUDIO_S16MSB: | |
| 679 /* | |
| 680 case AUDIO_U16SYS: | |
| 681 case AUDIO_S16SYS: | |
| 682 */ | |
| 683 bit_depth = 16; | |
| 684 break; | |
| 685 default: | |
| 686 bit_depth = 0; | |
| 687 } | |
| 688 return bit_depth; | |
| 689 } | |
| 690 | |
| 691 /* Need to translate between SDL/SDL_Sound audiospec | |
| 692 * and OpenAL conventions */ | |
| 693 static ALenum TranslateFormat(Sound_AudioInfo* info) | |
| 694 { | |
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695 ALubyte bit_depth; |
| 0 | 696 |
| 697 bit_depth = GetBitDepth(info->format); | |
| 698 if(0 == bit_depth) | |
| 699 { | |
| 700 fprintf(stderr, "Warning: Unknown bit depth. Setting to 16\n"); | |
| 701 bit_depth = 16; | |
| 702 } | |
| 703 | |
| 704 if(2 == info->channels) | |
| 705 { | |
| 706 if(16 == bit_depth) | |
| 707 { | |
| 708 return AL_FORMAT_STEREO16; | |
| 709 } | |
| 710 else | |
| 711 { | |
| 712 return AL_FORMAT_STEREO8; | |
| 713 } | |
| 714 } | |
| 715 else | |
| 716 { | |
| 717 if(16 == bit_depth) | |
| 718 { | |
| 719 return AL_FORMAT_MONO16; | |
| 720 } | |
| 721 else | |
| 722 { | |
| 723 return AL_FORMAT_MONO8; | |
| 724 } | |
| 725 } | |
| 726 /* Make compiler happy. Shouldn't get here */ | |
| 727 return AL_FORMAT_STEREO16; | |
| 728 } | |
| 729 | |
| 1 | 730 |
| 731 /* This will compute the total playing time | |
| 732 * based upon the number of bytes and audio info. | |
| 733 * (In prinicple, it should compute the time for any given length) | |
| 734 */ | |
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735 static ALuint Compute_Total_Time_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, size_t total_bytes) |
| 1 | 736 { |
| 737 double total_sec; | |
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738 ALuint total_msec; |
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739 ALuint bytes_per_sec; |
| 1 | 740 |
| 741 if(0 == total_bytes) | |
| 742 { | |
| 743 return 0; | |
| 744 } | |
| 745 /* To compute Bytes per second, do | |
| 746 * samples_per_sec * bytes_per_sample * number_of_channels | |
| 747 */ | |
| 748 bytes_per_sec = frequency * bytes_per_sample * channels; | |
| 749 | |
| 750 /* Now to get total time (sec), do | |
| 751 * total_bytes / bytes_per_sec | |
| 752 */ | |
| 753 total_sec = total_bytes / (double)bytes_per_sec; | |
| 754 | |
| 755 /* Now convert seconds to milliseconds | |
| 756 * Add .5 to the float to do rounding before the final cast | |
| 757 */ | |
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758 total_msec = (ALuint) ( (total_sec * 1000) + 0.5 ); |
| 1 | 759 /* |
| 760 fprintf(stderr, "freq=%d, bytes_per_sample=%d, channels=%d, total_msec=%d\n", frequency, bytes_per_sample, channels, total_msec); | |
| 761 */ | |
| 762 return total_msec; | |
| 763 } | |
| 764 | |
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765 static ALuint Compute_Total_Time(Sound_AudioInfo *info, size_t total_bytes) |
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766 { |
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767 ALuint bytes_per_sample; |
| 1 | 768 |
| 769 if(0 == total_bytes) | |
| 770 { | |
| 771 return 0; | |
| 772 } | |
| 773 /* SDL has a mask trick I was not aware of. Mask the upper bits | |
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774 * of the format, and you get 8 or 16 which is the bits per sample. |
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775 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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776 * I tested this under 32-bit and 64-bit and big and little endian |
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777 * to make sure this still works since I have since moved from |
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778 * Uint32 to unspecified size types like ALuint. |
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779 */ |
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780 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
| 1 | 781 |
| 782 return Compute_Total_Time_Decomposed(bytes_per_sample, info->rate, info->channels, total_bytes); | |
| 783 } /* End Compute_Total_Time */ | |
| 784 | |
| 785 | |
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786 static size_t Compute_Total_Bytes_Decomposed(ALuint bytes_per_sample, ALuint frequency, ALubyte channels, ALuint total_msec) |
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787 { |
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788 double total_sec; |
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789 ALuint bytes_per_sec; |
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790 size_t total_bytes; |
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791 |
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792 if(0 >= total_msec) |
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793 { |
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794 return 0; |
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795 } |
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796 /* To compute Bytes per second, do |
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797 * samples_per_sec * bytes_per_sample * number_of_channels |
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798 */ |
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799 bytes_per_sec = frequency * bytes_per_sample * channels; |
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800 |
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801 /* convert milliseconds to seconds */ |
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802 total_sec = total_msec / 1000.0; |
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803 |
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804 /* Now to get total bytes */ |
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805 total_bytes = (size_t)(((double)bytes_per_sec * total_sec) + 0.5); |
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806 |
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807 /* fprintf(stderr, "freq=%d, bytes_per_sample=%d, channels=%d, total_msec=%d, total_bytes=%d\n", frequency, bytes_per_sample, channels, total_msec, total_bytes); |
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808 */ |
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809 |
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810 return total_bytes; |
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811 } |
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812 |
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813 static size_t Compute_Total_Bytes(Sound_AudioInfo *info, ALuint total_msec) |
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814 { |
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815 ALuint bytes_per_sample; |
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816 |
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817 if(0 >= total_msec) |
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818 { |
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819 return 0; |
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820 } |
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821 /* SDL has a mask trick I was not aware of. Mask the upper bits |
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822 * of the format, and you get 8 or 16 which is the bits per sample. |
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823 * Divide by 8bits_per_bytes and you get bytes_per_sample |
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824 * I tested this under 32-bit and 64-bit and big and little endian |
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825 * to make sure this still works since I have since moved from |
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826 * Uint32 to unspecified size types like ALuint. |
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827 */ |
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828 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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829 |
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830 return Compute_Total_Bytes_Decomposed(bytes_per_sample, info->rate, info->channels, total_msec); |
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831 } |
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832 |
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833 /* The back-end decoders seem to need to decode in quantized frame sizes. |
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834 * So if I can pad the bytes to the next quanta, things might go more smoothly. |
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835 */ |
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836 static size_t Compute_Total_Bytes_With_Frame_Padding(Sound_AudioInfo *info, ALuint total_msec) |
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837 { |
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838 ALuint bytes_per_sample; |
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839 ALuint bytes_per_frame; |
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840 size_t evenly_divisible_frames; |
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841 size_t remainder_frames; |
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842 size_t return_bytes; |
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843 |
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844 size_t total_bytes = Compute_Total_Bytes(info, total_msec); |
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845 |
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846 bytes_per_sample = (ALuint) ((info->format & 0xFF) / 8); |
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847 |
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848 bytes_per_frame = bytes_per_sample * info->channels; |
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849 |
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850 evenly_divisible_frames = total_bytes / bytes_per_frame; |
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851 remainder_frames = total_bytes % bytes_per_frame; |
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852 |
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853 return_bytes = (evenly_divisible_frames * bytes_per_frame) + (remainder_frames * bytes_per_frame); |
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854 |
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855 /* Experimentally, some times I see to come up short in |
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856 * actual bytes decoded and I see a second pass is needed. |
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857 * I'm worried this may have additional performance implications. |
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858 * Sometimes in the second pass (depending on file), |
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859 * I have seen between 0 and 18 bytes. |
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860 * I'm tempted to pad the bytes by some arbitrary amount. |
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861 * However, I think currently the way SDL_sound is implemented, |
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862 * there is a big waste of memory up front instead of per-pass, |
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863 * so maybe I shouldn't worry about this. |
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864 */ |
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865 /* |
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866 return_bytes += 64; |
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867 */ |
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868 /* |
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869 fprintf(stderr, "remainder_frames=%d, padded_total_bytes=%d\n", remainder_frames, return_bytes); |
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870 */ |
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871 return return_bytes; |
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872 |
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873 } |
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874 |
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875 |
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876 |
| 1 | 877 |
| 0 | 878 /**************** REMOVED ****************************/ |
| 879 /* This was removed because I originally thought | |
| 880 * OpenAL could return a pointer to the buffer data, | |
| 881 * but I was wrong. If something like that is ever | |
| 882 * implemented, then this might become useful. | |
| 883 */ | |
| 884 #if 0 | |
| 885 /* Reconstruct_Sound_Sample and Set_AudioInfo only | |
| 886 * are needed if the Seek memory optimization is | |
| 887 * used. Also, the Loki dist doesn't seem to support | |
| 888 * AL_DATA which I need for it. | |
| 889 */ | |
| 890 #ifndef DISABLE_SEEK_MEMORY_OPTIMIZATION | |
| 891 | |
| 892 static void Set_AudioInfo(Sound_AudioInfo* info, ALint frequency, ALint bits, ALint channels) | |
| 893 { | |
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894 info->rate = (ALuint)frequency; |
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895 info->channels = (ALubyte)channels; |
| 0 | 896 |
| 897 /* Not sure if it should be signed or unsigned. Hopefully | |
| 898 * that detail won't be needed. | |
| 899 */ | |
| 900 if(8 == bits) | |
| 901 { | |
| 902 info->format = AUDIO_U8; | |
| 903 } | |
| 904 else | |
| 905 { | |
| 906 info->format = AUDIO_U16SYS; | |
| 907 } | |
| 908 fprintf(stderr, "Audio info: freq=%d, chan=%d, format=%d\n", | |
| 909 info->rate, info->channels, info->format); | |
| 910 | |
| 911 } | |
| 912 | |
| 913 | |
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914 static ALint Reconstruct_Sound_Sample(ALmixer_Data* data) |
| 0 | 915 { |
| 916 ALenum error; | |
| 917 ALint* data_from_albuffer; | |
| 918 ALint freq; | |
| 919 ALint bits; | |
| 920 ALint channels; | |
| 921 ALint size; | |
| 922 | |
| 923 /* Create memory all initiallized to 0. */ | |
| 924 data->sample = (Sound_Sample*)calloc(1, sizeof(Sound_Sample)); | |
| 925 if(NULL == data->sample) | |
| 926 { | |
| 927 ALmixer_SetError("Out of memory for Sound_Sample"); | |
| 928 return -1; | |
| 929 } | |
| 930 | |
| 931 /* Clear errors */ | |
| 932 alGetError(); | |
| 933 | |
| 934 alGetBufferi(data->buffer[0], AL_FREQUENCY, &freq); | |
| 935 if((error = alGetError()) != AL_NO_ERROR) | |
| 936 { | |
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937 ALmixer_SetError("alGetBufferi(AL_FREQUENCY): %s", alGetString(error) ); |
| 0 | 938 free(data->sample); |
| 939 data->sample = NULL; | |
| 940 return -1; | |
| 941 } | |
| 942 | |
| 943 alGetBufferi(data->buffer[0], AL_BITS, &bits); | |
| 944 if((error = alGetError()) != AL_NO_ERROR) | |
| 945 { | |
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946 ALmixer_SetError("alGetBufferi(AL_BITS): %s", alGetString(error) ); |
| 0 | 947 free(data->sample); |
| 948 data->sample = NULL; | |
| 949 return -1; | |
| 950 } | |
| 951 | |
| 952 alGetBufferi(data->buffer[0], AL_CHANNELS, &channels); | |
| 953 if((error = alGetError()) != AL_NO_ERROR) | |
| 954 { | |
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955 ALmixer_SetError("alGetBufferi(AL_CHANNELS): %s", alGetString(error) ); |
| 0 | 956 free(data->sample); |
| 957 data->sample = NULL; | |
| 958 return -1; | |
| 959 } | |
| 960 | |
| 961 alGetBufferi(data->buffer[0], AL_SIZE, &size); | |
| 962 if((error = alGetError()) != AL_NO_ERROR) | |
| 963 { | |
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964 ALmixer_SetError("alGetBufferi(AL_SIZE): %s", alGetString(error) ); |
| 0 | 965 free(data->sample); |
| 966 data->sample = NULL; | |
| 967 return -1; | |
| 968 } | |
| 969 | |
| 970 alGetBufferi(data->buffer[0], AL_DATA, data_from_albuffer); | |
| 971 if((error = alGetError()) != AL_NO_ERROR) | |
| 972 { | |
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973 ALmixer_SetError("alGetBufferi(AL_DATA): %s", alGetString(error) ); |
| 0 | 974 free(data->sample); |
| 975 data->sample = NULL; | |
| 976 return -1; | |
| 977 } | |
| 978 | |
| 979 if(size <= 0) | |
| 980 { | |
| 981 ALmixer_SetError("No data in al buffer"); | |
| 982 free(data->sample); | |
| 983 data->sample = NULL; | |
| 984 return -1; | |
| 985 } | |
| 986 | |
| 987 /* Now that we have all the attributes, we need to | |
| 988 * allocate memory for the buffer and reconstruct | |
| 989 * the AudioInfo attributes. | |
| 990 */ | |
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991 data->sample->buffer = malloc(size*sizeof(ALbyte)); |
| 0 | 992 if(NULL == data->sample->buffer) |
| 993 { | |
| 994 ALmixer_SetError("Out of memory for sample->buffer"); | |
| 995 free(data->sample); | |
| 996 data->sample = NULL; | |
| 997 return -1; | |
| 998 } | |
| 999 | |
| 1000 memcpy(data->sample->buffer, data_from_albuffer, size); | |
| 1001 data->sample->buffer_size = size; | |
| 1002 | |
| 1003 /* Fill up the Sound_AudioInfo structures */ | |
| 1004 Set_AudioInfo(&data->sample->desired, freq, bits, channels); | |
| 1005 Set_AudioInfo(&data->sample->actual, freq, bits, channels); | |
| 1006 | |
| 1007 return 0; | |
| 1008 } | |
| 1009 | |
| 1010 #endif /* End DISABLE_SEEK_MEMORY_OPTIMIZATION */ | |
| 1011 #endif | |
| 1012 /*************** END REMOVED *************************/ | |
| 1013 | |
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1014 static void Invoke_Channel_Done_Callback(ALint which_channel, ALboolean did_finish_naturally) |
| 0 | 1015 { |
| 1016 if(NULL == Channel_Done_Callback) | |
| 1017 { | |
| 1018 return; | |
| 1019 } | |
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1020 Channel_Done_Callback(which_channel, ALmixer_Channel_List[which_channel].alsource, ALmixer_Channel_List[which_channel].almixer_data, did_finish_naturally, Channel_Done_Callback_Userdata); |
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1021 } |
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1022 |
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1023 static ALint LookUpBuffer(ALuint buffer, ALmixer_Buffer_Map* buffer_map_list, ALuint num_items_in_list) |
| 0 | 1024 { |
| 1025 /* Only the first value is used for the key */ | |
| 1 | 1026 ALmixer_Buffer_Map key = { 0, 0, NULL, 0 }; |
| 1027 ALmixer_Buffer_Map* found_item = NULL; | |
| 0 | 1028 key.albuffer = buffer; |
| 1029 | |
| 1030 /* Use the ANSI C binary search feature (yea!) */ | |
| 1 | 1031 found_item = (ALmixer_Buffer_Map*)bsearch(&key, buffer_map_list, num_items_in_list, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
| 0 | 1032 if(NULL == found_item) |
| 1033 { | |
| 1034 ALmixer_SetError("Can't find buffer"); | |
| 1035 return -1; | |
| 1036 } | |
| 1037 return found_item->index; | |
| 1038 } | |
| 1039 | |
| 1040 | |
| 1041 /* FIXME: Need to pass back additional info to be useful. | |
| 1042 * Bit rate, stereo/mono (num chans), time in msec? | |
| 1043 * Precoded/streamed flag so user can plan for future data? | |
| 1044 */ | |
| 1 | 1045 /* |
| 1046 * channels: 1 for mono, 2 for stereo | |
| 1047 * | |
| 1048 */ | |
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1049 static void Invoke_Channel_Data_Callback(ALint which_channel, ALbyte* data, ALuint num_bytes, ALuint frequency, ALubyte channels, ALushort format, ALboolean decode_mode_is_predecoded) |
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1050 { |
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1051 ALboolean is_unsigned; |
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1052 ALubyte bits_per_sample = GetBitDepth(format); |
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1053 ALuint bytes_per_sample; |
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1054 ALuint length_in_msec; |
| 1 | 1055 |
| 1056 if(GetSignednessValue(format) == ALMIXER_UNSIGNED_VALUE) | |
| 1057 { | |
| 1058 is_unsigned = 1; | |
| 1059 } | |
| 1060 else | |
| 1061 { | |
| 1062 is_unsigned = 0; | |
| 1063 } | |
| 1064 | |
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1065 bytes_per_sample = (ALuint) (bits_per_sample / 8); |
| 1 | 1066 |
| 1067 length_in_msec = Compute_Total_Time_Decomposed(bytes_per_sample, frequency, channels, num_bytes); | |
| 1068 | |
| 0 | 1069 /* |
| 1070 fprintf(stderr, "%x %x %x %x, bytes=%d, whichchan=%d, freq=%d, channels=%d\n", data[0], data[1], data[2], data[3], num_bytes, channels, frequency, channels); | |
| 1071 */ | |
| 1072 if(NULL == Channel_Data_Callback) | |
| 1073 { | |
| 1074 return; | |
| 1075 } | |
| 1 | 1076 /* |
| 1077 * Channel_Data_Callback(which_channel, data, num_bytes, frequency, channels, GetBitDepth(format), format, decode_mode_is_predecoded); | |
| 1078 */ | |
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1079 Channel_Data_Callback(which_channel, ALmixer_Channel_List[which_channel].alsource, data, num_bytes, frequency, channels, bits_per_sample, is_unsigned, decode_mode_is_predecoded, length_in_msec, Channel_Data_Callback_Userdata); |
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1080 } |
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1081 |
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1082 static void Invoke_Predecoded_Channel_Data_Callback(ALint channel, ALmixer_Data* data) |
| 0 | 1083 { |
| 1084 if(NULL == data->sample) | |
| 1085 { | |
| 1086 return; | |
| 1087 } | |
| 1088 /* The buffer position is complicated because if the current data was seeked, | |
| 1089 * we must adjust the buffer to the seek position | |
| 1090 */ | |
| 1091 Invoke_Channel_Data_Callback(channel, | |
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1092 (((ALbyte*) data->sample->buffer) + (data->total_bytes - data->loaded_bytes) ), |
| 0 | 1093 data->loaded_bytes, |
| 1094 data->sample->desired.rate, | |
| 1095 data->sample->desired.channels, | |
| 1096 data->sample->desired.format, | |
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1097 AL_TRUE |
| 0 | 1098 ); |
| 1099 } | |
| 1100 | |
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1101 static void Invoke_Streamed_Channel_Data_Callback(ALint channel, ALmixer_Data* data, ALuint buffer) |
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1102 { |
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1103 ALint index; |
| 0 | 1104 if(NULL == data->buffer_map_list) |
| 1105 { | |
| 1106 return; | |
| 1107 } | |
| 1108 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
| 1109 /* This should catch the case where all buffers are unqueued | |
| 1110 * and the "current" buffer is id: 0 | |
| 1111 */ | |
| 1112 if(-1 == index) | |
| 1113 { | |
| 1114 return; | |
| 1115 } | |
| 1116 Invoke_Channel_Data_Callback(channel, | |
| 1117 data->buffer_map_list[index].data, | |
| 1118 data->buffer_map_list[index].num_bytes, | |
| 1119 data->sample->desired.rate, | |
| 1120 data->sample->desired.channels, | |
| 1121 data->sample->desired.format, | |
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1122 AL_FALSE |
| 0 | 1123 ); |
| 1124 } | |
| 1125 | |
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1126 /* Converts milliseconds to byte positions. |
| 0 | 1127 * This is needed for seeking on predecoded samples |
| 1128 */ | |
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1129 static ALuint Convert_Msec_To_Byte_Pos(Sound_AudioInfo *info, ALuint ms) |
| 0 | 1130 { |
| 1131 float frames_per_ms; | |
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1132 ALuint frame_offset; |
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1133 ALuint frame_size; |
| 0 | 1134 if(info == NULL) |
| 1135 { | |
| 1136 fprintf(stderr, "Error, info is NULL\n"); | |
| 1137 } | |
| 1138 | |
| 1139 /* "frames" == "sample frames" */ | |
| 1140 frames_per_ms = ((float) info->rate) / 1000.0f; | |
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1141 frame_offset = (ALuint) (frames_per_ms * ((float) ms)); |
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1142 frame_size = (ALuint) ((info->format & 0xFF) / 8) * info->channels; |
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1143 return frame_offset * frame_size; |
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1144 } |
| 0 | 1145 |
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1146 static ALint Set_Predecoded_Seek_Position(ALmixer_Data* data, ALuint byte_position) |
| 0 | 1147 { |
| 1148 ALenum error; | |
| 1149 /* clear error */ | |
| 1150 alGetError(); | |
| 1151 | |
| 1152 /* Is it greater than, or greater-than or equal to ?? */ | |
| 1153 if(byte_position > data->total_bytes) | |
| 1154 { | |
| 1155 /* We can't go past the end, so set to end? */ | |
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1156 /* |
| 0 | 1157 fprintf(stderr, "Error, can't seek past end\n"); |
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1158 */ |
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1159 |
| 0 | 1160 /* In case the below thing doesn't work, |
| 1161 * just rewind the whole thing. | |
| 1162 * | |
| 1163 alBufferData(data->buffer[0], | |
| 1164 TranslateFormat(&data->sample->desired), | |
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1165 (ALbyte*) data->sample->buffer, |
| 0 | 1166 data->total_bytes, |
| 1167 data->sample->desired.rate | |
| 1168 ); | |
| 1169 */ | |
| 1170 | |
| 1171 /* I was trying to set to the end, (1 byte remaining), | |
| 1172 * but I was getting freezes. I'm thinking it might be | |
| 1173 * another Power of 2 bug in the Loki dist. I tried 2, | |
| 1174 * and it still hung. 4 didn't hang, but I got a clip | |
| 1175 * artifact. 8 seemed to work okay. | |
| 1176 */ | |
| 1177 alBufferData(data->buffer[0], | |
| 1178 TranslateFormat(&data->sample->desired), | |
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1179 (((ALbyte*) data->sample->buffer) + (data->total_bytes - 8) ), |
| 0 | 1180 8, |
| 1181 data->sample->desired.rate | |
| 1182 ); | |
| 1183 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1184 { | |
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1185 ALmixer_SetError("Can't seek past end and alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1186 return -1; |
| 1187 } | |
| 1188 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
| 1189 * query command to work because I don't know if it will mutilate the | |
| 1190 * size for its own purposes or return the original size | |
| 1191 */ | |
| 1192 data->loaded_bytes = 8; | |
| 1193 | |
| 1194 /* Not sure if this should be an error or not */ | |
| 1195 /* | |
| 1196 ALmixer_SetError("Can't Seek past end"); | |
| 1197 return -1; | |
| 1198 */ | |
| 1199 return 0; | |
| 1200 } | |
| 1201 | |
| 1202 alBufferData(data->buffer[0], | |
| 1203 TranslateFormat(&data->sample->desired), | |
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1204 &(((ALbyte*)data->sample->buffer)[byte_position]), |
| 0 | 1205 data->total_bytes - byte_position, |
| 1206 data->sample->desired.rate | |
| 1207 ); | |
| 1208 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1209 { | |
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1210 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1211 return -1; |
| 1212 } | |
| 1213 /* Need to set the loaded_bytes field because I don't trust the OpenAL | |
| 1214 * query command to work because I don't know if it will mutilate the | |
| 1215 * size for its own purposes or return the original size | |
| 1216 */ | |
| 1217 data->loaded_bytes = data->total_bytes - byte_position; | |
| 1218 | |
| 1219 return 0; | |
| 1220 } | |
| 1221 | |
| 1222 /* Because we have multiple queue buffers and OpenAL won't let | |
| 1223 * us access them, we need to keep copies of each buffer around | |
| 1224 */ | |
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1225 static ALint CopyDataToAccessBuffer(ALmixer_Data* data, ALuint num_bytes, ALuint buffer) |
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1226 { |
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1227 ALint index; |
| 0 | 1228 /* We only want to copy if access_data is true. |
| 1229 * This is determined by whether memory has been | |
| 1230 * allocated in the buffer_map_list or not | |
| 1231 */ | |
| 1232 if(NULL == data->buffer_map_list) | |
| 1233 { | |
| 1234 return -1; | |
| 1235 } | |
| 1236 index = LookUpBuffer(buffer, data->buffer_map_list, data->max_queue_buffers); | |
| 1237 if(-1 == index) | |
| 1238 { | |
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1239 /* |
| 0 | 1240 fprintf(stderr, ">>>>>>>CopyData catch, albuffer=%d\n",buffer); |
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1241 */ |
| 0 | 1242 return -1; |
| 1243 } | |
| 1244 /* Copy the data to the access buffer */ | |
| 1245 memcpy(data->buffer_map_list[index].data, data->sample->buffer, num_bytes); | |
| 1246 data->buffer_map_list[index].num_bytes = data->sample->buffer_size; | |
| 1247 | |
| 1248 return 0; | |
| 1249 } | |
| 1250 | |
| 1251 | |
| 1252 /* For streamed data, gets more data | |
| 1253 * and prepares it in the active Mix_chunk | |
| 1254 */ | |
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1255 static ALuint GetMoreData(ALmixer_Data* data, ALuint buffer) |
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1256 { |
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1257 ALuint bytes_decoded; |
| 0 | 1258 ALenum error; |
| 1259 if(NULL == data) | |
| 1260 { | |
| 1261 ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n"); | |
| 1262 return 0; | |
| 1263 } | |
| 1264 | |
| 1265 bytes_decoded = Sound_Decode(data->sample); | |
| 1266 if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 1267 { | |
| 1268 fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n"); | |
| 1269 ALmixer_SetError(Sound_GetError()); | |
| 1270 /* Force cleanup through FreeData | |
| 1271 Sound_FreeSample(data->sample); | |
| 1272 */ | |
| 1273 return 0; | |
| 1274 } | |
| 1275 | |
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1276 /* fprintf(stderr, "GetMoreData bytes_decoded=%d\n", bytes_decoded); */ |
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1277 |
| 0 | 1278 /* Don't forget to add check for EOF */ |
| 1279 /* Will return 0 bytes and pass the buck to check sample->flags */ | |
| 1280 if(0 == bytes_decoded) | |
| 1281 { | |
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1282 data->eof = 1; |
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1283 |
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1284 #if 0 |
| 0 | 1285 fprintf(stderr, "Hit eof while trying to buffer\n"); |
| 1286 if(data->sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 1287 { | |
| 1288 fprintf(stderr, "\tEOF flag\n"); | |
| 1289 } | |
| 1290 if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK) | |
| 1291 { | |
| 1292 fprintf(stderr, "\tCanSeek flag\n"); | |
| 1293 } | |
| 1294 if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 1295 { | |
| 1296 fprintf(stderr, "\tEAGAIN flag\n"); | |
| 1297 } | |
| 1298 if(data->sample->flags & SOUND_SAMPLEFLAG_NONE) | |
| 1299 { | |
| 1300 fprintf(stderr, "\tNONE flag\n"); | |
| 1301 } | |
| 1302 #endif | |
| 1303 return 0; | |
| 1304 } | |
| 1305 | |
| 1306 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 1307 /******* REMOVE ME ********************************/ | |
| 1308 /***************** ANOTHER EXPERIEMENT *******************/ | |
| 1309 /* The PROBLEM: It seems that the Loki distribution has problems | |
| 1310 * with Queuing when the buffer size is not a power of two | |
| 1311 * and additional buffers must come after it. | |
| 1312 * The behavior is inconsistent, but one of several things | |
| 1313 * usually happens: | |
| 1314 * Playback is normal | |
| 1315 * Playback immediately stops after the non-pow2 buffer | |
| 1316 * Playback gets distorted on the non-pow2 buffer | |
| 1317 * The entire program segfaults. | |
| 1318 * The workaround is to always specify a power of two buffer size | |
| 1319 * and hope that SDL_sound always fill it. (By lucky coincidence, | |
| 1320 * I already submitted the Ogg fix.) However, this won't catch | |
| 1321 * cases where a loop happens because the read at the end of the | |
| 1322 * file is typically less than the buffer size. | |
| 1323 * | |
| 1324 * This fix addresses this issue, however it may break in | |
| 1325 * other conditions. Always decode in buffer sizes of powers of 2. | |
| 1326 * | |
| 1327 * The HACK: | |
| 1328 * If the buffer is short, try filling it up with 0's | |
| 1329 * to meet the user requested buffer_size which | |
| 1330 * is probably a nice number OpenAL likes, in | |
| 1331 * hopes to avoid a possible Loki bug with | |
| 1332 * short buffers. If looping (which is the main | |
| 1333 * reason for this), the negative side effect is | |
| 1334 * that it may take longer for the loop to start | |
| 1335 * because it must play dead silence. Or if the decoder | |
| 1336 * doesn't guarantee to return the requested bytes | |
| 1337 * (like Ogg), then you will get breakup in between | |
| 1338 * packets. | |
| 1339 */ | |
| 1340 if( (bytes_decoded) < data->sample->buffer_size) | |
| 1341 { | |
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1342 ALubyte bit_depth; |
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1343 ALubyte signedness_value; |
| 0 | 1344 int silence_value; |
| 1345 /* Crap, memset value needs to be the "silent" value, | |
| 1346 * but it will differ for signed/unsigned and bit depth | |
| 1347 */ | |
| 1348 bit_depth = GetBitDepth(data->sample->desired.format); | |
| 1349 signedness_value = GetSignednessValue(data->sample->desired.format); | |
| 1350 if(ALMIXER_SIGNED_VALUE == signedness_value) | |
| 1351 { | |
| 1352 /* I'm guessing that if it's signed, then 0 is the | |
| 1353 * "silent" value */ | |
| 1354 silence_value = 0; | |
| 1355 } | |
| 1356 else | |
| 1357 { | |
| 1358 if(8 == bit_depth) | |
| 1359 { | |
| 1360 /* If 8 bit, I'm guessing it's (2^7)-1 = 127 */ | |
| 1361 silence_value = 127; | |
| 1362 } | |
| 1363 else | |
| 1364 { | |
| 1365 /* For 16 bit, I'm guessing it's (2^15)-1 = 32767 */ | |
| 1366 silence_value = 32767; | |
| 1367 } | |
| 1368 } | |
| 1369 /* Now fill up the rest of the data buffer with the | |
| 1370 * silence_value. | |
| 1371 * I don't think I have to worry about endian issues for | |
| 1372 * this part since the data is for internal use only | |
| 1373 * at this point. | |
| 1374 */ | |
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1375 memset( &( ((ALbyte*)(data->sample->buffer))[bytes_decoded] ), silence_value, data->sample->buffer_size - bytes_decoded); |
| 0 | 1376 /* Now reset the bytes_decoded to reflect the entire |
| 1377 * buffer to tell alBufferData what our full size is. | |
| 1378 */ | |
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1379 /* |
| 0 | 1380 fprintf(stderr, "ALTERED bytes decoded for silence: Original end was %d\n", bytes_decoded); |
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1381 */ |
| 0 | 1382 bytes_decoded = data->sample->buffer_size; |
| 1383 } | |
| 1384 /*********** END EXPERIMENT ******************************/ | |
| 1385 /******* END REMOVE ME ********************************/ | |
| 1386 #endif | |
| 1387 | |
| 1388 /* Now copy the data to the OpenAL buffer */ | |
| 1389 /* We can't just set a pointer because the API needs | |
| 1390 * its own copy to assist hardware acceleration */ | |
| 1391 alBufferData(buffer, | |
| 1392 TranslateFormat(&data->sample->desired), | |
| 1393 data->sample->buffer, | |
| 1394 bytes_decoded, | |
| 1395 data->sample->desired.rate | |
| 1396 ); | |
| 1397 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1398 { | |
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1399 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1400 return 0; |
| 1401 } | |
| 1402 | |
| 1403 /* If we need to, copy the data also to the access area | |
| 1404 * (the function will do the check for us) | |
| 1405 */ | |
| 1406 CopyDataToAccessBuffer(data, bytes_decoded, buffer); | |
| 1407 return bytes_decoded; | |
| 1408 } | |
| 1409 | |
| 1410 | |
| 1411 | |
| 1412 | |
| 1413 /******************** EXPERIEMENT **************************** | |
| 1414 * Test function to force maximum buffer filling during loops | |
| 1415 * REMOVE LATER | |
| 1416 *********************************************/ | |
| 1417 #if 0 | |
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1418 static ALint GetMoreData2(ALmixer_Data* data, ALuint buffer) |
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1419 { |
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1420 ALint bytes_decoded; |
| 0 | 1421 ALenum error; |
| 1422 if(NULL == data) | |
| 1423 { | |
| 1424 ALmixer_SetError("Cannot GetMoreData() because ALmixer_Data* is NULL\n"); | |
| 1425 return -1; | |
| 1426 } | |
| 1427 | |
| 1428 if(AL_FALSE == alIsBuffer(buffer)) | |
| 1429 { | |
| 1430 fprintf(stderr, "NOT A BUFFER>>>>>>>>>>>>>>>\n"); | |
| 1431 return -1; | |
| 1432 } | |
| 1433 fprintf(stderr, "Entered GetMoreData222222: buffer id is %d\n", buffer); | |
| 1434 | |
| 1435 /* | |
| 1436 fprintf(stderr, "Decode in GetMoreData\n"); | |
| 1437 */ | |
| 1438 | |
| 1439 #if 0 | |
| 1440 if(buffer%2 == 1) | |
| 1441 { | |
| 1442 fprintf(stderr, "Setting buffer size to 16384\n"); | |
| 1443 Sound_SetBufferSize(data->sample, 16384); | |
| 1444 } | |
| 1445 else | |
| 1446 { | |
| 1447 fprintf(stderr, "Setting buffer size to 8192\n"); | |
| 1448 Sound_SetBufferSize(data->sample, 8192); | |
| 1449 } | |
| 1450 #endif | |
| 1451 | |
| 1452 bytes_decoded = Sound_Decode(data->sample); | |
| 1453 if(data->sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 1454 { | |
| 1455 fprintf(stderr, "Sound_Decode triggered an ERROR>>>>>>\n"); | |
| 1456 ALmixer_SetError(Sound_GetError()); | |
| 1457 /* | |
| 1458 Sound_FreeSample(data->sample); | |
| 1459 */ | |
| 1460 return -1; | |
| 1461 } | |
| 1462 /* Don't forget to add check for EOF */ | |
| 1463 /* Will return 0 bytes and pass the buck to check sample->flags */ | |
| 1464 if(0 == bytes_decoded) | |
| 1465 { | |
| 1466 #if 1 | |
| 1467 fprintf(stderr, "Hit eof while trying to buffer\n"); | |
| 1468 data->eof = 1; | |
| 1469 if(data->sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 1470 { | |
| 1471 fprintf(stderr, "\tEOF flag\n"); | |
| 1472 } | |
| 1473 if(data->sample->flags & SOUND_SAMPLEFLAG_CANSEEK) | |
| 1474 { | |
| 1475 fprintf(stderr, "\tCanSeek flag\n"); | |
| 1476 } | |
| 1477 if(data->sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 1478 { | |
| 1479 fprintf(stderr, "\tEAGAIN flag\n"); | |
| 1480 } | |
| 1481 if(data->sample->flags & SOUND_SAMPLEFLAG_NONE) | |
| 1482 { | |
| 1483 fprintf(stderr, "\tNONE flag\n"); | |
| 1484 } | |
| 1485 #endif | |
| 1486 return 0; | |
| 1487 } | |
| 1488 | |
| 1489 if(bytes_decoded < 16384) | |
| 1490 { | |
| 1491 char* tempbuffer1 = (char*)malloc(16384); | |
| 1492 char* tempbuffer2 = (char*)malloc(16384); | |
| 1493 int retval; | |
| 1494 memcpy(tempbuffer1, data->sample->buffer, bytes_decoded); | |
| 1495 retval = Sound_SetBufferSize(data->sample, 16384-bytes_decoded); | |
| 1496 if(retval == 1) | |
| 1497 { | |
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1498 ALuint new_bytes; |
| 0 | 1499 Sound_Rewind(data->sample); |
| 1500 new_bytes = Sound_Decode(data->sample); | |
| 1501 fprintf(stderr, "Orig bytes: %d, Make up bytes_decoded=%d, total=%d\n", bytes_decoded, new_bytes, new_bytes+bytes_decoded); | |
| 1502 | |
| 1503 memcpy(tempbuffer2, data->sample->buffer, new_bytes); | |
| 1504 | |
| 1505 retval = Sound_SetBufferSize(data->sample, 16384); | |
| 1506 fprintf(stderr, "Finished reset...now danger copy\n"); | |
| 1507 memcpy(data->sample->buffer, tempbuffer1,bytes_decoded); | |
| 1508 | |
| 1509 fprintf(stderr, "Finished reset...now danger copy2\n"); | |
| 1510 memcpy( &( ((char*)(data->sample->buffer))[bytes_decoded] ), tempbuffer2, new_bytes); | |
| 1511 | |
| 1512 fprintf(stderr, "Finished \n"); | |
| 1513 | |
| 1514 free(tempbuffer1); | |
| 1515 free(tempbuffer2); | |
| 1516 bytes_decoded += new_bytes; | |
| 1517 fprintf(stderr, "ASSERT bytes should equal 16384: %d\n", bytes_decoded); | |
| 1518 } | |
| 1519 else | |
| 1520 { | |
| 1521 fprintf(stderr, "Experiment failed: %s\n", Sound_GetError()); | |
| 1522 } | |
| 1523 } | |
| 1524 | |
| 1525 /* Now copy the data to the OpenAL buffer */ | |
| 1526 /* We can't just set a pointer because the API needs | |
| 1527 * its own copy to assist hardware acceleration */ | |
| 1528 alBufferData(buffer, | |
| 1529 TranslateFormat(&data->sample->desired), | |
| 1530 data->sample->buffer, | |
| 1531 bytes_decoded, | |
| 1532 data->sample->desired.rate | |
| 1533 ); | |
| 1534 if( (error = alGetError()) != AL_NO_ERROR) | |
| 1535 { | |
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1536 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 1537 return -1; |
| 1538 } | |
| 1539 | |
| 1540 fprintf(stderr, "GetMoreData2222 returning %d bytes decoded\n", bytes_decoded); | |
| 1541 return bytes_decoded; | |
| 1542 } | |
| 1543 #endif | |
| 1544 | |
| 1545 /************ END EXPERIEMENT - REMOVE ME *************************/ | |
| 1546 | |
| 1547 | |
| 1548 | |
| 1549 | |
| 1550 | |
| 1551 | |
| 1552 | |
| 1553 | |
| 1554 | |
| 1555 /* This function will look up the source for the corresponding channel */ | |
| 1556 /* Must return 0 on error instead of -1 because of unsigned int */ | |
|
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1557 static ALuint Internal_GetSource(ALint channel) |
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1558 { |
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1559 ALint i; |
| 0 | 1560 /* Make sure channel is in bounds */ |
| 1561 if(channel >= Number_of_Channels_global) | |
| 1562 { | |
| 1563 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 1564 return 0; | |
| 1565 } | |
| 1566 /* If the user specified -1, then return the an available source */ | |
| 1567 if(channel < 0) | |
| 1568 { | |
| 1569 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 1570 { | |
| 1571 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1572 { | |
| 1573 return ALmixer_Channel_List[i].alsource; | |
| 1574 } | |
| 1575 } | |
| 1576 /* If we get here, all sources are in use */ | |
| 1577 /* Error message seems too harsh | |
| 1578 ALmixer_SetError("All sources are in use"); | |
| 1579 */ | |
| 1580 return 0; | |
| 1581 } | |
| 1582 /* Last case: Return the source for the channel */ | |
| 1583 return ALmixer_Channel_List[channel].alsource; | |
| 1584 } | |
| 1585 | |
| 1586 /* This function will look up the channel for the corresponding source */ | |
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1587 static ALint Internal_GetChannel(ALuint source) |
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1588 { |
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1589 ALint i; |
| 0 | 1590 /* Only the first value is used for the key */ |
| 1591 Source_Map key = { 0, 0 }; | |
| 1592 Source_Map* found_item = NULL; | |
| 1593 key.source = source; | |
| 1594 | |
| 1595 /* If the source is 0, look up the first available channel */ | |
| 1596 if(0 == source) | |
| 1597 { | |
| 1598 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 1599 { | |
| 1600 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1601 { | |
| 1602 return i; | |
| 1603 } | |
| 1604 } | |
| 1605 /* If we get here, all sources are in use */ | |
| 1606 /* Error message seems too harsh | |
| 1607 ALmixer_SetError("All channels are in use"); | |
| 1608 */ | |
| 1609 return -1; | |
| 1610 } | |
| 1611 | |
| 1612 | |
| 1613 /* Else, look up the source and return the channel */ | |
| 1614 if(AL_FALSE == alIsSource(source)) | |
| 1615 { | |
| 1616 ALmixer_SetError("Is not a source"); | |
| 1617 return -1; | |
| 1618 } | |
| 1619 | |
| 1620 /* Use the ANSI C binary search feature (yea!) */ | |
| 1621 found_item = (Source_Map*)bsearch(&key, Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 1622 if(NULL == found_item) | |
| 1623 { | |
| 1624 ALmixer_SetError("Source is valid but not registered with ALmixer (to a channel)"); | |
| 1625 return -1; | |
| 1626 } | |
| 1627 return found_item->channel; | |
| 1628 } | |
| 1629 | |
| 1630 | |
| 1631 | |
| 1632 /* This function will find the first available channel (not in use) | |
| 1633 * from the specified start channel. Reserved channels to not qualify | |
| 1634 * as available. | |
| 1635 */ | |
|
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1636 static ALint Internal_FindFreeChannel(ALint start_channel) |
| 0 | 1637 { |
| 1638 /* Start at the number of reserved so we skip over | |
| 1639 * all the reserved channels. | |
| 1640 */ | |
|
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1641 ALint i = Number_of_Reserve_Channels_global; |
| 0 | 1642 /* Quick check to see if we're out of bounds */ |
| 1643 if(start_channel >= Number_of_Channels_global) | |
| 1644 { | |
| 1645 return -1; | |
| 1646 } | |
| 1647 | |
| 1648 /* If the start channel is even higher than the reserved, | |
| 1649 * then start at the higher value. | |
| 1650 */ | |
| 1651 if(start_channel > Number_of_Reserve_Channels_global) | |
| 1652 { | |
| 1653 i = start_channel; | |
| 1654 } | |
| 1655 | |
| 1656 /* i has already been set */ | |
| 1657 for( ; i<Number_of_Channels_global; i++) | |
| 1658 { | |
| 1659 if( ! ALmixer_Channel_List[i].channel_in_use ) | |
| 1660 { | |
| 1661 return i; | |
| 1662 } | |
| 1663 } | |
| 1664 /* If we get here, all sources are in use */ | |
| 1665 return -1; | |
| 1666 } | |
| 1667 | |
| 1668 | |
| 1669 | |
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1670 /* Will return the number of channels halted |
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1671 * or 0 for error |
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1672 */ |
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1673 static ALint Internal_HaltChannel(ALint channel, ALboolean did_finish_naturally) |
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1674 { |
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1675 ALint retval = 0; |
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1676 ALint counter = 0; |
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1677 ALenum error; |
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1678 ALint buffers_still_queued; |
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1679 ALint buffers_processed; |
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1680 |
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1681 if(channel >= Number_of_Channels_global) |
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1682 { |
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1683 ALmixer_SetError("Cannot halt channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); |
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1684 return -1; |
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1685 } |
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1686 /* If the user specified a specific channel */ |
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1687 if(channel >= 0) |
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1688 { |
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1689 /* only need to process channel if in use */ |
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1690 if(ALmixer_Channel_List[channel].channel_in_use) |
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1691 { |
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1692 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1693 if((error = alGetError()) != AL_NO_ERROR) |
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1694 { |
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1695 fprintf(stderr, "14Testing error: %s\n", |
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1696 alGetString(error)); |
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1697 } |
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1698 /* Here's the situation. My old method of using |
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1699 * alSourceUnqueueBuffers() seemed to be invalid in light |
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1700 * of all the problems I suffered through with getting |
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1701 * the CoreData backend to work with this code. |
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1702 * As such, I'm changing all the code to set the buffer to |
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1703 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1704 * doesn't need to apply here. For non-queued, Loki, |
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1705 * Creative Windows, and CoreAudio seem to leave the |
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1706 * buffer queued (Old Mac didn't.) For queued, we need to |
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1707 * remove the processed buffers and force remove the |
|
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1708 * still-queued buffers. |
|
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|
1709 */ |
|
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1710 alGetSourcei( |
|
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|
1711 ALmixer_Channel_List[channel].alsource, |
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1712 AL_BUFFERS_QUEUED, &buffers_still_queued |
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1713 ); |
|
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1714 if((error = alGetError()) != AL_NO_ERROR) |
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1715 { |
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1716 fprintf(stderr, "17Testing Error with buffers_still_queued: %s", |
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1717 alGetString(error)); |
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1718 ALmixer_SetError("Failed detecting still queued buffers: %s", |
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1719 alGetString(error) ); |
|
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1720 retval = -1; |
|
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1721 } |
|
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1722 alGetSourcei( |
|
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1723 ALmixer_Channel_List[channel].alsource, |
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1724 AL_BUFFERS_PROCESSED, &buffers_processed |
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1725 ); |
|
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1726 if((error = alGetError()) != AL_NO_ERROR) |
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|
1727 { |
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1728 fprintf(stderr, "17Testing Error with buffers_processed: %s", |
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1729 alGetString(error)); |
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1730 ALmixer_SetError("Failed detecting still processed buffers: %s", |
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1731 alGetString(error) ); |
|
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1732 retval = -1; |
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1733 } |
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1734 /* If either of these is greater than 0, it means we need |
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1735 * to clear the source |
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1736 */ |
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1737 if((buffers_still_queued > 0) || (buffers_processed > 0)) |
|
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|
1738 { |
|
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|
1739 alSourcei(ALmixer_Channel_List[channel].alsource, |
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1740 AL_BUFFER, |
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1741 AL_NONE); |
|
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1742 if((error = alGetError()) != AL_NO_ERROR) |
|
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1743 { |
|
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1744 fprintf(stderr, "17Testing Error with clearing buffer from source: %s", |
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1745 alGetString(error)); |
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1746 ALmixer_SetError("Failed to clear buffer from source: %s", |
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1747 alGetString(error) ); |
|
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1748 retval = -1; |
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1749 } |
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1750 } |
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1751 |
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|
1752 ALmixer_Channel_List[channel].almixer_data->num_buffers_in_use = 0; |
|
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1753 |
|
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1754 /* Launch callback for consistency? */ |
|
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|
1755 Invoke_Channel_Done_Callback(channel, did_finish_naturally); |
|
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1756 |
|
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1757 Clean_Channel(channel); |
|
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1758 Is_Playing_global--; |
|
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1759 counter++; |
|
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1760 } |
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1761 } |
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1762 /* The user wants to halt all channels */ |
|
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1763 else |
|
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1764 { |
|
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1765 ALint i; |
|
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1766 for(i=0; i<Number_of_Channels_global; i++) |
|
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1767 { |
|
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1768 /* only need to process channel if in use */ |
|
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1769 if(ALmixer_Channel_List[i].channel_in_use) |
|
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1770 { |
|
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1771 alSourceStop(ALmixer_Channel_List[i].alsource); |
|
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1772 if((error = alGetError()) != AL_NO_ERROR) |
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1773 { |
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1774 fprintf(stderr, "19Testing error: %s\n", |
|
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1775 alGetString(error)); |
|
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1776 } |
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1777 |
|
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1778 /* Here's the situation. My old method of using |
|
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1779 * alSourceUnqueueBuffers() seemed to be invalid in light |
|
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1780 * of all the problems I suffered through with getting |
|
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1781 * the CoreData backend to work with this code. |
|
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1782 * As such, I'm changing all the code to set the buffer to |
|
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1783 * AL_NONE. Furthermore, the queued vs. non-queued issue |
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1784 * doesn't need to apply here. For non-queued, Loki, |
|
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1785 * Creative Windows, and CoreAudio seem to leave the |
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1786 * buffer queued (Old Mac didn't.) For queued, we need to |
|
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1787 * remove the processed buffers and force remove the |
|
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1788 * still-queued buffers. |
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1789 */ |
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1790 alGetSourcei( |
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1791 ALmixer_Channel_List[i].alsource, |
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1792 AL_BUFFERS_QUEUED, &buffers_still_queued |
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1793 ); |
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1794 if((error = alGetError()) != AL_NO_ERROR) |
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1795 { |
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1796 fprintf(stderr, "17Testing Error with buffers_still_queued: %s", |
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1797 alGetString(error)); |
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1798 ALmixer_SetError("Failed detecting still queued buffers: %s", |
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1799 alGetString(error) ); |
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1800 retval = -1; |
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1801 } |
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1802 alGetSourcei( |
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1803 ALmixer_Channel_List[i].alsource, |
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1804 AL_BUFFERS_PROCESSED, &buffers_processed |
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1805 ); |
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1806 if((error = alGetError()) != AL_NO_ERROR) |
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1807 { |
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1808 fprintf(stderr, "17Testing Error with buffers_processed: %s", |
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1809 alGetString(error)); |
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1810 ALmixer_SetError("Failed detecting still processed buffers: %s", |
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1811 alGetString(error) ); |
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1812 retval = -1; |
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1813 } |
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1814 /* If either of these is greater than 0, it means we need |
|
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|
1815 * to clear the source |
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1816 */ |
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1817 if((buffers_still_queued > 0) || (buffers_processed > 0)) |
|
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1818 { |
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1819 alSourcei(ALmixer_Channel_List[i].alsource, |
|
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1820 AL_BUFFER, |
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1821 AL_NONE); |
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1822 if((error = alGetError()) != AL_NO_ERROR) |
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1823 { |
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1824 fprintf(stderr, "17Testing Error with clearing buffer from source: %s", |
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1825 alGetString(error)); |
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1826 ALmixer_SetError("Failed to clear buffer from source: %s", |
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1827 alGetString(error) ); |
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1828 retval = -1; |
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1829 } |
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1830 } |
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1831 |
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1832 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
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1833 |
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1834 /* Launch callback for consistency? */ |
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1835 Invoke_Channel_Done_Callback(i, did_finish_naturally); |
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1836 |
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1837 Clean_Channel(i); |
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1838 Is_Playing_global--; |
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1839 |
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1840 /* Increment the counter */ |
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1841 counter++; |
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1842 } |
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1843 /* Let's halt everything just in case there |
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1844 * are bugs. |
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1845 */ |
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1846 /* |
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1847 else |
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1848 { |
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1849 alSourceStop(ALmixer_Channel_List[channel].alsource); |
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1850 / * Can't clean because the in_use counter for |
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1851 * data will get messed up * / |
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1852 Clean_Channel(channel); |
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1853 } |
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1854 */ |
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1855 /* Just in case */ |
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1856 Is_Playing_global = 0; |
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1857 } |
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1858 } |
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1859 if(-1 == retval) |
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1860 { |
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1861 return -1; |
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1862 } |
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1863 return counter; |
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1864 } |
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1865 |
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1866 |
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1867 /* Will return the source halted or the total number of channels |
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1868 * if all were halted or 0 for error |
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1869 */ |
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1870 static ALint Internal_HaltSource(ALuint source, ALboolean did_finish_naturally) |
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1871 { |
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1872 ALint channel; |
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1873 if(0 == source) |
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1874 { |
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1875 /* Will return the number of sources halted */ |
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1876 return Internal_HaltChannel(-1, did_finish_naturally); |
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1877 } |
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1878 |
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1879 channel = Internal_GetChannel(source); |
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1880 if(-1 == channel) |
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1881 { |
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1882 ALmixer_SetError("Cannot halt source: %s", ALmixer_GetError()); |
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1883 return -1; |
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1884 } |
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1885 return Internal_HaltChannel(channel, did_finish_naturally); |
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1886 } |
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1887 |
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1888 |
| 0 | 1889 |
| 1890 /* Note: Behaves, almost like SDL_mixer, but keep in mind | |
| 1891 * that there is no "music" channel anymore, so 0 | |
| 1892 * will remove everything. (Note, I no longer allow 0 | |
| 1893 * so it gets set to the default number.) | |
| 1894 * Also, callbacks for deleted channels will not be called. | |
| 1895 * I really need to do error checking, for realloc and | |
| 1896 * GenSources, but reversing the damage is too painful | |
| 1897 * for me to think about at the moment, so it's not in here. | |
| 1898 */ | |
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1899 static ALint Internal_AllocateChannels(ALint numchans) |
| 0 | 1900 { |
| 1901 ALenum error; | |
| 1902 int i; | |
| 1903 /* Return info */ | |
| 1904 if(numchans < 0) | |
| 1905 { | |
| 1906 return Number_of_Channels_global; | |
| 1907 } | |
| 1908 if(0 == numchans) | |
| 1909 { | |
| 1910 numchans = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 1911 } | |
| 1912 /* No change */ | |
| 1913 if(numchans == Number_of_Channels_global) | |
| 1914 { | |
| 1915 return Number_of_Channels_global; | |
| 1916 } | |
| 1917 /* We need to increase the number of channels */ | |
| 1918 if(numchans > Number_of_Channels_global) | |
| 1919 { | |
| 1920 /* Not sure how safe this is, but SDL_mixer does it | |
| 1921 * the same way */ | |
| 1922 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
| 1923 | |
| 1924 /* Allocate memory for the list of sources that map to the channels */ | |
| 1925 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
| 1926 | |
| 1927 for(i=Number_of_Channels_global; i<numchans; i++) | |
| 1928 { | |
| 1929 Init_Channel(i); | |
| 1930 /* Generate a new source and associate it with the channel */ | |
| 1931 alGenSources(1, &ALmixer_Channel_List[i].alsource); | |
| 1932 if((error = alGetError()) != AL_NO_ERROR) | |
| 1933 { | |
| 1934 fprintf(stderr, "12Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
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1935 alGetString(error)); |
| 0 | 1936 } |
| 1937 /* Copy the source so the SourceMap has it too */ | |
| 1938 Source_Map_List[i].source = ALmixer_Channel_List[i].alsource; | |
| 1939 Source_Map_List[i].channel = i; | |
| 1940 /* Clean the channel because there are some things that need to | |
| 1941 * be done that can't happen until the source is set | |
| 1942 */ | |
| 1943 Clean_Channel(i); | |
| 1944 } | |
| 1945 | |
| 1946 /* The Source_Map_List must be sorted by source for binary searches | |
| 1947 */ | |
| 1948 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
| 1949 | |
| 1950 Number_of_Channels_global = numchans; | |
| 1951 return numchans; | |
| 1952 } | |
| 1953 /* Need to remove channels. This might be dangerous */ | |
| 1954 if(numchans < Number_of_Channels_global) | |
| 1955 { | |
| 1956 for(i=numchans; i<Number_of_Channels_global; i++) | |
| 1957 { | |
| 1958 /* Halt the channel */ | |
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1959 Internal_HaltChannel(i, AL_FALSE); |
| 0 | 1960 |
| 1961 /* Delete source associated with the channel */ | |
| 1962 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
| 1963 if((error = alGetError()) != AL_NO_ERROR) | |
| 1964 { | |
| 1965 fprintf(stderr, "13Testing error: %s\n", | |
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1966 alGetString(error)); |
| 0 | 1967 } |
| 1968 } | |
| 1969 | |
| 1970 | |
| 1971 /* Not sure how safe this is, but SDL_mixer does it | |
| 1972 * the same way */ | |
| 1973 ALmixer_Channel_List = (struct ALmixer_Channel*) realloc( ALmixer_Channel_List, numchans * sizeof(struct ALmixer_Channel)); | |
| 1974 | |
| 1975 /* The tricky part is that we must remove the entries | |
| 1976 * in the source map that correspond to the deleted channels. | |
| 1977 * We'll resort the map by channels so we can pick them off | |
| 1978 * in order. | |
| 1979 */ | |
| 1980 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map_by_channel); | |
| 1981 | |
| 1982 /* Deallocate memory for the list of sources that map to the channels */ | |
| 1983 Source_Map_List = (Source_Map*) realloc(Source_Map_List, numchans * sizeof(Source_Map)); | |
| 1984 | |
| 1985 /* Now resort the map by source and the correct num of chans */ | |
| 1986 qsort(Source_Map_List, numchans, sizeof(Source_Map), Compare_Source_Map); | |
| 1987 | |
| 1988 /* Reset the number of channels */ | |
| 1989 Number_of_Channels_global = numchans; | |
| 1990 return numchans; | |
| 1991 } | |
| 1992 /* Shouldn't ever reach here */ | |
| 1993 return -1; | |
| 1994 | |
| 1995 } | |
| 1996 | |
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1997 static ALint Internal_ReserveChannels(ALint num) |
| 0 | 1998 { |
| 1999 /* Can't reserve more than the max num of channels */ | |
| 2000 /* Actually, I'll allow it for people who just want to | |
| 2001 * set the value really high to effectively disable | |
| 2002 * auto-assignment | |
| 2003 */ | |
| 2004 | |
| 2005 /* Return the current number of reserved channels */ | |
| 2006 if(num < 0) | |
| 2007 { | |
| 2008 return Number_of_Reserve_Channels_global; | |
| 2009 } | |
| 2010 Number_of_Reserve_Channels_global = num; | |
| 2011 return Number_of_Reserve_Channels_global; | |
| 2012 } | |
| 2013 | |
| 2014 | |
| 2015 /* This will rewind the SDL_Sound sample for streamed | |
| 2016 * samples and start buffering up the data for the next | |
| 2017 * playback. This may require samples to be halted | |
| 2018 */ | |
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2019 static ALint Internal_RewindData(ALmixer_Data* data) |
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2020 { |
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2021 ALint retval = 0; |
| 0 | 2022 /* |
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2023 ALint bytes_returned; |
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2024 ALint i; |
| 0 | 2025 */ |
| 2026 if(NULL == data) | |
| 2027 { | |
| 2028 ALmixer_SetError("Cannot rewind because data is NULL\n"); | |
| 2029 return -1; | |
| 2030 } | |
| 2031 | |
| 2032 | |
| 2033 /* Might have to require Halt */ | |
| 2034 /* Okay, we assume Halt or natural stop has already | |
| 2035 * cleared the data buffers | |
| 2036 */ | |
| 2037 if(data->in_use) | |
| 2038 { | |
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2039 /* |
| 0 | 2040 fprintf(stderr, "Warning sample is in use. May not be able to rewind\n"); |
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2041 */ |
| 0 | 2042 /* |
| 2043 ALmixer_SetError("Data is in use. Cannot rewind unless all sources using the data are halted\n"); | |
| 2044 return -1; | |
| 2045 */ | |
| 2046 } | |
| 2047 | |
| 2048 | |
| 2049 /* Because Seek can alter things even in predecoded data, | |
| 2050 * decoded data must also be rewound | |
| 2051 */ | |
| 2052 if(data->decoded_all) | |
| 2053 { | |
| 2054 data->eof = 0; | |
| 2055 | |
| 2056 #if 0 | |
| 2057 #if defined(DISABLE_PREDECODED_SEEK) | |
| 2058 /* Since we can't seek predecoded stuff, it should be rewound */ | |
| 2059 return 0; | |
| 2060 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 2061 /* This case is if the Sound_Sample has been deleted. | |
| 2062 * It assumes the data is already at the beginning. | |
| 2063 */ | |
| 2064 if(NULL == data->sample) | |
| 2065 { | |
| 2066 return 0; | |
| 2067 } | |
| 2068 /* Else, the sample has already been reallocated, | |
| 2069 * and we can fall to normal behavior | |
| 2070 */ | |
| 2071 #endif | |
| 2072 #endif | |
| 2073 /* If access_data, was enabled, the sound sample | |
| 2074 * still exists and we can do stuff. | |
| 2075 * If it's NULL, we can't do anything, but | |
| 2076 * it should already be "rewound". | |
| 2077 */ | |
| 2078 if(NULL == data->sample) | |
| 2079 { | |
| 2080 return 0; | |
| 2081 } | |
| 2082 /* Else, the sample has already been reallocated, | |
| 2083 * and we can fall to normal behavior | |
| 2084 */ | |
| 2085 | |
| 2086 Set_Predecoded_Seek_Position(data, 0); | |
| 2087 /* | |
| 2088 return data->total_bytes; | |
| 2089 */ | |
| 2090 return 0; | |
| 2091 } | |
| 2092 | |
| 2093 /* Remaining stuff for streamed data */ | |
| 2094 | |
| 2095 data->eof = 0; | |
| 2096 retval = Sound_Rewind(data->sample); | |
| 2097 if(0 == retval) | |
| 2098 { | |
| 2099 ALmixer_SetError( Sound_GetError() ); | |
| 2100 return -1; | |
| 2101 } | |
| 2102 #if 0 | |
| 2103 /* Clear error */ | |
| 2104 alGetError(); | |
| 2105 for(i=0; i<data->num_buffers; i++) | |
| 2106 { | |
| 2107 bytes_returned = GetMoreData(data, data->buffer[i]); | |
| 2108 if(-1 == bytes_returned) | |
| 2109 { | |
| 2110 return -1; | |
| 2111 } | |
| 2112 else if(0 == bytes_returned) | |
| 2113 { | |
| 2114 return -1; | |
| 2115 } | |
| 2116 retval += bytes_returned; | |
| 2117 | |
| 2118 } | |
| 2119 #endif | |
| 2120 | |
| 2121 | |
| 2122 | |
| 2123 return retval; | |
| 2124 } | |
| 2125 | |
| 2126 | |
| 2127 | |
| 2128 | |
|
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2129 static ALint Internal_RewindChannel(ALint channel) |
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2130 { |
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2131 ALint retval = 0; |
| 0 | 2132 ALenum error; |
| 2133 ALint state; | |
| 2134 | |
| 2135 if(channel >= Number_of_Channels_global) | |
| 2136 { | |
| 2137 ALmixer_SetError("Cannot rewind channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2138 return -1; | |
| 2139 } | |
| 2140 | |
| 2141 if((error = alGetError()) != AL_NO_ERROR) | |
| 2142 { | |
| 2143 fprintf(stderr, "24Testing error: %s\n", | |
|
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2144 alGetString(error)); |
| 0 | 2145 } |
| 2146 /* Clear error */ | |
| 2147 alGetError(); | |
| 2148 | |
| 2149 /* If the user specified a specific channel */ | |
| 2150 if(channel >= 0) | |
| 2151 { | |
| 2152 /* only need to process channel if in use */ | |
| 2153 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 2154 { | |
| 2155 | |
| 2156 /* What should I do? Do I just rewind the channel | |
| 2157 * or also rewind the data? Since the data is | |
| 2158 * shared, let's make it the user's responsibility | |
| 2159 * to rewind the data. | |
| 2160 */ | |
| 2161 if(ALmixer_Channel_List[channel].almixer_data->decoded_all) | |
| 2162 { | |
| 2163 alGetSourcei( | |
| 2164 ALmixer_Channel_List[channel].alsource, | |
| 2165 AL_SOURCE_STATE, &state | |
| 2166 ); | |
| 2167 if((error = alGetError()) != AL_NO_ERROR) | |
| 2168 { | |
| 2169 fprintf(stderr, "25Testing error: %s\n", | |
|
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2170 alGetString(error)); |
| 0 | 2171 } |
| 2172 alSourceRewind(ALmixer_Channel_List[channel].alsource); | |
| 2173 if((error = alGetError()) != AL_NO_ERROR) | |
| 2174 { | |
| 2175 ALmixer_SetError("%s", | |
|
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2176 alGetString(error) ); |
| 0 | 2177 retval = -1; |
| 2178 } | |
| 2179 /* Need to resume playback if it was originally playing */ | |
| 2180 if(AL_PLAYING == state) | |
| 2181 { | |
| 2182 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 2183 if((error = alGetError()) != AL_NO_ERROR) | |
| 2184 { | |
| 2185 ALmixer_SetError("%s", | |
|
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2186 alGetString(error) ); |
| 0 | 2187 retval = -1; |
| 2188 } | |
| 2189 } | |
| 2190 else if(AL_PAUSED == state) | |
| 2191 { | |
| 2192 /* HACK: The problem is that when paused, after | |
| 2193 * the Rewind, I can't get it off the INITIAL | |
| 2194 * state without restarting | |
| 2195 */ | |
| 2196 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 2197 if((error = alGetError()) != AL_NO_ERROR) | |
| 2198 { | |
| 2199 fprintf(stderr, "25Testing error: %s\n", | |
|
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2200 alGetString(error)); |
| 0 | 2201 } |
| 2202 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
| 2203 if((error = alGetError()) != AL_NO_ERROR) | |
| 2204 { | |
| 2205 ALmixer_SetError("%s", | |
|
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2206 alGetString(error) ); |
| 0 | 2207 retval = -1; |
| 2208 } | |
| 2209 } | |
| 2210 } | |
| 2211 else | |
| 2212 { | |
| 2213 /* Streamed data is different. Rewinding the channel | |
| 2214 * does no good. Rewinding the data will have an | |
| 2215 * effect, but it will be lagged based on how | |
| 2216 * much data is queued. Recommend users call Halt | |
| 2217 * before rewind if they want immediate results. | |
| 2218 */ | |
| 2219 retval = Internal_RewindData(ALmixer_Channel_List[channel].almixer_data); | |
| 2220 } | |
| 2221 } | |
| 2222 } | |
|
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2223 /* The user wants to rewind all channels */ |
| 0 | 2224 else |
| 2225 { | |
|
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2226 ALint i; |
| 0 | 2227 for(i=0; i<Number_of_Channels_global; i++) |
| 2228 { | |
| 2229 /* only need to process channel if in use */ | |
| 2230 if(ALmixer_Channel_List[i].channel_in_use) | |
| 2231 { | |
| 2232 /* What should I do? Do I just rewind the channel | |
| 2233 * or also rewind the data? Since the data is | |
| 2234 * shared, let's make it the user's responsibility | |
| 2235 * to rewind the data. | |
| 2236 */ | |
| 2237 if(ALmixer_Channel_List[i].almixer_data->decoded_all) | |
| 2238 { | |
| 2239 alGetSourcei( | |
| 2240 ALmixer_Channel_List[i].alsource, | |
| 2241 AL_SOURCE_STATE, &state | |
| 2242 ); | |
| 2243 if((error = alGetError()) != AL_NO_ERROR) | |
| 2244 { | |
| 2245 fprintf(stderr, "26Testing error: %s\n", | |
|
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2246 alGetString(error)); |
| 0 | 2247 } |
| 2248 alSourceRewind(ALmixer_Channel_List[i].alsource); | |
| 2249 if((error = alGetError()) != AL_NO_ERROR) | |
| 2250 { | |
| 2251 ALmixer_SetError("%s", | |
|
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2252 alGetString(error) ); |
| 0 | 2253 retval = -1; |
| 2254 } | |
| 2255 /* Need to resume playback if it was originally playing */ | |
| 2256 if(AL_PLAYING == state) | |
| 2257 { | |
| 2258 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 2259 if((error = alGetError()) != AL_NO_ERROR) | |
| 2260 { | |
| 2261 ALmixer_SetError("%s", | |
|
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2262 alGetString(error) ); |
| 0 | 2263 retval = -1; |
| 2264 } | |
| 2265 } | |
| 2266 else if(AL_PAUSED == state) | |
| 2267 { | |
| 2268 /* HACK: The problem is that when paused, after | |
| 2269 * the Rewind, I can't get it off the INITIAL | |
| 2270 * state without restarting | |
| 2271 */ | |
| 2272 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 2273 if((error = alGetError()) != AL_NO_ERROR) | |
| 2274 { | |
| 2275 fprintf(stderr, "27Testing error: %s\n", | |
|
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|
2276 alGetString(error)); |
| 0 | 2277 } |
| 2278 alSourcePause(ALmixer_Channel_List[i].alsource); | |
| 2279 if((error = alGetError()) != AL_NO_ERROR) | |
| 2280 { | |
| 2281 ALmixer_SetError("%s", | |
|
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2282 alGetString(error) ); |
| 0 | 2283 retval = -1; |
| 2284 } | |
| 2285 } | |
| 2286 } | |
| 2287 else | |
| 2288 { | |
| 2289 /* Streamed data is different. Rewinding the channel | |
| 2290 * does no good. Rewinding the data will have an | |
| 2291 * effect, but it will be lagged based on how | |
| 2292 * much data is queued. Recommend users call Halt | |
| 2293 * before rewind if they want immediate results. | |
| 2294 */ | |
| 2295 retval = Internal_RewindData(ALmixer_Channel_List[i].almixer_data); | |
| 2296 } | |
| 2297 } | |
| 2298 } | |
| 2299 } | |
| 2300 return retval; | |
| 2301 } | |
| 2302 | |
| 2303 | |
|
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|
2304 static ALint Internal_RewindSource(ALuint source) |
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2305 { |
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2306 ALint channel; |
| 0 | 2307 if(0 == source) |
| 2308 { | |
|
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2309 return Internal_RewindChannel(-1) + 1; |
| 0 | 2310 } |
| 2311 | |
| 2312 channel = Internal_GetChannel(source); | |
| 2313 if(-1 == channel) | |
| 2314 { | |
| 2315 ALmixer_SetError("Cannot rewind source: %s", ALmixer_GetError()); | |
| 2316 return 0; | |
| 2317 } | |
|
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|
2318 return Internal_RewindChannel(channel) + 1; |
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2319 } |
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2320 |
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2321 |
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2322 |
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2323 |
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2324 |
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|
2325 static ALint Internal_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
|
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2326 { |
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|
2327 ALenum error; |
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|
2328 int ret_flag = 0; |
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|
2329 if(NULL == data) |
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2330 { |
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|
2331 ALmixer_SetError("Can't play because data is NULL\n"); |
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2332 return -1; |
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2333 } |
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|
2334 |
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|
2335 /* There isn't a good way to share streamed files because |
|
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|
2336 * the decoded data doesn't stick around. |
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2337 * You must "Load" a brand new instance of |
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2338 * the data. If you try using the same data, |
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2339 * bad things may happen. This check will attempt |
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|
2340 * to prevent sharing |
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|
2341 */ |
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|
2342 if(0 == data->decoded_all) |
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2343 { |
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2344 if(data->in_use) |
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2345 { |
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2346 ALmixer_SetError("Can't play shared streamed sample because it is already in use"); |
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2347 return -1; |
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|
2348 } |
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|
2349 |
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|
2350 /* Make sure SDL_sound sample is not at EOF. |
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2351 * This mainly affects streamed files, |
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2352 * so the check is placed here |
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2353 */ |
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|
2354 if(data->eof) |
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2355 { |
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|
2356 if( -1 == Internal_RewindData(data) ) |
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2357 { |
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|
2358 ALmixer_SetError("Can't play sample because it is at EOF and cannot rewind"); |
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2359 return -1; |
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2360 } |
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2361 } |
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2362 } |
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2363 /* We need to provide the user with the first available channel */ |
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2364 if(-1 == channel) |
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|
2365 { |
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|
2366 ALint i; |
|
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2367 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) |
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2368 { |
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2369 if(0 == ALmixer_Channel_List[i].channel_in_use) |
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|
2370 { |
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|
2371 channel = i; |
|
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|
2372 break; |
|
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|
2373 } |
|
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|
2374 } |
|
279d0427ef26
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parents:
1
diff
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|
2375 /* if we couldn't find a channel, return an error */ |
|
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1
diff
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|
2376 if(i == Number_of_Channels_global) |
|
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|
2377 { |
|
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diff
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|
2378 ALmixer_SetError("No channels available for playing"); |
|
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1
diff
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|
2379 return -1; |
|
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1
diff
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|
2380 } |
|
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diff
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|
2381 } |
|
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diff
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|
2382 /* If we didn't assign the channel number, make sure it's not |
|
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diff
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|
2383 * out of bounds or in use */ |
|
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|
2384 else |
|
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|
2385 { |
|
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|
2386 if(channel >= Number_of_Channels_global) |
|
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|
2387 { |
|
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1
diff
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|
2388 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
|
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|
2389 return -1; |
|
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|
2390 } |
|
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diff
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|
2391 else if(ALmixer_Channel_List[channel].channel_in_use) |
|
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|
2392 { |
|
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diff
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|
2393 ALmixer_SetError("Requested channel (%d) is in use", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
|
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|
2394 return -1; |
|
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|
2395 } |
|
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|
2396 } |
|
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|
2397 /* Make sure the user doesn't enter some meaningless value */ |
|
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diff
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|
2398 if(loops < -1) |
|
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|
2399 { |
|
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diff
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|
2400 loops = -1; |
|
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|
2401 } |
|
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diff
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|
2402 |
|
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|
2403 /* loops will probably have to change to be controlled by SDL_Sound */ |
|
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|
2404 |
|
279d0427ef26
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diff
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|
2405 /* Set up the initial values for playing */ |
|
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diff
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|
2406 ALmixer_Channel_List[channel].channel_in_use = 1; |
|
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diff
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|
2407 data->in_use++; |
|
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|
2408 |
|
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|
2409 /* Shouldn't need updating until a callback is fired |
|
279d0427ef26
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diff
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|
2410 * (assuming that we call Play in this function |
|
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|
2411 */ |
|
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diff
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|
2412 ALmixer_Channel_List[channel].needs_stream = 0; |
|
279d0427ef26
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diff
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|
2413 ALmixer_Channel_List[channel].almixer_data = data; |
|
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diff
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|
2414 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
|
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|
2415 |
|
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diff
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|
2416 /* If user entered -1 (or less), set to -1 */ |
|
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diff
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|
2417 if(ticks < 0) |
|
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|
2418 { |
|
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diff
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|
2419 ALmixer_Channel_List[channel].expire_ticks = -1; |
|
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|
2420 } |
|
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|
2421 else |
|
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diff
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|
2422 { |
|
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diff
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|
2423 ALmixer_Channel_List[channel].expire_ticks = ticks; |
|
279d0427ef26
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|
2424 } |
|
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diff
changeset
|
2425 |
|
279d0427ef26
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diff
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|
2426 |
|
279d0427ef26
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diff
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|
2427 ALmixer_Channel_List[channel].halted = 0; |
|
279d0427ef26
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diff
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|
2428 ALmixer_Channel_List[channel].paused = 0; |
|
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diff
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|
2429 |
|
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diff
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|
2430 /* Ran just use OpenAL to control loops if predecoded and infinite */ |
|
279d0427ef26
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diff
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|
2431 ALmixer_Channel_List[channel].loops = loops; |
|
279d0427ef26
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diff
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|
2432 if( (-1 == loops) && (data->decoded_all) ) |
|
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|
2433 { |
|
279d0427ef26
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1
diff
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|
2434 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
279d0427ef26
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|
2435 } |
|
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diff
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|
2436 else |
|
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1
diff
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|
2437 { |
|
279d0427ef26
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1
diff
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|
2438 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
|
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|
2439 } |
|
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1
diff
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|
2440 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
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|
2441 { |
|
279d0427ef26
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diff
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|
2442 fprintf(stderr, "13Testing error: %s\n", |
|
279d0427ef26
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1
diff
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|
2443 alGetString(error)); |
|
279d0427ef26
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diff
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|
2444 } |
|
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1
diff
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|
2445 |
|
279d0427ef26
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1
diff
changeset
|
2446 #if 0 |
|
279d0427ef26
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diff
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|
2447 /* Because of the corner case, predecoded |
|
279d0427ef26
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1
diff
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|
2448 * files must add +1 to the loops. |
|
279d0427ef26
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diff
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|
2449 * Streams do not have this problem |
|
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1
diff
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|
2450 * because they can use the eof flag to |
|
279d0427ef26
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1
diff
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|
2451 * avoid the conflict. |
|
279d0427ef26
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1
diff
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|
2452 * Sharing data chunks prevents the use of the eof flag. |
|
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diff
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|
2453 * Since streams, cannot share, only predecoded |
|
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|
2454 * files are affected |
|
279d0427ef26
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|
2455 */ |
|
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diff
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|
2456 if(data->decoded_all) |
|
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|
2457 { |
|
279d0427ef26
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diff
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|
2458 /* Corner Case: Now that play calls are pushed |
|
279d0427ef26
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diff
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|
2459 * off to update(), the start call must |
|
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1
diff
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|
2460 * also come through here. So, start loops |
|
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|
2461 * must be +1 |
|
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|
2462 */ |
|
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|
2463 if(-1 == loops) |
|
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|
2464 { |
|
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1
diff
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|
2465 /* -1 is a special case, and you don't want |
|
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|
2466 * to add +1 to it */ |
|
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|
2467 ALmixer_Channel_List[channel].loops = -1; |
|
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|
2468 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
279d0427ef26
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|
2469 } |
|
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|
2470 else |
|
279d0427ef26
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1
diff
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|
2471 { |
|
279d0427ef26
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diff
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|
2472 ALmixer_Channel_List[channel].loops = loops+1; |
|
279d0427ef26
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|
2473 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_FALSE); |
|
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diff
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|
2474 } |
|
279d0427ef26
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|
2475 } |
|
279d0427ef26
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|
2476 else |
|
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diff
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|
2477 { |
|
279d0427ef26
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diff
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|
2478 ALmixer_Channel_List[channel].loops = loops; |
|
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1
diff
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|
2479 /* Can we really loop on streamed data? */ |
|
279d0427ef26
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diff
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|
2480 alSourcei(ALmixer_Channel_List[channel].alsource, AL_LOOPING, AL_TRUE); |
|
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diff
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|
2481 } |
|
279d0427ef26
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1
diff
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|
2482 #endif |
|
279d0427ef26
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1
diff
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|
2483 |
|
279d0427ef26
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1
diff
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|
2484 /* Should I start playing here or pass the buck to update? */ |
|
279d0427ef26
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1
diff
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|
2485 /* Unlike SDL_SoundMixer, I think I'll do it here because |
|
279d0427ef26
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1
diff
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|
2486 * this library isn't a *total* hack and OpenAL has more |
|
279d0427ef26
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1
diff
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|
2487 * built in functionality I need, so less needs to be |
|
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1
diff
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|
2488 * controlled and directed through the update function. |
|
279d0427ef26
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1
diff
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|
2489 * The downside is less functionality is centralized. |
|
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|
2490 * The upside is that the update function should be |
|
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2491 * easier to maintain. |
|
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|
2492 */ |
|
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|
2493 |
|
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|
2494 /* Clear the error flag */ |
|
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|
2495 alGetError(); |
|
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diff
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|
2496 if(data->decoded_all) |
|
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|
2497 { |
|
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diff
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|
2498 /* Bind the data to the source */ |
|
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diff
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|
2499 alSourcei( |
|
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|
2500 ALmixer_Channel_List[channel].alsource, |
|
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|
2501 AL_BUFFER, |
|
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|
2502 data->buffer[0]); |
|
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diff
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|
2503 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
2504 { |
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|
2505 ALmixer_SetError("Could not bind data to source: %s", |
|
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|
2506 alGetString(error) ); |
|
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|
2507 Clean_Channel(channel); |
|
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|
2508 return -1; |
|
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|
2509 } |
|
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|
2510 |
|
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diff
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|
2511 /* Make data available if access_data is enabled */ |
|
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diff
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|
2512 Invoke_Predecoded_Channel_Data_Callback(channel, data); |
|
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|
2513 } |
|
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|
2514 else |
|
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|
2515 { |
|
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|
2516 /* Need to use the streaming buffer for binding */ |
|
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|
2517 |
|
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|
2518 ALuint bytes_returned; |
|
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|
2519 ALuint j; |
|
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diff
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|
2520 data->num_buffers_in_use=0; |
|
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|
2521 /****** MODIFICATION must go here *********/ |
|
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|
2522 /* Since buffer queuing is pushed off until here to |
|
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diff
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|
2523 * avoid buffer conflicts, we must start reading |
|
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diff
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|
2524 * data here. First we make sure we have at least one |
|
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diff
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|
2525 * packet. Then we queue up until we hit our limit. |
|
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|
2526 */ |
|
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diff
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|
2527 bytes_returned = GetMoreData( |
|
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|
2528 data, |
|
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diff
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|
2529 data->buffer[0]); |
|
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|
2530 if(0 == bytes_returned) |
|
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|
2531 { |
|
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|
2532 /* No data or error */ |
|
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|
2533 ALmixer_SetError("Could not get data for streamed PlayChannel: %s", ALmixer_GetError()); |
|
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|
2534 Clean_Channel(channel); |
|
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|
2535 return -1; |
|
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|
2536 } |
|
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|
2537 /* Increment the number of buffers in use */ |
|
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|
2538 data->num_buffers_in_use++; |
|
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|
2539 |
|
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|
2540 |
|
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|
2541 /* Now we need to fill up the rest of the buffers. |
|
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|
2542 * There is a corner case where we run out of data |
|
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|
2543 * before the last buffer is filled. |
|
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|
2544 * Stop conditions are we run out of |
|
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|
2545 * data or we max out our preload buffers. |
|
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|
2546 */ |
|
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|
2547 |
|
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|
2548 /* |
|
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|
2549 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", 0, data->buffer[0]); |
|
6
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|
2550 */ |
|
2
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|
2551 for(j=1; j<data->num_startup_buffers; j++) |
|
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|
2552 { |
|
6
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|
2553 /* |
|
2
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|
2554 fprintf(stderr, "Filling buffer #%d (AL id is %d)\n", j, data->buffer[j]); |
|
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diff
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|
2555 fprintf(stderr, ">>>>>>>>>>>>>>>>>>HACK for GetMoreData2\n"); |
|
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|
2556 */ |
|
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|
2557 bytes_returned = GetMoreData( |
|
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|
2558 data, |
|
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|
2559 data->buffer[j]); |
|
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|
2560 /* |
|
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diff
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|
2561 * This might be a problem. I made a mistake with the types. I accidentally |
|
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diff
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|
2562 * made the bytes returned an ALint and returned -1 on error. |
|
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diff
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|
2563 * Bytes returned should be a ALuint, so now I no longer have a -1 case |
|
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|
2564 * to check. I hope I didn't break anything here |
|
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|
2565 */ |
|
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|
2566 #if 0 |
|
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|
2567 if(bytes_returned < 0) |
|
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|
2568 { |
|
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|
2569 /* Error found */ |
|
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diff
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|
2570 ALmixer_SetError("Could not get data for additional startup buffers for PlayChannel: %s", ALmixer_GetError()); |
|
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|
2571 /* We'll continue on because we do have some valid data */ |
|
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|
2572 ret_flag = -1; |
|
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|
2573 break; |
|
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|
2574 } |
|
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|
2575 else if(0 == bytes_returned) |
|
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|
2576 #endif |
|
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|
2577 if(0 == bytes_returned) |
|
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|
2578 { |
|
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|
2579 /* No more data to buffer */ |
|
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|
2580 /* Check for loops */ |
|
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|
2581 if( ALmixer_Channel_List[channel].loops != 0 ) |
|
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|
2582 { |
|
6
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|
2583 /* |
|
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|
2584 fprintf(stderr, "Need to rewind. In RAMPUP, handling loop\n"); |
|
6
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|
2585 */ |
|
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|
2586 if(0 == Sound_Rewind(data->sample)) |
|
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|
2587 { |
|
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|
2588 fprintf(stderr, "error in rewind\n"); |
|
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|
2589 ALmixer_SetError( Sound_GetError() ); |
|
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|
2590 ALmixer_Channel_List[channel].loops = 0; |
|
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|
2591 ret_flag = -1; |
|
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|
2592 /* We'll continue on because we do have some valid data */ |
|
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|
2593 break; |
|
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|
2594 } |
|
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|
2595 /* Remember to reset the data->eof flag */ |
|
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|
2596 data->eof = 0; |
|
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|
2597 if(ALmixer_Channel_List[channel].loops > 0) |
|
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|
2598 { |
|
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|
2599 ALmixer_Channel_List[channel].loops--; |
|
6
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|
2600 /* |
|
2
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|
2601 fprintf(stderr, "Inside 000 >>>>>>>>>>Loops=%d\n", ALmixer_Channel_List[channel].loops); |
|
6
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|
2602 */ |
|
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|
2603 } |
|
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|
2604 /* Would like to redo the loop, but due to |
|
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|
2605 * Sound_Rewind() bugs, we would risk falling |
|
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|
2606 * into an infinite loop |
|
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|
2607 */ |
|
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diff
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|
2608 bytes_returned = GetMoreData( |
|
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diff
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|
2609 data, |
|
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1
diff
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|
2610 data->buffer[j]); |
|
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|
2611 if(bytes_returned <= 0) |
|
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|
2612 { |
|
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|
2613 ALmixer_SetError("Could not get data: %s", ALmixer_GetError()); |
|
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|
2614 /* We'll continue on because we do have some valid data */ |
|
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|
2615 ret_flag = -1; |
|
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|
2616 break; |
|
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|
2617 } |
|
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|
2618 } |
|
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|
2619 else |
|
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|
2620 { |
|
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|
2621 /* No loops to do so quit here */ |
|
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2622 break; |
|
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2623 } |
|
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|
2624 } |
|
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|
2625 /* Increment the number of buffers in use */ |
|
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|
2626 data->num_buffers_in_use++; |
|
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2627 } |
|
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|
2628 /* |
|
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|
2629 fprintf(stderr, "In PlayChannel, about to queue: source=%d, num_buffers_in_use=%d\n", |
|
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|
2630 ALmixer_Channel_List[channel].alsource, |
|
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2631 data->num_buffers_in_use); |
|
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2632 */ |
|
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|
2633 |
|
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2634 alSourceQueueBuffers( |
|
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|
2635 ALmixer_Channel_List[channel].alsource, |
|
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|
2636 data->num_buffers_in_use, |
|
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|
2637 data->buffer); |
|
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2638 if((error = alGetError()) != AL_NO_ERROR) |
|
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2639 { |
|
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|
2640 ALmixer_SetError("Could not bind data to source: %s", |
|
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|
2641 alGetString(error) ); |
|
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|
2642 Clean_Channel(channel); |
|
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2643 return -1; |
|
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|
2644 } |
|
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|
2645 /* This is part of the hideous Nvidia workaround. In order to figure out |
|
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2646 * which buffer to show during callbacks (for things like |
|
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|
2647 * o-scopes), I must keep a copy of the buffers that are queued in my own |
|
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2648 * data structure. This code will be called only if |
|
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2649 * "access_data" was set, indicated by whether the queue is NULL. |
|
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|
2650 */ |
|
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|
2651 if(data->circular_buffer_queue != NULL) |
|
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|
2652 { |
|
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|
2653 ALuint k; |
|
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2654 ALuint queue_ret_flag; |
|
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|
2655 for(k=0; k<data->num_buffers_in_use; k++) |
|
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|
2656 { |
|
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diff
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|
2657 // fprintf(stderr, "56c: CircularQueue_PushBack.\n"); |
|
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|
2658 queue_ret_flag = CircularQueueUnsignedInt_PushBack(data->circular_buffer_queue, data->buffer[k]); |
|
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|
2659 if(0 == queue_ret_flag) |
|
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|
2660 { |
|
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diff
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|
2661 fprintf(stderr, "Serious internal error: CircularQueue could not push into queue.\n"); |
|
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|
2662 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
|
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2663 } |
|
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|
2664 /* |
|
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|
2665 else |
|
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|
2666 { |
|
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|
2667 fprintf(stderr, "Queue in PlayTimed\n"); |
|
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|
2668 CircularQueueUnsignedInt_Print(data->circular_buffer_queue); |
|
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2669 } |
|
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|
2670 */ |
|
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|
2671 } |
|
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|
2672 } |
|
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diff
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|
2673 |
|
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|
2674 |
|
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|
2675 /****** END **********/ |
|
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2676 } |
|
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diff
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|
2677 /* We have finished loading the data (predecoded or queued) |
|
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2678 * so now we can play |
|
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|
2679 */ |
|
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diff
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|
2680 alSourcePlay(ALmixer_Channel_List[channel].alsource); |
|
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diff
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|
2681 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
2682 { |
|
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|
2683 ALmixer_SetError("Play failed: %s", |
|
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|
2684 alGetString(error) ); |
|
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|
2685 Clean_Channel(channel); |
|
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|
2686 return -1; |
|
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|
2687 } |
|
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|
2688 |
|
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|
2689 /* Add to the counter that something is playing */ |
|
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|
2690 Is_Playing_global++; |
|
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diff
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2691 if(-1 == ret_flag) |
|
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2692 { |
|
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|
2693 fprintf(stderr, "BACKDOOR ERROR >>>>>>>>>>>>>>>>>>\n"); |
|
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2694 return -1; |
|
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2695 } |
|
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2696 return channel; |
|
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2697 } |
|
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|
2698 |
|
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2699 |
|
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2700 /* In case the user wants to specify a source instead of a channel, |
|
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2701 * they may use this function. This function will look up the |
|
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2702 * source-to-channel map, and convert the call into a |
|
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2703 * PlayChannelTimed() function call. |
|
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2704 * Returns the channel it's being played on. |
|
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2705 * Note: If you are prefer this method, then you need to be careful |
|
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2706 * about using PlayChannel, particularly if you request the |
|
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|
2707 * first available channels because source and channels have |
|
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|
2708 * a one-to-one mapping in this API. It is quite easy for |
|
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|
2709 * a channel/source to already be in use because of this. |
|
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2710 * In this event, an error message will be returned to you. |
|
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2711 */ |
|
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|
2712 static ALuint Internal_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
|
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|
2713 { |
|
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|
2714 ALint channel; |
|
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|
2715 ALint retval; |
|
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|
2716 if(0 == source) |
|
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|
2717 { |
|
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|
2718 retval = Internal_PlayChannelTimed(-1, data, loops, ticks); |
|
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2719 if(-1 == retval) |
|
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|
2720 { |
|
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|
2721 return 0; |
|
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|
2722 } |
|
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|
2723 else |
|
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2724 { |
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2725 return Internal_GetSource(retval); |
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2726 } |
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2727 } |
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2728 |
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2729 channel = Internal_GetChannel(source); |
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2730 if(-1 == channel) |
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2731 { |
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2732 ALmixer_SetError("Cannot Play source: %s", ALmixer_GetError()); |
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2733 return 0; |
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2734 } |
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2735 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
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2736 if(-1 == retval) |
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2737 { |
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2738 return 0; |
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2739 } |
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2740 else |
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2741 { |
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2742 return source; |
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2743 } |
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2744 /* make compiler happy */ |
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2745 return 0; |
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2746 } |
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2747 |
| 0 | 2748 |
| 2749 | |
| 2750 | |
| 2751 /* Returns the channel or number of channels actually paused */ | |
| 2752 | |
|
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2753 static ALint Internal_PauseChannel(ALint channel) |
| 0 | 2754 { |
| 2755 ALenum error; | |
| 2756 ALint state; | |
|
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2757 ALint retval = 0; |
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2758 ALint counter = 0; |
| 0 | 2759 |
| 2760 if(channel >= Number_of_Channels_global) | |
| 2761 { | |
| 2762 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2763 return -1; | |
| 2764 } | |
| 2765 | |
| 2766 if((error = alGetError()) != AL_NO_ERROR) | |
| 2767 { | |
| 2768 fprintf(stderr, "28Testing error: %s\n", | |
|
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2769 alGetString(error)); |
| 0 | 2770 } |
| 2771 /* Clear error */ | |
| 2772 alGetError(); | |
| 2773 | |
| 2774 /* If the user specified a specific channel */ | |
| 2775 if(channel >= 0) | |
| 2776 { | |
| 2777 /* only need to process channel if in use */ | |
| 2778 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 2779 { | |
| 2780 /* We don't want to repause if already | |
| 2781 * paused because the fadeout/expire | |
| 2782 * timing will get messed up | |
| 2783 */ | |
| 2784 alGetSourcei( | |
| 2785 ALmixer_Channel_List[channel].alsource, | |
| 2786 AL_SOURCE_STATE, &state | |
| 2787 ); | |
| 2788 if((error = alGetError()) != AL_NO_ERROR) | |
| 2789 { | |
| 2790 fprintf(stderr, "29Testing error: %s\n", | |
|
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2791 alGetString(error)); |
| 0 | 2792 } |
| 2793 if(AL_PLAYING == state) | |
| 2794 { | |
| 2795 /* Count the actual number of channels being paused */ | |
| 2796 counter++; | |
| 2797 | |
| 2798 alSourcePause(ALmixer_Channel_List[channel].alsource); | |
| 2799 if((error = alGetError()) != AL_NO_ERROR) | |
| 2800 { | |
| 2801 ALmixer_SetError("%s", | |
|
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2802 alGetString(error) ); |
| 0 | 2803 retval = -1; |
| 2804 } | |
| 2805 /* We need to pause the expire time count down */ | |
| 2806 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
| 2807 { | |
|
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2808 ALuint current_time = ALmixer_GetTicks(); |
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2809 ALuint diff_time; |
| 0 | 2810 diff_time = current_time - |
| 2811 ALmixer_Channel_List[channel].start_time; | |
| 2812 /* When we unpause, we will want to reset | |
| 2813 * the start time so we can continue | |
| 2814 * to base calculations off GetTicks(). | |
| 2815 * This means we need to subtract the amount | |
| 2816 * of time already used up from expire_ticks. | |
| 2817 */ | |
| 2818 ALmixer_Channel_List[channel].expire_ticks = | |
| 2819 ALmixer_Channel_List[channel].expire_ticks - | |
| 2820 diff_time; | |
| 2821 /* Because -1 is a special value, we can't | |
| 2822 * allow the time to go negative | |
| 2823 */ | |
| 2824 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
| 2825 { | |
| 2826 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 2827 } | |
| 2828 } | |
| 2829 /* Do the same as expire time for fading */ | |
| 2830 if(ALmixer_Channel_List[channel].fade_enabled) | |
| 2831 { | |
|
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2832 ALuint current_time = ALmixer_GetTicks(); |
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2833 ALuint diff_time; |
| 0 | 2834 diff_time = current_time - |
| 2835 ALmixer_Channel_List[channel].fade_start_time; | |
| 2836 /* When we unpause, we will want to reset | |
| 2837 * the start time so we can continue | |
| 2838 * to base calculations off GetTicks(). | |
| 2839 * This means we need to subtract the amount | |
| 2840 * of time already used up from expire_ticks. | |
| 2841 */ | |
| 2842 ALmixer_Channel_List[channel].fade_expire_ticks = | |
| 2843 ALmixer_Channel_List[channel].fade_expire_ticks - | |
| 2844 diff_time; | |
| 2845 /* Don't allow the time to go negative */ | |
| 2846 if(ALmixer_Channel_List[channel].expire_ticks < 0) | |
| 2847 { | |
| 2848 ALmixer_Channel_List[channel].expire_ticks = 0; | |
| 2849 } | |
| 2850 } /* End fade check */ | |
| 2851 } /* End if PLAYING */ | |
| 2852 } /* End If in use */ | |
| 2853 } /* End specific channel */ | |
| 2854 /* The user wants to halt all channels */ | |
| 2855 else | |
| 2856 { | |
|
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2857 ALint i; |
| 0 | 2858 for(i=0; i<Number_of_Channels_global; i++) |
| 2859 { | |
| 2860 /* only need to process channel if in use */ | |
| 2861 if(ALmixer_Channel_List[i].channel_in_use) | |
| 2862 { | |
| 2863 /* We don't want to repause if already | |
| 2864 * paused because the fadeout/expire | |
| 2865 * timing will get messed up | |
| 2866 */ | |
| 2867 alGetSourcei( | |
| 2868 ALmixer_Channel_List[i].alsource, | |
| 2869 AL_SOURCE_STATE, &state | |
| 2870 ); | |
| 2871 if((error = alGetError()) != AL_NO_ERROR) | |
| 2872 { | |
| 2873 fprintf(stderr, "30Testing error: %s\n", | |
|
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2874 alGetString(error)); |
| 0 | 2875 } |
| 2876 if(AL_PLAYING == state) | |
| 2877 { | |
| 2878 /* Count the actual number of channels being paused */ | |
| 2879 counter++; | |
| 2880 | |
| 2881 alSourcePause(ALmixer_Channel_List[i].alsource); | |
| 2882 if((error = alGetError()) != AL_NO_ERROR) | |
| 2883 { | |
| 2884 ALmixer_SetError("%s", | |
|
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2885 alGetString(error) ); |
| 0 | 2886 retval = -1; |
| 2887 } | |
| 2888 /* We need to pause the expire time count down */ | |
| 2889 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 2890 { | |
|
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2891 ALuint current_time = ALmixer_GetTicks(); |
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2892 ALuint diff_time; |
| 0 | 2893 diff_time = current_time - |
| 2894 ALmixer_Channel_List[i].start_time; | |
| 2895 /* When we unpause, we will want to reset | |
| 2896 * the start time so we can continue | |
| 2897 * to base calculations off GetTicks(). | |
| 2898 * This means we need to subtract the amount | |
| 2899 * of time already used up from expire_ticks. | |
| 2900 */ | |
| 2901 ALmixer_Channel_List[i].expire_ticks = | |
| 2902 ALmixer_Channel_List[i].expire_ticks - | |
| 2903 diff_time; | |
| 2904 /* Because -1 is a special value, we can't | |
| 2905 * allow the time to go negative | |
| 2906 */ | |
| 2907 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
| 2908 { | |
| 2909 ALmixer_Channel_List[i].expire_ticks = 0; | |
| 2910 } | |
| 2911 } | |
| 2912 /* Do the same as expire time for fading */ | |
| 2913 if(ALmixer_Channel_List[i].fade_enabled) | |
| 2914 { | |
|
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2915 ALuint current_time = ALmixer_GetTicks(); |
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2916 ALuint diff_time; |
| 0 | 2917 diff_time = current_time - |
| 2918 ALmixer_Channel_List[i].fade_start_time; | |
| 2919 /* When we unpause, we will want to reset | |
| 2920 * the start time so we can continue | |
| 2921 * to base calculations off GetTicks(). | |
| 2922 * This means we need to subtract the amount | |
| 2923 * of time already used up from expire_ticks. | |
| 2924 */ | |
| 2925 ALmixer_Channel_List[i].fade_expire_ticks = | |
| 2926 ALmixer_Channel_List[i].fade_expire_ticks - | |
| 2927 diff_time; | |
| 2928 /* Don't allow the time to go negative */ | |
| 2929 if(ALmixer_Channel_List[i].expire_ticks < 0) | |
| 2930 { | |
| 2931 ALmixer_Channel_List[i].expire_ticks = 0; | |
| 2932 } | |
| 2933 } /* End fade check */ | |
| 2934 } /* End if PLAYING */ | |
| 2935 } /* End channel in use */ | |
| 2936 } /* End for-loop */ | |
| 2937 } | |
| 2938 if(-1 == retval) | |
| 2939 { | |
| 2940 return -1; | |
| 2941 } | |
| 2942 return counter; | |
| 2943 } | |
| 2944 | |
| 2945 /* Returns the channel or number of channels actually paused */ | |
|
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2946 static ALint Internal_PauseSource(ALuint source) |
|
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2947 { |
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2948 ALint channel; |
| 0 | 2949 if(0 == source) |
| 2950 { | |
| 2951 return Internal_PauseChannel(-1); | |
| 2952 } | |
| 2953 | |
| 2954 channel = Internal_GetChannel(source); | |
| 2955 if(-1 == channel) | |
| 2956 { | |
| 2957 ALmixer_SetError("Cannot pause source: %s", ALmixer_GetError()); | |
| 2958 return -1; | |
| 2959 } | |
| 2960 return Internal_PauseChannel(channel); | |
| 2961 } | |
| 2962 | |
| 2963 | |
| 2964 | |
|
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2965 static ALint Internal_ResumeChannel(ALint channel) |
| 0 | 2966 { |
| 2967 ALint state; | |
| 2968 ALenum error; | |
|
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2969 ALint retval = 0; |
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2970 ALint counter = 0; |
| 0 | 2971 |
| 2972 if(channel >= Number_of_Channels_global) | |
| 2973 { | |
| 2974 ALmixer_SetError("Cannot pause channel %d because it exceeds maximum number of channels (%d)\n", channel, Number_of_Channels_global); | |
| 2975 return -1; | |
| 2976 } | |
| 2977 | |
| 2978 if((error = alGetError()) != AL_NO_ERROR) | |
| 2979 { | |
| 2980 fprintf(stderr, "31Testing error: %s\n", | |
|
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2981 alGetString(error)); |
| 0 | 2982 } |
| 2983 /* Clear error */ | |
| 2984 alGetError(); | |
| 2985 | |
| 2986 /* If the user specified a specific channel */ | |
| 2987 if(channel >= 0) | |
| 2988 { | |
| 2989 /* only need to process channel if in use */ | |
| 2990 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 2991 { | |
| 2992 alGetSourcei( | |
| 2993 ALmixer_Channel_List[channel].alsource, | |
| 2994 AL_SOURCE_STATE, &state | |
| 2995 ); | |
| 2996 if((error = alGetError()) != AL_NO_ERROR) | |
| 2997 { | |
| 2998 fprintf(stderr, "32Testing error: %s\n", | |
|
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2999 alGetString(error)); |
| 0 | 3000 } |
| 3001 if(AL_PAUSED == state) | |
| 3002 { | |
| 3003 /* Count the actual number of channels resumed */ | |
| 3004 counter++; | |
| 3005 | |
| 3006 /* We need to resume the expire time count down */ | |
| 3007 if(ALmixer_Channel_List[channel].expire_ticks != -1) | |
| 3008 { | |
|
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3009 ALmixer_Channel_List[channel].start_time = ALmixer_GetTicks(); |
| 0 | 3010 } |
| 3011 /* Do the same as expire time for fading */ | |
| 3012 if(ALmixer_Channel_List[channel].fade_enabled) | |
| 3013 { | |
|
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3014 ALmixer_Channel_List[channel].fade_start_time = ALmixer_GetTicks(); |
| 0 | 3015 } |
| 3016 | |
| 3017 alSourcePlay(ALmixer_Channel_List[channel].alsource); | |
| 3018 if((error = alGetError()) != AL_NO_ERROR) | |
| 3019 { | |
| 3020 ALmixer_SetError("%s", | |
|
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3021 alGetString(error) ); |
| 0 | 3022 retval = -1; |
| 3023 } | |
| 3024 } | |
| 3025 } | |
| 3026 } | |
| 3027 /* The user wants to halt all channels */ | |
| 3028 else | |
| 3029 { | |
|
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3030 ALint i; |
| 0 | 3031 for(i=0; i<Number_of_Channels_global; i++) |
| 3032 { | |
| 3033 /* only need to process channel if in use */ | |
| 3034 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3035 { | |
| 3036 alGetSourcei( | |
| 3037 ALmixer_Channel_List[i].alsource, | |
| 3038 AL_SOURCE_STATE, &state | |
| 3039 ); | |
| 3040 if((error = alGetError()) != AL_NO_ERROR) | |
| 3041 { | |
| 3042 fprintf(stderr, "33Testing error: %s\n", | |
|
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3043 alGetString(error)); |
| 0 | 3044 } |
| 3045 if(AL_PAUSED == state) | |
| 3046 { | |
| 3047 /* Count the actual number of channels resumed */ | |
| 3048 counter++; | |
| 3049 | |
| 3050 /* We need to resume the expire time count down */ | |
| 3051 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 3052 { | |
|
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3053 ALmixer_Channel_List[i].start_time = ALmixer_GetTicks(); |
| 0 | 3054 } |
| 3055 /* Do the same as expire time for fading */ | |
| 3056 if(ALmixer_Channel_List[i].fade_enabled) | |
| 3057 { | |
|
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3058 ALmixer_Channel_List[i].fade_start_time = ALmixer_GetTicks(); |
| 0 | 3059 } |
| 3060 | |
| 3061 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 3062 if((error = alGetError()) != AL_NO_ERROR) | |
| 3063 { | |
| 3064 ALmixer_SetError("%s", | |
|
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3065 alGetString(error) ); |
| 0 | 3066 retval = -1; |
| 3067 } | |
| 3068 } | |
| 3069 } | |
| 3070 } | |
| 3071 } | |
| 3072 if(-1 == retval) | |
| 3073 { | |
| 3074 return -1; | |
| 3075 } | |
| 3076 return counter; | |
| 3077 } | |
| 3078 | |
| 3079 | |
|
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3080 static ALint Internal_ResumeSource(ALuint source) |
|
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3081 { |
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3082 ALint channel; |
| 0 | 3083 if(0 == source) |
| 3084 { | |
| 3085 return Internal_ResumeChannel(-1); | |
| 3086 } | |
| 3087 | |
| 3088 channel = Internal_GetChannel(source); | |
| 3089 if(-1 == channel) | |
| 3090 { | |
| 3091 ALmixer_SetError("Cannot resume source: %s", ALmixer_GetError()); | |
| 3092 return -1; | |
| 3093 } | |
| 3094 return Internal_ResumeChannel(channel); | |
| 3095 } | |
| 3096 | |
| 3097 | |
| 3098 /* Might consider setting eof to 0 as a "feature" | |
| 3099 * This will allow seek to end to stay there because | |
| 3100 * Play automatically rewinds if at the end */ | |
|
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3101 static ALint Internal_SeekData(ALmixer_Data* data, ALuint msec) |
|
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3102 { |
|
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3103 ALint retval; |
| 0 | 3104 |
| 3105 if(NULL == data) | |
| 3106 { | |
| 3107 ALmixer_SetError("Cannot Seek because data is NULL"); | |
| 3108 return -1; | |
| 3109 } | |
| 3110 | |
| 3111 /* Seek for predecoded files involves moving the chunk pointer around */ | |
| 3112 if(data->decoded_all) | |
| 3113 { | |
|
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3114 ALuint byte_position; |
| 0 | 3115 |
| 3116 /* OpenAL doesn't seem to like it if I change the buffer | |
| 3117 * while playing (crashes), so I must require that Seek only | |
| 3118 * be done when the data is not in use. | |
| 3119 * Since data may be shared among multiple sources, | |
| 3120 * I can't shut them down myself, so I have to return an error. | |
| 3121 */ | |
| 3122 if(data->in_use) | |
| 3123 { | |
| 3124 ALmixer_SetError("Cannot seek on predecoded data while instances are playing"); | |
| 3125 return -1; | |
| 3126 } | |
| 3127 #if 0 | |
| 3128 #if defined(DISABLE_PREDECODED_SEEK) | |
| 3129 ALmixer_SetError("Seek support for predecoded samples was not compiled in"); | |
| 3130 return -1; | |
| 3131 | |
| 3132 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 3133 /* By default, ALmixer frees the Sound_Sample for predecoded | |
| 3134 * samples because of the potential memory waste. | |
| 3135 * However, to seek a sample, we need to have a full | |
| 3136 * copy of the data around. So the strategy is to | |
| 3137 * recreate a hackish Sound_Sample to be used for seeking | |
| 3138 * purposes. If Sound_Sample is NULL, we will reallocate | |
| 3139 * memory for it and then procede as if everything | |
| 3140 * was normal. | |
| 3141 */ | |
| 3142 if(NULL == data->sample) | |
| 3143 { | |
| 3144 if( -1 == Reconstruct_Sound_Sample(data) ) | |
| 3145 { | |
| 3146 return -1; | |
| 3147 } | |
| 3148 } | |
| 3149 #endif | |
| 3150 #endif | |
| 3151 /* If access_data was set, then we still have the | |
| 3152 * Sound_Sample and we can move around in the data. | |
| 3153 * If it was not set, the data has been freed and we | |
| 3154 * cannot do anything because there is no way to | |
| 3155 * recover the data because OpenAL won't let us | |
| 3156 * get access to the buffers | |
| 3157 */ | |
| 3158 if(NULL == data->sample) | |
| 3159 { | |
| 3160 ALmixer_SetError("Cannot seek because access_data flag was set false when data was initialized"); | |
| 3161 return -1; | |
| 3162 } | |
| 3163 | |
| 3164 byte_position = Convert_Msec_To_Byte_Pos(&data->sample->desired, msec); | |
| 3165 return( Set_Predecoded_Seek_Position(data, byte_position) ); | |
| 3166 } | |
| 3167 else | |
| 3168 { | |
| 3169 /* Reset eof flag?? */ | |
| 3170 data->eof = 0; | |
| 3171 retval = Sound_Seek(data->sample, msec); | |
| 3172 if(0 == retval) | |
| 3173 { | |
| 3174 ALmixer_SetError(Sound_GetError()); | |
| 3175 | |
| 3176 fprintf(stderr, "Sound seek error: %s\n", ALmixer_GetError()); | |
| 3177 /* Try rewinding to clean up? */ | |
| 3178 /* | |
| 3179 Internal_RewindData(data); | |
| 3180 */ | |
| 3181 return -1; | |
| 3182 } | |
| 3183 return 0; | |
| 3184 } | |
| 3185 | |
| 3186 return 0; | |
| 3187 } | |
| 3188 | |
| 3189 | |
| 3190 | |
|
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3191 static ALint Internal_FadeInChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
| 0 | 3192 { |
| 3193 ALfloat value; | |
| 3194 ALenum error; | |
| 3195 ALfloat original_value; | |
|
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3196 ALuint current_time = ALmixer_GetTicks(); |
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3197 ALint retval; |
| 0 | 3198 |
| 3199 | |
| 3200 | |
| 3201 if(channel >= Number_of_Channels_global) | |
| 3202 { | |
| 3203 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 3204 return -1; | |
| 3205 } | |
| 3206 /* Let's call PlayChannelTimed to do the job. | |
| 3207 * There are two catches: | |
| 3208 * First is that we must set the volumes before the play call(s). | |
| 3209 * Second is that we must initialize the channel values | |
| 3210 */ | |
| 3211 | |
| 3212 if(channel < 0) | |
| 3213 { | |
| 3214 /* This might cause a problem for threads/race conditions. | |
| 3215 * We need to set the volume on an unknown channel, | |
| 3216 * so we need to request a channel first. Remember | |
| 3217 * that requesting a channel doesn't lock and it | |
| 3218 * could be surrendered to somebody else before we claim it. | |
| 3219 */ | |
| 3220 channel = Internal_GetChannel(0); | |
| 3221 if(-1 == channel) | |
| 3222 { | |
| 3223 return -1; | |
| 3224 } | |
| 3225 } | |
| 3226 else if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3227 { | |
| 3228 ALmixer_SetError("Channel %d is already in use", channel); | |
| 3229 return -1; | |
| 3230 } | |
| 3231 | |
| 3232 | |
| 3233 /* Get the original volume in case of a problem */ | |
| 3234 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3235 AL_GAIN, &original_value); |
| 0 | 3236 |
| 3237 if((error = alGetError()) != AL_NO_ERROR) | |
| 3238 { | |
| 3239 fprintf(stderr, "35Testing error: %s\n", | |
|
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3240 alGetString(error)); |
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3241 } |
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3242 ALmixer_Channel_List[channel].fade_end_volume = original_value; |
| 0 | 3243 |
| 3244 /* Get the Min volume */ | |
| 3245 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3246 AL_MIN_GAIN, &value); | |
| 3247 if((error = alGetError()) != AL_NO_ERROR) | |
| 3248 { | |
| 3249 fprintf(stderr, "36Testing error: %s\n", | |
|
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3250 alGetString(error)); |
| 0 | 3251 } |
| 3252 ALmixer_Channel_List[channel].fade_start_volume = value; | |
| 3253 | |
| 3254 /* Set the actual volume */ | |
| 3255 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3256 AL_GAIN, value); |
| 0 | 3257 if((error = alGetError()) != AL_NO_ERROR) |
| 3258 { | |
| 3259 fprintf(stderr, "37Testing error: %s\n", | |
|
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3260 alGetString(error)); |
| 0 | 3261 } |
| 3262 | |
| 3263 | |
| 3264 /* Now call PlayChannelTimed */ | |
| 3265 retval = Internal_PlayChannelTimed(channel, data, loops, expire_ticks); | |
| 3266 if(-1 == retval) | |
| 3267 { | |
| 3268 /* Chance of failure is actually pretty high since | |
| 3269 * a channel might already be in use or streamed | |
| 3270 * data can be shared | |
| 3271 */ | |
| 3272 /* Restore the original value to avoid accidental | |
| 3273 * distruption of playback | |
| 3274 */ | |
| 3275 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3276 AL_GAIN, original_value); |
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3277 if((error = alGetError()) != AL_NO_ERROR) |
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3278 { |
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3279 fprintf(stderr, "38Testing error: %s\n", |
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3280 alGetString(error)); |
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3281 } |
| 0 | 3282 return retval; |
| 3283 } | |
| 3284 | |
| 3285 /* We can't accept 0 as a value because of div-by-zero. | |
| 3286 * If zero, just call PlayChannelTimed at normal | |
| 3287 * volume | |
| 3288 */ | |
| 3289 if(0 == fade_ticks) | |
| 3290 { | |
| 3291 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3292 AL_GAIN, |
| 0 | 3293 ALmixer_Channel_List[channel].fade_end_volume |
| 3294 ); | |
|
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3295 if((error = alGetError()) != AL_NO_ERROR) |
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3296 { |
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3297 fprintf(stderr, "39Testing error: %s\n", |
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3298 alGetString(error)); |
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3299 } |
| 0 | 3300 |
| 3301 return retval; | |
| 3302 } | |
| 3303 | |
| 3304 /* Enable fading effects via the flag */ | |
| 3305 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3306 /* Set fade start time */ | |
| 3307 ALmixer_Channel_List[channel].fade_start_time | |
| 3308 = ALmixer_Channel_List[channel].start_time; | |
| 3309 /* Set the fade expire ticks */ | |
| 3310 ALmixer_Channel_List[channel].fade_expire_ticks = fade_ticks; | |
| 3311 | |
| 3312 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3313 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / fade_ticks; | |
| 3314 | |
| 3315 return retval; | |
| 3316 | |
| 3317 } | |
| 3318 | |
| 3319 | |
|
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3320 static ALuint Internal_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
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|
3321 { |
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|
3322 ALint channel; |
|
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3323 ALint retval; |
| 0 | 3324 if(0 == source) |
| 3325 { | |
| 3326 retval = Internal_FadeInChannelTimed(-1, data, loops, fade_ticks, expire_ticks); | |
| 3327 if(-1 == retval) | |
| 3328 { | |
| 3329 return 0; | |
| 3330 } | |
| 3331 else | |
| 3332 { | |
| 3333 return Internal_GetSource(retval); | |
| 3334 } | |
| 3335 } | |
| 3336 | |
| 3337 channel = Internal_GetChannel(source); | |
| 3338 if(-1 == channel) | |
| 3339 { | |
| 3340 ALmixer_SetError("Cannot FadeIn source: %s", ALmixer_GetError()); | |
| 3341 return 0; | |
| 3342 } | |
| 3343 retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks); | |
| 3344 if(-1 == retval) | |
| 3345 { | |
| 3346 return 0; | |
| 3347 } | |
| 3348 else | |
| 3349 { | |
| 3350 return source; | |
| 3351 } | |
| 3352 /* make compiler happy */ | |
| 3353 return 0; | |
| 3354 } | |
| 3355 | |
| 3356 | |
| 3357 | |
| 3358 | |
| 3359 /* Will fade out currently playing channels. | |
| 3360 * It starts at the current volume level and goes down */ | |
|
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3361 static ALint Internal_FadeOutChannel(ALint channel, ALuint ticks) |
| 0 | 3362 { |
| 3363 ALfloat value; | |
| 3364 ALenum error; | |
|
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3365 ALuint current_time = ALmixer_GetTicks(); |
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|
3366 ALuint counter = 0; |
| 0 | 3367 |
| 3368 /* We can't accept 0 as a value because of div-by-zero. | |
| 3369 * If zero, just call Halt at normal | |
| 3370 * volume | |
| 3371 */ | |
| 3372 if(0 == ticks) | |
| 3373 { | |
|
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|
3374 return Internal_HaltChannel(channel, AL_FALSE); |
| 0 | 3375 } |
| 3376 | |
| 3377 | |
| 3378 if(channel >= Number_of_Channels_global) | |
| 3379 { | |
| 3380 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 3381 return -1; | |
| 3382 } | |
| 3383 | |
| 3384 if(channel >= 0) | |
| 3385 { | |
| 3386 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3387 { | |
| 3388 /* Get the current volume */ | |
| 3389 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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3390 AL_GAIN, &value); |
| 0 | 3391 ALmixer_Channel_List[channel].fade_start_volume = value; |
| 3392 if((error = alGetError()) != AL_NO_ERROR) | |
| 3393 { | |
| 3394 fprintf(stderr, "40Testing error: %s\n", | |
|
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3395 alGetString(error)); |
| 0 | 3396 } |
| 3397 | |
| 3398 /* Get the Min volume */ | |
| 3399 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3400 AL_MIN_GAIN, &value); | |
| 3401 if((error = alGetError()) != AL_NO_ERROR) | |
| 3402 { | |
| 3403 fprintf(stderr, "41Testing error: %s\n", | |
|
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3404 alGetString(error)); |
| 0 | 3405 } |
| 3406 ALmixer_Channel_List[channel].fade_end_volume = value; | |
| 3407 | |
| 3408 /* Set expire start time */ | |
| 3409 ALmixer_Channel_List[channel].start_time = current_time; | |
| 3410 /* Set the expire ticks */ | |
| 3411 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
| 3412 /* Set fade start time */ | |
| 3413 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
| 3414 /* Set the fade expire ticks */ | |
| 3415 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
| 3416 /* Enable fading effects via the flag */ | |
| 3417 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3418 | |
| 3419 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3420 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
| 3421 | |
| 3422 counter++; | |
| 3423 } | |
| 3424 } | |
| 3425 /* Else need to fade out all channels */ | |
| 3426 else | |
| 3427 { | |
|
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3428 ALint i; |
| 0 | 3429 for(i=0; i<Number_of_Channels_global; i++) |
| 3430 { | |
| 3431 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3432 { | |
| 3433 /* Get the current volume */ | |
| 3434 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
3435 AL_GAIN, &value); |
| 0 | 3436 ALmixer_Channel_List[i].fade_start_volume = value; |
| 3437 if((error = alGetError()) != AL_NO_ERROR) | |
| 3438 { | |
| 3439 fprintf(stderr, "42Testing error: %s\n", | |
|
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3440 alGetString(error)); |
| 0 | 3441 } |
| 3442 | |
| 3443 /* Get the Min volume */ | |
| 3444 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 3445 AL_MIN_GAIN, &value); | |
| 3446 if((error = alGetError()) != AL_NO_ERROR) | |
| 3447 { | |
| 3448 fprintf(stderr, "43Testing error: %s\n", | |
|
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3449 alGetString(error)); |
| 0 | 3450 } |
| 3451 ALmixer_Channel_List[i].fade_end_volume = value; | |
| 3452 | |
| 3453 /* Set expire start time */ | |
| 3454 ALmixer_Channel_List[i].start_time = current_time; | |
| 3455 /* Set the expire ticks */ | |
| 3456 ALmixer_Channel_List[i].expire_ticks = ticks; | |
| 3457 /* Set fade start time */ | |
| 3458 ALmixer_Channel_List[i].fade_start_time = current_time; | |
| 3459 /* Set the fade expire ticks */ | |
| 3460 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
| 3461 /* Enable fading effects via the flag */ | |
| 3462 ALmixer_Channel_List[i].fade_enabled = 1; | |
| 3463 | |
| 3464 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3465 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
| 3466 | |
| 3467 counter++; | |
| 3468 } | |
| 3469 } /* End for loop */ | |
| 3470 } | |
| 3471 return counter; | |
| 3472 } | |
| 3473 | |
| 3474 | |
|
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|
3475 static ALint Internal_FadeOutSource(ALuint source, ALuint ticks) |
|
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|
3476 { |
|
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|
3477 ALint channel; |
| 0 | 3478 if(0 == source) |
| 3479 { | |
| 3480 return Internal_FadeOutChannel(-1, ticks); | |
| 3481 } | |
| 3482 | |
| 3483 channel = Internal_GetChannel(source); | |
| 3484 if(-1 == channel) | |
| 3485 { | |
| 3486 ALmixer_SetError("Cannot FadeOut source: %s", ALmixer_GetError()); | |
| 3487 return -1; | |
| 3488 } | |
| 3489 return Internal_FadeOutChannel(channel, ticks); | |
| 3490 } | |
| 3491 | |
| 3492 | |
| 3493 /* Will fade currently playing channels. | |
| 3494 * It starts at the current volume level and go to target | |
| 3495 * Only affects channels that are playing | |
| 3496 */ | |
|
2
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|
3497 static ALint Internal_FadeChannel(ALint channel, ALuint ticks, ALfloat volume) |
| 0 | 3498 { |
| 3499 ALfloat value; | |
| 3500 ALenum error; | |
|
2
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|
3501 ALuint current_time = ALmixer_GetTicks(); |
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|
3502 ALuint counter = 0; |
| 0 | 3503 |
| 3504 if(channel >= Number_of_Channels_global) | |
| 3505 { | |
| 3506 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 3507 return -1; | |
| 3508 } | |
| 3509 | |
| 3510 if(channel >= 0) | |
| 3511 { | |
| 3512 if(volume < ALmixer_Channel_List[channel].min_volume) | |
| 3513 { | |
| 3514 volume = ALmixer_Channel_List[channel].min_volume; | |
| 3515 } | |
| 3516 else if(volume > ALmixer_Channel_List[channel].max_volume) | |
| 3517 { | |
| 3518 volume = ALmixer_Channel_List[channel].max_volume; | |
| 3519 } | |
| 3520 | |
| 3521 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 3522 { | |
| 3523 if(ticks > 0) | |
| 3524 { | |
| 3525 /* Get the current volume */ | |
| 3526 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3527 AL_GAIN, &value); |
| 0 | 3528 if((error = alGetError()) != AL_NO_ERROR) |
| 3529 { | |
| 3530 fprintf(stderr, "44Testing error: %s\n", | |
|
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|
3531 alGetString(error)); |
| 0 | 3532 } |
| 3533 ALmixer_Channel_List[channel].fade_start_volume = value; | |
| 3534 | |
| 3535 /* Set the target volume */ | |
| 3536 ALmixer_Channel_List[channel].fade_end_volume = volume; | |
| 3537 | |
| 3538 /* Set fade start time */ | |
| 3539 ALmixer_Channel_List[channel].fade_start_time = current_time; | |
| 3540 /* Set the fade expire ticks */ | |
| 3541 ALmixer_Channel_List[channel].fade_expire_ticks = ticks; | |
| 3542 /* Enable fading effects via the flag */ | |
| 3543 ALmixer_Channel_List[channel].fade_enabled = 1; | |
| 3544 | |
| 3545 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3546 ALmixer_Channel_List[channel].fade_inv_time = 1.0f / ticks; | |
| 3547 } | |
| 3548 else | |
| 3549 { | |
| 3550 alSourcef(ALmixer_Channel_List[channel].alsource, | |
|
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|
3551 AL_GAIN, volume); |
| 0 | 3552 if((error = alGetError()) != AL_NO_ERROR) |
| 3553 { | |
| 3554 fprintf(stderr, "45Testing error: %s\n", | |
|
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|
3555 alGetString(error)); |
| 0 | 3556 } |
| 3557 } | |
| 3558 counter++; | |
| 3559 } | |
| 3560 } | |
| 3561 /* Else need to fade out all channels */ | |
| 3562 else | |
| 3563 { | |
|
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|
3564 ALint i; |
| 0 | 3565 for(i=0; i<Number_of_Channels_global; i++) |
| 3566 { | |
| 3567 if(volume < ALmixer_Channel_List[i].min_volume) | |
| 3568 { | |
| 3569 volume = ALmixer_Channel_List[i].min_volume; | |
| 3570 } | |
| 3571 else if(volume > ALmixer_Channel_List[i].max_volume) | |
| 3572 { | |
| 3573 volume = ALmixer_Channel_List[i].max_volume; | |
| 3574 } | |
| 3575 | |
| 3576 if(ALmixer_Channel_List[i].channel_in_use) | |
| 3577 { | |
| 3578 if(ticks > 0) | |
| 3579 { | |
| 3580 /* Get the current volume */ | |
| 3581 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
|
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|
3582 AL_GAIN, &value); |
| 0 | 3583 if((error = alGetError()) != AL_NO_ERROR) |
| 3584 { | |
| 3585 fprintf(stderr, "46Testing error: %s\n", | |
|
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|
3586 alGetString(error)); |
| 0 | 3587 } |
| 3588 ALmixer_Channel_List[i].fade_start_volume = value; | |
| 3589 | |
| 3590 /* Set target volume */ | |
| 3591 ALmixer_Channel_List[i].fade_end_volume = volume; | |
| 3592 | |
| 3593 /* Set fade start time */ | |
| 3594 ALmixer_Channel_List[i].fade_start_time = current_time; | |
| 3595 /* Set the fade expire ticks */ | |
| 3596 ALmixer_Channel_List[i].fade_expire_ticks = ticks; | |
| 3597 /* Enable fading effects via the flag */ | |
| 3598 ALmixer_Channel_List[i].fade_enabled = 1; | |
| 3599 | |
| 3600 /* Set 1/(endtime-starttime) or 1/deltaT */ | |
| 3601 ALmixer_Channel_List[i].fade_inv_time = 1.0f / ticks; | |
| 3602 } | |
| 3603 else | |
| 3604 { | |
| 3605 alSourcef(ALmixer_Channel_List[i].alsource, | |
|
2
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|
3606 AL_GAIN, volume); |
| 0 | 3607 if((error = alGetError()) != AL_NO_ERROR) |
| 3608 { | |
| 3609 fprintf(stderr, "47Testing error: %s\n", | |
|
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|
3610 alGetString(error)); |
| 0 | 3611 } |
| 3612 } | |
| 3613 counter++; | |
| 3614 } | |
| 3615 } /* End for loop */ | |
| 3616 } | |
| 3617 return counter; | |
| 3618 } | |
| 3619 | |
|
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|
3620 static ALint Internal_FadeSource(ALuint source, ALuint ticks, ALfloat volume) |
|
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|
3621 { |
|
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|
3622 ALint channel; |
| 0 | 3623 if(0 == source) |
| 3624 { | |
| 3625 return Internal_FadeChannel(-1, ticks, volume); | |
| 3626 } | |
| 3627 | |
| 3628 channel = Internal_GetChannel(source); | |
| 3629 if(-1 == channel) | |
| 3630 { | |
| 3631 ALmixer_SetError("Cannot Fade source: %s", ALmixer_GetError()); | |
| 3632 return -1; | |
| 3633 } | |
| 3634 return Internal_FadeChannel(channel, ticks, volume); | |
| 3635 } | |
| 3636 | |
| 3637 | |
| 3638 | |
| 3639 | |
| 3640 /* Set a volume regardless if it's in use or not. | |
| 3641 */ | |
|
2
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diff
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|
3642 static ALboolean Internal_SetVolumeChannel(ALint channel, ALfloat volume) |
| 0 | 3643 { |
| 3644 ALenum error; | |
|
2
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|
3645 ALboolean retval = AL_TRUE; |
|
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|
3646 |
|
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|
3647 if(channel >= Number_of_Channels_global) |
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diff
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|
3648 { |
|
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diff
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|
3649 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
|
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diff
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|
3650 return AL_FALSE; |
|
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|
3651 } |
|
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|
3652 |
|
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diff
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|
3653 if(channel >= 0) |
|
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|
3654 { |
|
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diff
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|
3655 if(volume < 0.0f) |
|
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diff
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|
3656 { |
|
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diff
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|
3657 volume = 0.0f; |
|
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diff
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|
3658 } |
|
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diff
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|
3659 else if(volume > 1.0f) |
|
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diff
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|
3660 { |
|
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diff
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|
3661 volume = 1.0f; |
|
279d0427ef26
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diff
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|
3662 } |
|
279d0427ef26
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diff
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|
3663 alSourcef(ALmixer_Channel_List[channel].alsource, |
|
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diff
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|
3664 AL_GAIN, volume); |
|
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1
diff
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|
3665 if((error = alGetError()) != AL_NO_ERROR) |
|
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1
diff
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|
3666 { |
|
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diff
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|
3667 ALmixer_SetError("%s", |
|
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diff
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|
3668 alGetString(error) ); |
|
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diff
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|
3669 retval = AL_FALSE; |
|
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1
diff
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|
3670 } |
|
279d0427ef26
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1
diff
changeset
|
3671 } |
|
279d0427ef26
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1
diff
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|
3672 else |
|
279d0427ef26
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diff
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|
3673 { |
|
279d0427ef26
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diff
changeset
|
3674 ALint i; |
|
279d0427ef26
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diff
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|
3675 for(i=0; i<Number_of_Channels_global; i++) |
|
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diff
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|
3676 { |
|
279d0427ef26
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diff
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|
3677 if(volume < 0.0f) |
|
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1
diff
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|
3678 { |
|
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diff
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|
3679 volume = 0.0f; |
|
279d0427ef26
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1
diff
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|
3680 } |
|
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diff
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|
3681 else if(volume > 1.0f) |
|
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diff
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|
3682 { |
|
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diff
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|
3683 volume = 1.0f; |
|
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diff
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|
3684 } |
|
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diff
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|
3685 alSourcef(ALmixer_Channel_List[i].alsource, |
|
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diff
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|
3686 AL_GAIN, volume); |
|
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diff
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|
3687 if((error = alGetError()) != AL_NO_ERROR) |
|
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|
3688 { |
|
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diff
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|
3689 ALmixer_SetError("%s", |
|
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diff
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|
3690 alGetString(error) ); |
|
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diff
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|
3691 retval = AL_FALSE; |
|
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|
3692 } |
|
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diff
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|
3693 } |
|
279d0427ef26
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diff
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|
3694 } |
|
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diff
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|
3695 return retval; |
|
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diff
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|
3696 } |
|
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|
3697 |
|
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diff
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|
3698 static ALboolean Internal_SetVolumeSource(ALuint source, ALfloat volume) |
|
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|
3699 { |
|
279d0427ef26
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diff
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|
3700 ALint channel; |
|
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|
3701 if(0 == source) |
|
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diff
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|
3702 { |
|
279d0427ef26
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diff
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|
3703 return Internal_SetVolumeChannel(-1, volume); |
|
279d0427ef26
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diff
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|
3704 } |
|
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diff
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|
3705 |
|
279d0427ef26
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diff
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|
3706 channel = Internal_GetChannel(source); |
|
279d0427ef26
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|
3707 if(-1 == channel) |
|
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|
3708 { |
|
279d0427ef26
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|
3709 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); |
|
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|
3710 return AL_FALSE; |
|
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3711 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3712 return Internal_SetVolumeChannel(channel, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3713 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3714 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3715 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3716 static ALfloat Internal_GetVolumeChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3717 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3718 ALfloat value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3719 ALenum error; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3720 ALfloat running_total = 0.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3721 ALfloat retval = 0.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3722 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3723 if(channel >= Number_of_Channels_global) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3724 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3725 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3726 return -1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3727 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3728 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3729 if(channel >= 0) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3730 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3731 alGetSourcef(ALmixer_Channel_List[channel].alsource, |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3732 AL_GAIN, &value); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3733 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3734 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3735 ALmixer_SetError("%s", alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3736 retval = -1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3737 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3738 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3739 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3740 retval = value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3741 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3742 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3743 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3744 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3745 ALint i; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3746 for(i=0; i<Number_of_Channels_global; i++) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3747 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3748 alGetSourcef(ALmixer_Channel_List[i].alsource, |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3749 AL_GAIN, &value); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3750 if((error = alGetError()) != AL_NO_ERROR) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3751 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3752 ALmixer_SetError("%s", alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3753 retval = -1; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3754 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3755 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3756 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3757 running_total += value; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3758 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3759 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3760 if(0 == Number_of_Channels_global) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3761 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3762 ALmixer_SetError("No channels are allocated"); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3763 retval = -1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3764 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3765 else |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3766 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3767 retval = running_total / Number_of_Channels_global; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3768 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3769 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3770 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3771 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3772 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3773 static ALfloat Internal_GetVolumeSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3774 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3775 ALint channel; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3776 if(0 == source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3777 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3778 return Internal_GetVolumeChannel(-1); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3779 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3780 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3781 channel = Internal_GetChannel(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3782 if(-1 == channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3783 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3784 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3785 return -1.0f; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3786 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3787 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3788 return Internal_GetVolumeChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3789 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3790 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3791 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3792 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3793 /* Set a volume regardless if it's in use or not. |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3794 */ |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3795 static ALboolean Internal_SetMaxVolumeChannel(ALint channel, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3796 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3797 ALenum error; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3798 ALboolean retval = AL_TRUE; |
| 0 | 3799 |
| 3800 if(channel >= Number_of_Channels_global) | |
| 3801 { | |
| 3802 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3803 return AL_FALSE; |
| 0 | 3804 } |
| 3805 | |
| 3806 if(channel >= 0) | |
| 3807 { | |
| 3808 if(volume < 0.0f) | |
| 3809 { | |
| 3810 volume = 0.0f; | |
| 3811 } | |
| 3812 else if(volume > 1.0f) | |
| 3813 { | |
| 3814 volume = 1.0f; | |
| 3815 } | |
| 3816 ALmixer_Channel_List[channel].max_volume = volume; | |
| 3817 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3818 AL_MAX_GAIN, volume); | |
| 3819 if((error = alGetError()) != AL_NO_ERROR) | |
| 3820 { | |
| 3821 ALmixer_SetError("%s", | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3822 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3823 retval = AL_FALSE; |
| 0 | 3824 } |
| 3825 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
| 3826 { | |
| 3827 ALmixer_Channel_List[channel].min_volume = volume; | |
| 3828 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3829 AL_MIN_GAIN, volume); | |
| 3830 if((error = alGetError()) != AL_NO_ERROR) | |
| 3831 { | |
| 3832 ALmixer_SetError("%s", | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3833 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3834 retval = AL_FALSE; |
| 0 | 3835 } |
| 3836 } | |
| 3837 } | |
| 3838 else | |
| 3839 { | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3840 ALint i; |
| 0 | 3841 for(i=0; i<Number_of_Channels_global; i++) |
| 3842 { | |
| 3843 if(volume < 0.0f) | |
| 3844 { | |
| 3845 volume = 0.0f; | |
| 3846 } | |
| 3847 else if(volume > 1.0f) | |
| 3848 { | |
| 3849 volume = 1.0f; | |
| 3850 } | |
| 3851 ALmixer_Channel_List[i].max_volume = volume; | |
| 3852 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 3853 AL_MAX_GAIN, volume); | |
| 3854 if((error = alGetError()) != AL_NO_ERROR) | |
| 3855 { | |
| 3856 ALmixer_SetError("%s", | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3857 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3858 retval = AL_FALSE; |
| 0 | 3859 } |
| 3860 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
| 3861 { | |
| 3862 ALmixer_Channel_List[i].min_volume = volume; | |
| 3863 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 3864 AL_MIN_GAIN, volume); | |
| 3865 if((error = alGetError()) != AL_NO_ERROR) | |
| 3866 { | |
| 3867 ALmixer_SetError("%s", | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3868 alGetString(error) ); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3869 retval = AL_FALSE; |
| 0 | 3870 } |
| 3871 } | |
| 3872 } | |
| 3873 } | |
| 3874 return retval; | |
| 3875 } | |
| 3876 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3877 static ALint Internal_SetMaxVolumeSource(ALuint source, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3878 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3879 ALint channel; |
| 0 | 3880 if(0 == source) |
| 3881 { | |
| 3882 return Internal_SetMaxVolumeChannel(-1, volume); | |
| 3883 } | |
| 3884 | |
| 3885 channel = Internal_GetChannel(source); | |
| 3886 if(-1 == channel) | |
| 3887 { | |
| 3888 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3889 return AL_FALSE; |
| 0 | 3890 } |
| 3891 return Internal_SetMaxVolumeChannel(channel, volume); | |
| 3892 } | |
| 3893 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
3894 static ALfloat Internal_GetMaxVolumeChannel(ALint channel) |
| 0 | 3895 { |
| 3896 /* | |
| 3897 ALfloat value; | |
| 3898 ALenum error; | |
| 3899 */ | |
| 3900 ALfloat running_total = 0.0f; | |
| 3901 ALfloat retval = 0.0f; | |
| 3902 | |
| 3903 if(channel >= Number_of_Channels_global) | |
| 3904 { | |
| 3905 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 3906 return -1.0f; | |
| 3907 } | |
| 3908 | |
| 3909 if(channel >= 0) | |
| 3910 { | |
| 3911 /* | |
| 3912 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 3913 AL_GAIN, &value); | |
| 3914 if((error = alGetError()) != AL_NO_ERROR) | |
| 3915 { | |
| 3916 ALmixer_SetError("%s", | |
|
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3917 alGetString(error) ); |
| 0 | 3918 retval = -1.0f; |
| 3919 } | |
| 3920 else | |
| 3921 { | |
| 3922 retval = value; | |
| 3923 } | |
| 3924 */ | |
| 3925 retval = ALmixer_Channel_List[channel].max_volume; | |
| 3926 | |
| 3927 } | |
| 3928 else | |
| 3929 { | |
|
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3930 ALint i; |
| 0 | 3931 for(i=0; i<Number_of_Channels_global; i++) |
| 3932 { | |
| 3933 /* | |
| 3934 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 3935 AL_GAIN, &value); | |
| 3936 if((error = alGetError()) != AL_NO_ERROR) | |
| 3937 { | |
| 3938 ALmixer_SetError("%s", | |
|
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3939 alGetString(error) ); |
| 0 | 3940 retval = -1; |
| 3941 } | |
| 3942 else | |
| 3943 { | |
| 3944 running_total += value; | |
| 3945 } | |
| 3946 */ | |
| 3947 running_total += ALmixer_Channel_List[i].max_volume; | |
| 3948 } | |
| 3949 if(0 == Number_of_Channels_global) | |
| 3950 { | |
| 3951 ALmixer_SetError("No channels are allocated"); | |
| 3952 retval = -1.0f; | |
| 3953 } | |
| 3954 else | |
| 3955 { | |
| 3956 retval = running_total / Number_of_Channels_global; | |
| 3957 } | |
| 3958 } | |
| 3959 return retval; | |
| 3960 } | |
| 3961 | |
| 3962 static ALfloat Internal_GetMaxVolumeSource(ALuint source) | |
| 3963 { | |
|
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3964 ALint channel; |
| 0 | 3965 if(0 == source) |
| 3966 { | |
| 3967 return Internal_GetMaxVolumeChannel(-1); | |
| 3968 } | |
| 3969 | |
| 3970 channel = Internal_GetChannel(source); | |
| 3971 if(-1 == channel) | |
| 3972 { | |
| 3973 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
| 3974 return -1.0f; | |
| 3975 } | |
| 3976 | |
| 3977 return Internal_GetMaxVolumeChannel(channel); | |
| 3978 } | |
| 3979 | |
| 3980 | |
| 3981 /* Set a volume regardless if it's in use or not. | |
| 3982 */ | |
|
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3983 static ALboolean Internal_SetMinVolumeChannel(ALint channel, ALfloat volume) |
| 0 | 3984 { |
| 3985 ALenum error; | |
|
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3986 ALboolean retval = AL_TRUE; |
| 0 | 3987 |
| 3988 if(channel >= Number_of_Channels_global) | |
| 3989 { | |
| 3990 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
|
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3991 return AL_FALSE; |
| 0 | 3992 } |
| 3993 | |
| 3994 if(channel >= 0) | |
| 3995 { | |
| 3996 if(volume < 0.0f) | |
| 3997 { | |
| 3998 volume = 0.0f; | |
| 3999 } | |
| 4000 else if(volume > 1.0f) | |
| 4001 { | |
| 4002 volume = 1.0f; | |
| 4003 } | |
| 4004 ALmixer_Channel_List[channel].min_volume = volume; | |
| 4005 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4006 AL_MIN_GAIN, volume); | |
| 4007 if((error = alGetError()) != AL_NO_ERROR) | |
| 4008 { | |
| 4009 ALmixer_SetError("%s", | |
|
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4010 alGetString(error) ); |
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4011 retval = AL_FALSE; |
| 0 | 4012 } |
| 4013 if(ALmixer_Channel_List[channel].max_volume < ALmixer_Channel_List[channel].min_volume) | |
| 4014 { | |
| 4015 ALmixer_Channel_List[channel].max_volume = volume; | |
| 4016 alSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4017 AL_MAX_GAIN, volume); | |
| 4018 if((error = alGetError()) != AL_NO_ERROR) | |
| 4019 { | |
| 4020 ALmixer_SetError("%s", | |
|
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4021 alGetString(error) ); |
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4022 retval = AL_FALSE; |
| 0 | 4023 } |
| 4024 } | |
| 4025 } | |
| 4026 else | |
| 4027 { | |
|
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4028 ALint i; |
| 0 | 4029 for(i=0; i<Number_of_Channels_global; i++) |
| 4030 { | |
| 4031 if(volume < 0.0f) | |
| 4032 { | |
| 4033 volume = 0.0f; | |
| 4034 } | |
| 4035 else if(volume > 1.0f) | |
| 4036 { | |
| 4037 volume = 1.0f; | |
| 4038 } | |
| 4039 ALmixer_Channel_List[i].min_volume = volume; | |
| 4040 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 4041 AL_MIN_GAIN, volume); | |
| 4042 if((error = alGetError()) != AL_NO_ERROR) | |
| 4043 { | |
| 4044 ALmixer_SetError("%s", | |
|
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4045 alGetString(error) ); |
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4046 retval = AL_FALSE; |
| 0 | 4047 } |
| 4048 if(ALmixer_Channel_List[i].max_volume < ALmixer_Channel_List[i].min_volume) | |
| 4049 { | |
| 4050 ALmixer_Channel_List[i].max_volume = volume; | |
| 4051 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 4052 AL_MAX_GAIN, volume); | |
| 4053 if((error = alGetError()) != AL_NO_ERROR) | |
| 4054 { | |
| 4055 ALmixer_SetError("%s", | |
|
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4056 alGetString(error) ); |
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4057 retval = AL_FALSE; |
| 0 | 4058 } |
| 4059 } | |
| 4060 } | |
| 4061 } | |
| 4062 return retval; | |
| 4063 } | |
| 4064 | |
|
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4065 static ALboolean Internal_SetMinVolumeSource(ALuint source, ALfloat volume) |
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4066 { |
|
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4067 ALint channel; |
| 0 | 4068 if(0 == source) |
| 4069 { | |
| 4070 return Internal_SetMinVolumeChannel(-1, volume); | |
| 4071 } | |
| 4072 | |
| 4073 channel = Internal_GetChannel(source); | |
| 4074 if(-1 == channel) | |
| 4075 { | |
| 4076 ALmixer_SetError("Cannot SetMaxVolume: %s", ALmixer_GetError()); | |
|
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4077 return AL_FALSE; |
| 0 | 4078 } |
| 4079 return Internal_SetMinVolumeChannel(channel, volume); | |
| 4080 } | |
| 4081 | |
|
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4082 static ALfloat Internal_GetMinVolumeChannel(ALint channel) |
| 0 | 4083 { |
| 4084 /* | |
| 4085 ALfloat value; | |
| 4086 ALenum error; | |
| 4087 */ | |
| 4088 ALfloat running_total = 0.0f; | |
| 4089 ALfloat retval = 0.0f; | |
| 4090 | |
| 4091 if(channel >= Number_of_Channels_global) | |
| 4092 { | |
| 4093 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 4094 return -1.0f; | |
| 4095 } | |
| 4096 | |
| 4097 if(channel >= 0) | |
| 4098 { | |
| 4099 /* | |
| 4100 alGetSourcef(ALmixer_Channel_List[channel].alsource, | |
| 4101 AL_GAIN, &value); | |
| 4102 if((error = alGetError()) != AL_NO_ERROR) | |
| 4103 { | |
| 4104 ALmixer_SetError("%s", | |
|
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|
4105 alGetString(error) ); |
| 0 | 4106 retval = -1.0f; |
| 4107 } | |
| 4108 else | |
| 4109 { | |
| 4110 retval = value; | |
| 4111 } | |
| 4112 */ | |
| 4113 retval = ALmixer_Channel_List[channel].min_volume; | |
| 4114 | |
| 4115 } | |
| 4116 else | |
| 4117 { | |
|
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|
4118 ALint i; |
| 0 | 4119 for(i=0; i<Number_of_Channels_global; i++) |
| 4120 { | |
| 4121 /* | |
| 4122 alGetSourcef(ALmixer_Channel_List[i].alsource, | |
| 4123 AL_GAIN, &value); | |
| 4124 if((error = alGetError()) != AL_NO_ERROR) | |
| 4125 { | |
| 4126 ALmixer_SetError("%s", | |
|
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|
4127 alGetString(error) ); |
| 0 | 4128 retval = -1; |
| 4129 } | |
| 4130 else | |
| 4131 { | |
| 4132 running_total += value; | |
| 4133 } | |
| 4134 */ | |
| 4135 running_total += ALmixer_Channel_List[i].min_volume; | |
| 4136 } | |
| 4137 if(0 == Number_of_Channels_global) | |
| 4138 { | |
| 4139 ALmixer_SetError("No channels are allocated"); | |
| 4140 retval = -1.0f; | |
| 4141 } | |
| 4142 else | |
| 4143 { | |
| 4144 retval = running_total / Number_of_Channels_global; | |
| 4145 } | |
| 4146 } | |
| 4147 return retval; | |
| 4148 } | |
| 4149 | |
| 4150 static ALfloat Internal_GetMinVolumeSource(ALuint source) | |
| 4151 { | |
|
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|
4152 ALint channel; |
| 0 | 4153 if(0 == source) |
| 4154 { | |
| 4155 return Internal_GetMinVolumeChannel(-1); | |
| 4156 } | |
| 4157 | |
| 4158 channel = Internal_GetChannel(source); | |
| 4159 if(-1 == channel) | |
| 4160 { | |
| 4161 ALmixer_SetError("Cannot GetVolume: %s", ALmixer_GetError()); | |
| 4162 return -1.0f; | |
| 4163 } | |
| 4164 | |
| 4165 return Internal_GetMinVolumeChannel(channel); | |
| 4166 } | |
| 4167 | |
| 4168 | |
| 4169 /* Changes the listener volume */ | |
|
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|
4170 static ALboolean Internal_SetMasterVolume(ALfloat volume) |
| 0 | 4171 { |
| 4172 ALenum error; | |
| 4173 alListenerf(AL_GAIN, volume); | |
| 4174 if((error = alGetError()) != AL_NO_ERROR) | |
| 4175 { | |
| 4176 ALmixer_SetError("%s", | |
|
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|
4177 alGetString(error) ); |
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4178 return AL_FALSE; |
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4179 } |
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|
4180 return AL_TRUE; |
| 0 | 4181 } |
| 4182 | |
| 4183 static ALfloat Internal_GetMasterVolume() | |
| 4184 { | |
| 4185 ALenum error; | |
| 4186 ALfloat volume; | |
| 4187 alGetListenerf(AL_GAIN, &volume); | |
| 4188 if((error = alGetError()) != AL_NO_ERROR) | |
| 4189 { | |
| 4190 ALmixer_SetError("%s", | |
|
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|
4191 alGetString(error) ); |
| 0 | 4192 return -1.0f; |
| 4193 } | |
| 4194 return volume; | |
| 4195 } | |
| 4196 | |
| 4197 | |
| 4198 | |
| 4199 | |
| 4200 /* Will fade out currently playing channels. | |
| 4201 * It starts at the current volume level and goes down */ | |
|
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|
4202 static ALint Internal_ExpireChannel(ALint channel, ALint ticks) |
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|
4203 { |
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|
4204 ALuint current_time = ALmixer_GetTicks(); |
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|
4205 ALuint counter = 0; |
| 0 | 4206 |
| 4207 /* We can't accept 0 as a value because of div-by-zero. | |
| 4208 * If zero, just call Halt at normal | |
| 4209 * volume | |
| 4210 */ | |
| 4211 if(0 == ticks) | |
| 4212 { | |
|
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|
4213 return Internal_HaltChannel(channel, AL_FALSE); |
| 0 | 4214 } |
| 4215 if(ticks < -1) | |
| 4216 { | |
| 4217 ticks = -1; | |
| 4218 } | |
| 4219 | |
| 4220 | |
| 4221 if(channel >= Number_of_Channels_global) | |
| 4222 { | |
| 4223 ALmixer_SetError("Requested channel (%d) exceeds maximum channel (%d) because only %d channels are allocated", channel, Number_of_Channels_global-1, Number_of_Channels_global); | |
| 4224 return -1; | |
| 4225 } | |
| 4226 | |
| 4227 if(channel >= 0) | |
| 4228 { | |
| 4229 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4230 { | |
| 4231 /* Set expire start time */ | |
| 4232 ALmixer_Channel_List[channel].start_time = current_time; | |
| 4233 /* Set the expire ticks */ | |
| 4234 ALmixer_Channel_List[channel].expire_ticks = ticks; | |
| 4235 | |
| 4236 counter++; | |
| 4237 } | |
| 4238 } | |
| 4239 /* Else need to fade out all channels */ | |
| 4240 else | |
| 4241 { | |
|
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4242 ALint i; |
| 0 | 4243 for(i=0; i<Number_of_Channels_global; i++) |
| 4244 { | |
| 4245 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4246 { | |
| 4247 /* Set expire start time */ | |
| 4248 ALmixer_Channel_List[i].start_time = current_time; | |
| 4249 /* Set the expire ticks */ | |
| 4250 ALmixer_Channel_List[i].expire_ticks = ticks; | |
| 4251 | |
| 4252 counter++; | |
| 4253 } | |
| 4254 } /* End for loop */ | |
| 4255 } | |
| 4256 return counter; | |
| 4257 } | |
| 4258 | |
| 4259 | |
|
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|
4260 static ALint Internal_ExpireSource(ALuint source, ALint ticks) |
|
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|
4261 { |
|
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|
4262 ALint channel; |
| 0 | 4263 if(0 == source) |
| 4264 { | |
| 4265 return Internal_ExpireChannel(-1, ticks); | |
| 4266 } | |
| 4267 | |
| 4268 channel = Internal_GetChannel(source); | |
| 4269 if(-1 == channel) | |
| 4270 { | |
| 4271 ALmixer_SetError("Cannot Expire source: %s", ALmixer_GetError()); | |
| 4272 return -1; | |
| 4273 } | |
| 4274 return Internal_ExpireChannel(channel, ticks); | |
| 4275 } | |
| 4276 | |
| 4277 | |
|
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|
4278 static ALint Internal_QueryChannel(ALint channel) |
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4279 { |
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|
4280 ALint i; |
|
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|
4281 ALint counter = 0; |
| 0 | 4282 if(channel >= Number_of_Channels_global) |
| 4283 { | |
| 4284 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4285 return -1; | |
| 4286 } | |
| 4287 | |
| 4288 if(channel >= 0) | |
| 4289 { | |
| 4290 return ALmixer_Channel_List[channel].channel_in_use; | |
| 4291 } | |
| 4292 | |
| 4293 /* Else, return the number of channels in use */ | |
| 4294 for(i=0; i<Number_of_Channels_global; i++) | |
| 4295 { | |
| 4296 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4297 { | |
| 4298 counter++; | |
| 4299 } | |
| 4300 } | |
| 4301 return counter; | |
| 4302 } | |
| 4303 | |
| 4304 | |
|
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4305 static ALint Internal_QuerySource(ALuint source) |
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4306 { |
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|
4307 ALint channel; |
| 0 | 4308 if(0 == source) |
| 4309 { | |
| 4310 return Internal_QueryChannel(-1); | |
| 4311 } | |
| 4312 | |
| 4313 channel = Internal_GetChannel(source); | |
| 4314 if(-1 == channel) | |
| 4315 { | |
| 4316 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4317 return -1; | |
| 4318 } | |
| 4319 | |
| 4320 return Internal_QueryChannel(channel); | |
| 4321 } | |
| 4322 | |
| 4323 | |
|
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|
4324 static ALuint Internal_CountUnreservedUsedChannels() |
|
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|
4325 { |
|
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|
4326 ALint i; |
|
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|
4327 ALuint counter = 0; |
| 0 | 4328 |
| 4329 | |
| 4330 /* Else, return the number of channels in use */ | |
| 4331 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 4332 { | |
| 4333 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4334 { | |
| 4335 counter++; | |
| 4336 } | |
| 4337 } | |
| 4338 return counter; | |
| 4339 } | |
| 4340 | |
|
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|
4341 static ALuint Internal_CountUnreservedFreeChannels() |
|
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|
4342 { |
|
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|
4343 ALint i; |
|
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|
4344 ALuint counter = 0; |
| 0 | 4345 |
| 4346 | |
| 4347 /* Else, return the number of channels in use */ | |
| 4348 for(i=Number_of_Reserve_Channels_global; i<Number_of_Channels_global; i++) | |
| 4349 { | |
| 4350 if( ! ALmixer_Channel_List[i].channel_in_use) | |
| 4351 { | |
| 4352 counter++; | |
| 4353 } | |
| 4354 } | |
| 4355 return counter; | |
| 4356 } | |
| 4357 | |
|
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|
4358 static ALuint Internal_CountAllUsedChannels() |
|
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|
4359 { |
|
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|
4360 ALint i; |
|
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|
4361 ALuint counter = 0; |
| 0 | 4362 |
| 4363 | |
| 4364 /* Else, return the number of channels in use */ | |
| 4365 for(i=0; i<Number_of_Channels_global; i++) | |
| 4366 { | |
| 4367 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4368 { | |
| 4369 counter++; | |
| 4370 } | |
| 4371 } | |
| 4372 return counter; | |
| 4373 } | |
| 4374 | |
|
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|
4375 static ALuint Internal_CountAllFreeChannels() |
|
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|
4376 { |
|
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|
4377 ALint i; |
|
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|
4378 ALuint counter = 0; |
| 0 | 4379 |
| 4380 | |
| 4381 /* Else, return the number of channels in use */ | |
| 4382 for(i=0; i<Number_of_Channels_global; i++) | |
| 4383 { | |
| 4384 if( ! ALmixer_Channel_List[i].channel_in_use) | |
| 4385 { | |
| 4386 counter++; | |
| 4387 } | |
| 4388 } | |
| 4389 return counter; | |
| 4390 } | |
| 4391 | |
| 4392 | |
|
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|
4393 static ALint Internal_PlayingChannel(ALint channel) |
|
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|
4394 { |
|
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|
4395 ALint i; |
|
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|
4396 ALint counter = 0; |
| 0 | 4397 ALint state; |
| 4398 | |
| 4399 if(channel >= Number_of_Channels_global) | |
| 4400 { | |
| 4401 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4402 return -1; | |
| 4403 } | |
| 4404 | |
| 4405 if(channel >= 0) | |
| 4406 { | |
| 4407 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4408 { | |
| 4409 alGetSourcei( | |
| 4410 ALmixer_Channel_List[channel].alsource, | |
| 4411 AL_SOURCE_STATE, &state | |
| 4412 ); | |
| 4413 if(AL_PLAYING == state) | |
| 4414 { | |
| 4415 return 1; | |
| 4416 } | |
| 4417 } | |
| 4418 return 0; | |
| 4419 } | |
| 4420 | |
| 4421 /* Else, return the number of channels in use */ | |
| 4422 for(i=0; i<Number_of_Channels_global; i++) | |
| 4423 { | |
| 4424 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4425 { | |
| 4426 alGetSourcei( | |
| 4427 ALmixer_Channel_List[i].alsource, | |
| 4428 AL_SOURCE_STATE, &state | |
| 4429 ); | |
| 4430 if(AL_PLAYING == state) | |
| 4431 { | |
| 4432 counter++; | |
| 4433 } | |
| 4434 } | |
| 4435 } | |
| 4436 return counter; | |
| 4437 } | |
| 4438 | |
| 4439 | |
|
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|
4440 static ALint Internal_PlayingSource(ALuint source) |
|
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|
4441 { |
|
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|
4442 ALint channel; |
| 0 | 4443 if(0 == source) |
| 4444 { | |
| 4445 return Internal_PlayingChannel(-1); | |
| 4446 } | |
| 4447 | |
| 4448 channel = Internal_GetChannel(source); | |
| 4449 if(-1 == channel) | |
| 4450 { | |
| 4451 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4452 return -1; | |
| 4453 } | |
| 4454 | |
| 4455 return Internal_PlayingChannel(channel); | |
| 4456 } | |
| 4457 | |
| 4458 | |
|
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|
4459 static ALint Internal_PausedChannel(ALint channel) |
|
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|
4460 { |
|
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|
4461 ALint i; |
|
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|
4462 ALint counter = 0; |
| 0 | 4463 ALint state; |
| 4464 | |
| 4465 if(channel >= Number_of_Channels_global) | |
| 4466 { | |
| 4467 ALmixer_SetError("Invalid channel: %d", channel); | |
| 4468 return -1; | |
| 4469 } | |
| 4470 | |
| 4471 if(channel >= 0) | |
| 4472 { | |
| 4473 if(ALmixer_Channel_List[channel].channel_in_use) | |
| 4474 { | |
| 4475 alGetSourcei( | |
| 4476 ALmixer_Channel_List[channel].alsource, | |
| 4477 AL_SOURCE_STATE, &state | |
| 4478 ); | |
| 4479 if(AL_PAUSED == state) | |
| 4480 { | |
| 4481 return 1; | |
| 4482 } | |
| 4483 } | |
| 4484 return 0; | |
| 4485 } | |
| 4486 | |
| 4487 /* Else, return the number of channels in use */ | |
| 4488 for(i=0; i<Number_of_Channels_global; i++) | |
| 4489 { | |
| 4490 if(ALmixer_Channel_List[i].channel_in_use) | |
| 4491 { | |
| 4492 alGetSourcei( | |
| 4493 ALmixer_Channel_List[i].alsource, | |
| 4494 AL_SOURCE_STATE, &state | |
| 4495 ); | |
| 4496 if(AL_PAUSED == state) | |
| 4497 { | |
| 4498 counter++; | |
| 4499 } | |
| 4500 } | |
| 4501 } | |
| 4502 return counter; | |
| 4503 } | |
| 4504 | |
| 4505 | |
|
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|
4506 static ALint Internal_PausedSource(ALuint source) |
|
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|
4507 { |
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|
4508 ALint channel; |
| 0 | 4509 if(0 == source) |
| 4510 { | |
| 4511 return Internal_PausedChannel(-1); | |
| 4512 } | |
| 4513 | |
| 4514 channel = Internal_GetChannel(source); | |
| 4515 if(-1 == channel) | |
| 4516 { | |
| 4517 ALmixer_SetError("Cannot query source: %s", ALmixer_GetError()); | |
| 4518 return -1; | |
| 4519 } | |
| 4520 | |
| 4521 return Internal_PausedChannel(channel); | |
| 4522 } | |
| 4523 | |
| 4524 | |
| 4525 | |
| 4526 | |
| 4527 | |
| 4528 | |
| 4529 /* Private function for Updating ALmixer. | |
| 4530 * This is a very big and ugly function. | |
| 4531 * It should return the number of buffers that were | |
| 4532 * queued during the call. The value might be | |
| 4533 * used to guage how long you might wait to | |
| 4534 * call the next update loop in case you are worried | |
| 4535 * about preserving CPU cycles. The idea is that | |
| 4536 * when a buffer is queued, there was probably some | |
| 4537 * CPU intensive looping which took awhile. | |
| 4538 * It's mainly provided as a convenience. | |
|
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|
4539 * Timing the call with ALmixer_GetTicks() would produce |
| 0 | 4540 * more accurate information. |
| 4541 * Returns a negative value if there was an error, | |
| 4542 * the value being the number of errors. | |
| 4543 */ | |
|
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|
4544 static ALint Update_ALmixer(void* data) |
|
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|
4545 { |
|
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|
4546 ALint retval = 0; |
|
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|
4547 ALint error_flag = 0; |
| 0 | 4548 ALenum error; |
| 4549 ALint state; | |
|
2
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|
4550 ALint i=0; |
|
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|
4551 |
|
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|
4552 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
4553 SDL_LockMutex(s_simpleLock); |
|
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|
4554 #endif |
| 0 | 4555 if(0 == ALmixer_Initialized) |
| 4556 { | |
|
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|
4557 #ifdef ENABLE_ALMIXER_THREADS |
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|
4558 SDL_UnlockMutex(s_simpleLock); |
|
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|
4559 #endif |
| 0 | 4560 return 0; |
| 4561 } | |
| 4562 | |
| 4563 /* Check the quick flag to see if anything needs updating */ | |
| 4564 /* If anything is playing, then we have to do work */ | |
| 4565 if( 0 == Is_Playing_global) | |
| 4566 { | |
|
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|
4567 #ifdef ENABLE_ALMIXER_THREADS |
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|
4568 SDL_UnlockMutex(s_simpleLock); |
|
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|
4569 #endif |
| 0 | 4570 return 0; |
| 4571 } | |
| 4572 /* Clear error */ | |
| 4573 if((error = alGetError()) != AL_NO_ERROR) | |
| 4574 { | |
| 4575 fprintf(stderr, "08Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4576 alGetString(error)); |
| 0 | 4577 } |
| 4578 alGetError(); | |
| 4579 | |
| 4580 for(i=0; i<Number_of_Channels_global; i++) | |
| 4581 { | |
| 4582 if( ALmixer_Channel_List[i].channel_in_use ) | |
| 4583 { | |
| 4584 | |
| 4585 /* For simplicity, before we do anything else, | |
| 4586 * we can check the timeout and fading values | |
| 4587 * and do the appropriate things | |
| 4588 */ | |
|
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|
4589 ALuint current_time = ALmixer_GetTicks(); |
| 0 | 4590 |
| 4591 /* Check to see if we need to halt due to Timed play */ | |
| 4592 if(ALmixer_Channel_List[i].expire_ticks != -1) | |
| 4593 { | |
|
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|
4594 ALuint target_time = (ALuint)ALmixer_Channel_List[i].expire_ticks |
| 0 | 4595 + ALmixer_Channel_List[i].start_time; |
| 4596 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
| 4597 AL_SOURCE_STATE, &state); | |
| 4598 if((error = alGetError()) != AL_NO_ERROR) | |
| 4599 { | |
| 4600 fprintf(stderr, "06Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4601 alGetString(error)); |
| 0 | 4602 } |
| 4603 | |
| 4604 /* Check the time, and also make sure that it is not | |
| 4605 * paused (if paused, we don't want to make the | |
| 4606 * evaluation because when resumed, we will adjust | |
| 4607 * the times to compensate for the pause). | |
| 4608 */ | |
| 4609 if( (current_time >= target_time) | |
| 4610 && (state != AL_PAUSED) ) | |
| 4611 { | |
| 4612 /* Stop the playback */ | |
|
10
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|
4613 Internal_HaltChannel(i, AL_FALSE); |
| 0 | 4614 if((error = alGetError()) != AL_NO_ERROR) |
| 4615 { | |
| 4616 fprintf(stderr, "07Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4617 alGetString(error)); |
| 0 | 4618 } |
| 4619 | |
| 4620 /* Everything should be done so go on to the next loop */ | |
| 4621 continue; | |
| 4622 } | |
| 4623 } /* End if time expired check */ | |
| 4624 | |
| 4625 /* Check to see if we need to adjust the volume for fading */ | |
| 4626 if( ALmixer_Channel_List[i].fade_enabled ) | |
| 4627 { | |
|
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|
4628 ALuint target_time = ALmixer_Channel_List[i].fade_expire_ticks |
| 0 | 4629 + ALmixer_Channel_List[i].fade_start_time; |
| 4630 alGetSourcei(ALmixer_Channel_List[i].alsource, | |
| 4631 AL_SOURCE_STATE, &state); | |
| 4632 if((error = alGetError()) != AL_NO_ERROR) | |
| 4633 { | |
| 4634 fprintf(stderr, "05Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4635 alGetString(error)); |
| 0 | 4636 } |
| 4637 | |
| 4638 /* Check the time, and also make sure that it is not | |
| 4639 * paused (if paused, we don't want to make the | |
| 4640 * evaluation because when resumed, we will adjust | |
| 4641 * the times to compensate for the pause). | |
| 4642 */ | |
| 4643 if(state != AL_PAUSED) | |
| 4644 { | |
| 4645 ALfloat t; | |
|
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|
4646 ALuint delta_time; |
| 0 | 4647 ALfloat current_volume; |
| 4648 if(current_time >= target_time) | |
| 4649 { | |
| 4650 /* Need to constrain value to the end time | |
| 4651 * (can't go pass the value for calculations) | |
| 4652 */ | |
| 4653 current_time = target_time; | |
| 4654 /* We can disable the fade flag now */ | |
| 4655 ALmixer_Channel_List[i].fade_enabled = 0; | |
| 4656 } | |
| 4657 /* Use the linear interpolation formula: | |
| 4658 * X = (1-t)x0 + tx1 | |
| 4659 * where x0 would be the start value | |
| 4660 * and x1 is the final value | |
| 4661 * and t is delta_time*inv_time (adjusts 0 <= time <= 1) | |
| 4662 * delta_time = current_time-start_time | |
| 4663 * inv_time = 1/ (end_time-start_time) | |
| 4664 * so t = current_time-start_time / (end_time-start_time) | |
| 4665 * | |
| 4666 */ | |
| 4667 delta_time = current_time - ALmixer_Channel_List[i].fade_start_time; | |
| 4668 t = (ALfloat) delta_time * ALmixer_Channel_List[i].fade_inv_time; | |
| 4669 | |
| 4670 current_volume = (1.0f-t) * ALmixer_Channel_List[i].fade_start_volume | |
| 4671 + t * ALmixer_Channel_List[i].fade_end_volume; | |
| 4672 | |
| 4673 /* Set the volume */ | |
| 4674 alSourcef(ALmixer_Channel_List[i].alsource, | |
| 4675 AL_MAX_GAIN, current_volume); | |
| 4676 if((error = alGetError()) != AL_NO_ERROR) | |
| 4677 { | |
| 4678 fprintf(stderr, "04Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4679 alGetString(error)); |
| 0 | 4680 } |
| 4681 | |
| 4682 /* | |
| 4683 fprintf(stderr, "Current time =%d\n", current_time); | |
| 4684 fprintf(stderr, "Current vol=%f on channel %d\n", current_volume, i); | |
| 4685 */ | |
| 4686 } /* End if not PAUSED */ | |
| 4687 } /* End if fade_enabled */ | |
| 4688 | |
| 4689 | |
| 4690 /* Okay, now that the time expired and fading stuff | |
| 4691 * is done, do the rest of the hard stuff | |
| 4692 */ | |
| 4693 | |
| 4694 | |
| 4695 /* For predecoded, check to see if done */ | |
| 4696 if( ALmixer_Channel_List[i].almixer_data->decoded_all ) | |
| 4697 { | |
| 4698 | |
| 4699 #if 0 | |
| 4700 /********* Remove this **********/ | |
| 4701 ALint buffers_processed; | |
| 4702 ALint buffers_still_queued; | |
| 4703 fprintf(stderr, "For Predecoded\n"); | |
| 4704 | |
| 4705 alGetSourcei( | |
| 4706 ALmixer_Channel_List[i].alsource, | |
| 4707 AL_SOURCE_STATE, &state | |
| 4708 ); | |
| 4709 switch(state) { | |
| 4710 case AL_PLAYING: | |
| 4711 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
| 4712 break; | |
| 4713 case AL_PAUSED: | |
| 4714 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
| 4715 break; | |
| 4716 case AL_STOPPED: | |
| 4717 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
| 4718 break; | |
| 4719 case AL_INITIAL: | |
| 4720 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
| 4721 break; | |
| 4722 default: | |
| 4723 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
| 4724 break; | |
| 4725 } | |
| 4726 | |
| 4727 alGetSourcei( | |
| 4728 ALmixer_Channel_List[i].alsource, | |
| 4729 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 4730 ); | |
| 4731 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
| 4732 | |
| 4733 alGetSourcei( | |
| 4734 ALmixer_Channel_List[i].alsource, | |
| 4735 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4736 ); | |
| 4737 | |
| 4738 /******** END REMOVE *******/ | |
| 4739 #endif | |
| 4740 /* FIXME: Ugh! Somewhere an alError is being thrown ("Invalid Enum Value"), but I can't | |
| 4741 * find it. It only seems to be thrown for OS X. I placed error messages after every al* | |
| 4742 * command I could find in the above loops, but the error doesn't seem to show | |
| 4743 * up until around here. I mistook it for a get queued buffers | |
| 4744 * error in OS X. I don't think there's an error down there. | |
| 4745 * For now, I'm clearing the error here. | |
| 4746 */ | |
| 4747 | |
| 4748 if((error = alGetError()) != AL_NO_ERROR) | |
| 4749 { | |
| 4750 fprintf(stderr, "03Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4751 alGetString(error)); |
| 0 | 4752 } |
| 4753 | |
| 4754 | |
| 4755 alGetSourcei( | |
| 4756 ALmixer_Channel_List[i].alsource, | |
| 4757 AL_SOURCE_STATE, &state | |
| 4758 ); | |
| 4759 if((error = alGetError()) != AL_NO_ERROR) | |
| 4760 { | |
| 4761 fprintf(stderr, "02Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4762 alGetString(error)); |
| 0 | 4763 } |
| 4764 | |
| 4765 | |
| 4766 if(AL_STOPPED == state) | |
| 4767 { | |
| 4768 /* Playback has ended. | |
| 4769 * Loop if necessary, or launch callback | |
| 4770 * and clear channel (or clear channel and | |
| 4771 * then launch callback?) | |
| 4772 */ | |
| 4773 | |
| 4774 | |
| 4775 /* Need to check for loops */ | |
| 4776 if(ALmixer_Channel_List[i].loops != 0) | |
| 4777 { | |
| 4778 /* Corner Case: If the buffer has | |
| 4779 * been modified using Seek, | |
| 4780 * the loop will start at the seek | |
| 4781 * position. | |
| 4782 */ | |
| 4783 if(ALmixer_Channel_List[i].loops != -1) | |
| 4784 { | |
| 4785 ALmixer_Channel_List[i].loops--; | |
| 4786 } | |
| 4787 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 4788 if((error = alGetError()) != AL_NO_ERROR) | |
| 4789 { | |
| 4790 fprintf(stderr, "50Testing error: %s\n", | |
|
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|
4791 alGetString(error)); |
| 0 | 4792 } |
| 4793 continue; | |
| 4794 } | |
| 4795 /* No loops. End play. */ | |
| 4796 else | |
| 4797 { | |
| 4798 /* Problem: It seems that when mixing | |
| 4799 * streamed and predecoded sources, | |
| 4800 * the previous instance lingers, | |
| 4801 * so we need to force remove | |
| 4802 * the data from the source. | |
| 4803 * The sharing problem | |
| 4804 * occurs when a previous predecoded buffer is played on | |
| 4805 * a source, and then a streamed source is played later | |
| 4806 * on that same source. OpenAL isn't consistently | |
| 4807 * removing the previous buffer so both get played. | |
| 4808 * (Different dists seem to have different quirks. | |
| 4809 * The problem might lead to crashes in the worst case.) | |
| 4810 */ | |
| 4811 /* Additional problem: There is another | |
| 4812 * inconsistency among OpenAL distributions. | |
| 4813 * Both Loki and Creative Windows seem to keep | |
| 4814 * the buffer queued which requires removing. | |
| 4815 * But the Creative Macintosh version does | |
| 4816 * not have any buffer queued after play | |
| 4817 * and it returns the error: Invalid Enum Value | |
| 4818 * if I try to unqueue it. | |
| 4819 * So I'm going to put in a check to see if I | |
| 4820 * can detect any buffers queued first | |
| 4821 * and then unqueue them if I can see them. | |
| 4822 * Additional note: The new CoreAudio based | |
| 4823 * implementation leaves it's buffer queued | |
| 4824 * like Loki and Creative Windows. But | |
| 4825 * considering all the problems I'm having | |
| 4826 * with the different distributions, this | |
| 4827 * check seems reasonable. | |
| 4828 */ | |
| 4829 ALint buffers_still_queued; | |
| 4830 if((error = alGetError()) != AL_NO_ERROR) | |
| 4831 { | |
| 4832 fprintf(stderr, "01Testing errpr before unqueue because getting stuff, for OS X this is expected: %s\n", | |
|
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|
4833 alGetString(error)); |
| 0 | 4834 } |
| 4835 | |
| 4836 alGetSourcei( | |
| 4837 ALmixer_Channel_List[i].alsource, | |
| 4838 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4839 ); | |
| 4840 if((error = alGetError()) != AL_NO_ERROR) | |
| 4841 { | |
| 4842 fprintf(stderr, "Error with unqueue, for OS X this is expected: %s\n", | |
|
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|
4843 alGetString(error)); |
| 0 | 4844 ALmixer_SetError("Failed detecting unqueued predecoded buffer (expected with OS X): %s", |
|
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|
4845 alGetString(error) ); |
| 0 | 4846 error_flag--; |
| 4847 } | |
| 4848 if(buffers_still_queued > 0) | |
| 4849 { | |
| 1 | 4850 |
| 4851 #if 0 /* This triggers an error in OS X Core Audio. */ | |
| 4852 alSourceUnqueueBuffers( | |
| 4853 ALmixer_Channel_List[i].alsource, | |
| 4854 1, | |
| 4855 ALmixer_Channel_List[i].almixer_data->buffer | |
| 4856 ); | |
| 4857 #else | |
| 4858 /* fprintf(stderr, "In the Bob Aron section...about to clear source\n"); | |
| 0 | 4859 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 1 | 4860 */ |
| 0 | 4861 /* Rather than force unqueuing the buffer, let's see if |
| 4862 * setting the buffer to none works (the OpenAL 1.0 | |
| 4863 * Reference Annotation suggests this should work). | |
| 4864 */ | |
| 4865 alSourcei(ALmixer_Channel_List[i].alsource, | |
| 4866 AL_BUFFER, AL_NONE); | |
| 4867 /* | |
| 4868 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 4869 */ | |
| 1 | 4870 #endif |
| 0 | 4871 if((error = alGetError()) != AL_NO_ERROR) |
| 4872 { | |
| 4873 fprintf(stderr, "Error with unqueue, after alSourceUnqueueBuffers, buffers_still_queued=%d, error is: %s", buffers_still_queued, | |
|
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|
4874 alGetString(error)); |
| 0 | 4875 ALmixer_SetError("Predecoded Unqueue buffer failed: %s", |
|
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|
4876 alGetString(error) ); |
| 0 | 4877 error_flag--; |
| 4878 } | |
| 4879 | |
| 4880 } | |
| 4881 | |
|
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|
4882 /* Launch callback */ |
|
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|
4883 Invoke_Channel_Done_Callback(i, AL_TRUE); |
|
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|
4884 |
| 0 | 4885 Clean_Channel(i); |
| 4886 /* Subtract counter */ | |
| 4887 Is_Playing_global--; | |
| 4888 | |
| 4889 | |
| 4890 /* We're done for this loop. | |
| 4891 * Go to next channel | |
| 4892 */ | |
| 4893 continue; | |
| 4894 } | |
| 4895 continue; | |
| 4896 } | |
| 4897 } /* End if decoded_all */ | |
| 4898 /* For streamed */ | |
| 4899 else | |
| 4900 { | |
| 4901 ALint buffers_processed; | |
| 4902 ALint buffers_still_queued; | |
| 4903 ALint current_buffer_id; | |
| 4904 | |
| 4905 ALuint unqueued_buffer_id; | |
| 4906 #if 0 | |
| 4907 /********* Remove this **********/ | |
| 4908 fprintf(stderr, "For Streamed\n"); | |
| 4909 | |
| 4910 alGetSourcei( | |
| 4911 ALmixer_Channel_List[i].alsource, | |
| 4912 AL_SOURCE_STATE, &state | |
| 4913 ); | |
| 4914 switch(state) { | |
| 4915 case AL_PLAYING: | |
| 4916 fprintf(stderr, "Channel '%d' is PLAYING\n", i); | |
| 4917 break; | |
| 4918 case AL_PAUSED: | |
| 4919 fprintf(stderr, "Channel '%d' is PAUSED\n",i); | |
| 4920 break; | |
| 4921 case AL_STOPPED: | |
| 4922 fprintf(stderr, "Channel '%d' is STOPPED\n",i); | |
| 4923 break; | |
| 4924 case AL_INITIAL: | |
| 4925 fprintf(stderr, "Channel '%d' is INITIAL\n",i); | |
| 4926 break; | |
| 4927 default: | |
| 4928 fprintf(stderr, "Channel '%d' is UNKNOWN\n",i); | |
| 4929 break; | |
| 4930 } | |
| 4931 /******** END REMOVE *******/ | |
| 4932 #endif | |
| 4933 /* Get the number of buffers still queued */ | |
| 4934 alGetSourcei( | |
| 4935 ALmixer_Channel_List[i].alsource, | |
| 4936 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 4937 ); | |
| 4938 if((error = alGetError()) != AL_NO_ERROR) | |
| 4939 { | |
| 4940 fprintf(stderr, "51Testing error: %s\n", | |
|
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|
4941 alGetString(error)); |
| 0 | 4942 } |
| 4943 /* Get the number of buffers processed | |
| 4944 * so we know if we need to refill | |
| 4945 */ | |
|
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|
4946 /* WARNING: It looks like Snow Leopard some times crashes on this call under x86_64 |
|
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|
4947 * typically when I suffer a lot of buffer underruns. |
|
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|
4948 */ |
|
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|
4949 // fprintf(stderr, "calling AL_BUFFERS_PROCESSED on source:%d", ALmixer_Channel_List[i].alsource); |
| 0 | 4950 alGetSourcei( |
| 4951 ALmixer_Channel_List[i].alsource, | |
| 4952 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 4953 ); | |
| 4954 if((error = alGetError()) != AL_NO_ERROR) | |
| 4955 { | |
| 4956 fprintf(stderr, "52Testing error: %s\n", | |
|
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|
4957 alGetString(error)); |
| 0 | 4958 } |
|
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|
4959 // fprintf(stderr, "finished AL_BUFFERS_PROCESSED, buffers_processed=%d", buffers_processed); |
| 0 | 4960 |
| 4961 /* WTF!!! The Nvidia distribution is failing on the alGetSourcei(source, AL_BUFFER, buf_id) call. | |
| 4962 * I need this call to figure out which buffer OpenAL is currently playing. | |
| 4963 * It keeps returning an "Invalid Enum" error. | |
| 4964 * This is totally inane! It's a basic query. | |
| 4965 * By the spec, this functionality is not explicitly defined so Nvidia refuses to | |
| 4966 * fix this behavior, even though all other distributions work fine with this. | |
| 4967 * The only workaround for this is for | |
| 4968 * a significant rewrite of my code which requires me to | |
| 4969 * duplicate the OpenAL queued buffers state with my own | |
| 4970 * code and try to derive what the current playing buffer is by indirect observation of | |
| 4971 * looking at buffers_processed. But of course this has a ton of downsides since my | |
| 4972 * queries do not give me perfect timing of what OpenAL is actually doing and | |
| 4973 * the fact that some of the distributions seem to have buffer queuing problems | |
| 4974 * with their query results (CoreAudio). This also means a ton of extra code | |
| 4975 * on my side. The lack of support of a 1 line call has required me to | |
| 4976 * implement yet another entire state machine. <sigh> | |
| 4977 */ | |
| 4978 #if 0 /* This code will not work until possibly OpenAL 1.1 because of Nvidia */ | |
| 4979 /* Get the id to the current buffer playing */ | |
| 4980 alGetSourcei( | |
| 4981 ALmixer_Channel_List[i].alsource, | |
| 4982 AL_BUFFER, ¤t_buffer_id | |
| 4983 ); | |
| 4984 if((error = alGetError()) != AL_NO_ERROR) | |
| 4985 { | |
| 4986 fprintf(stderr, "53Testing error: %s\n", | |
|
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|
4987 alGetString(error)); |
| 0 | 4988 } |
| 4989 | |
| 4990 /* Before the hard stuff, check to see if the | |
| 4991 * current queued AL buffer has changed. | |
| 4992 * If it has, we should launch a data callback if | |
| 4993 * necessary | |
| 4994 */ | |
| 4995 if( ((ALuint)current_buffer_id) != | |
| 4996 ALmixer_Channel_List[i].almixer_data->current_buffer) | |
| 4997 { | |
| 4998 ALmixer_Channel_List[i].almixer_data->current_buffer | |
| 4999 = (ALuint)current_buffer_id; | |
| 5000 | |
| 5001 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
| 5002 } | |
| 5003 #else | |
| 5004 /* Only do this if "access_data" was requested (i.e. the circular_buffer!=NULL) | |
| 5005 * And if one of the two are true: | |
| 5006 * Either buffers_processed > 0 (because the current_buffer might have changed) | |
| 5007 * or if the current_buffer==0 (because we are in an initial state or recovering from | |
| 5008 * a buffer underrun) | |
| 5009 */ | |
| 5010 if((ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5011 && ( | |
| 5012 (buffers_processed > 0) || (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) | |
| 5013 ) | |
| 5014 ) | |
| 5015 { | |
|
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|
5016 ALint k; |
|
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|
5017 ALuint queue_ret_flag; |
|
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|
5018 ALubyte is_out_of_sync = 0; |
|
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|
5019 ALuint my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
| 1 | 5020 /* Ugh, I have to deal with signed/unsigned mismatch here. */ |
| 5021 ALint buffers_unplayed_int = buffers_still_queued - buffers_processed; | |
|
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|
5022 ALuint unplayed_buffers; |
| 1 | 5023 if(buffers_unplayed_int < 0) |
| 5024 { | |
| 5025 unplayed_buffers = 0; | |
| 5026 } | |
| 5027 else | |
| 5028 { | |
|
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|
5029 unplayed_buffers = (ALuint)buffers_unplayed_int; |
| 1 | 5030 } |
| 0 | 5031 /* |
| 5032 fprintf(stderr, "Queue in processed check, before pop, buffers_processed=%d\n", buffers_processed); | |
| 5033 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5034 */ | |
| 5035 /* We can't make any determinations solely based on the number of buffers_processed | |
| 5036 * because currently, we only unqueue 1 buffer per loop. That means if 2 or more | |
| 5037 * buffers became processed in one loop, the following loop, we would have | |
| 5038 * at least that_many-1 buffers_processed (plus possible new processed). | |
| 5039 * If we tried to just remove 1 buffer from our queue, we would be incorrect | |
| 5040 * because we would not actually reflect the current playing buffer. | |
| 5041 * So the solution seems to be to make sure our queue is the same size | |
| 5042 * as the number of buffers_queued-buffers_processed, and return the head of our queue | |
| 5043 * as the current playing buffer. | |
| 5044 */ | |
| 5045 /* Also, we have a corner case. When we first start playing or if we have | |
| 5046 * a buffer underrun, we have not done a data callback. | |
| 5047 * In this case, we need to see if there is any new data in our queue | |
| 5048 * and if so, launch that data callback. | |
| 5049 */ | |
| 5050 /* Warning, this code risks the possibility of no data callback being fired if | |
| 5051 * the system is really late (or skipped buffers). | |
| 5052 */ | |
| 5053 | |
| 5054 /* First, let's syncronize our queue with the OpenAL queue */ | |
| 5055 #if 0 | |
| 5056 fprintf(stderr, "inside, Buffers processed=%d, Buffers queued=%d, my queue=%d\n", | |
| 5057 buffers_processed, buffers_still_queued, my_queue_size); | |
| 1 | 5058 #endif |
|
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|
5059 is_out_of_sync = 1; |
|
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|
5060 for(k=0; k<buffers_processed; k++) |
| 0 | 5061 { |
|
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|
5062 queue_ret_flag = CircularQueueUnsignedInt_PopFront( |
|
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|
5063 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
|
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|
5064 if(0 == queue_ret_flag) |
| 0 | 5065 { |
|
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|
5066 fprintf(stderr, "53 Error popping queue\n"); |
|
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|
5067 } |
|
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|
5068 } |
| 0 | 5069 my_queue_size = CircularQueueUnsignedInt_Size(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
| 5070 /* We have several possibilities we need to handle: | |
| 5071 * 1) We are in an initial state or underrun and need to do a data callback on the head. | |
| 5072 * 2) We were out of sync and need to do a new data callback on the new head. | |
| 5073 * 3) We were not out of sync but just had left over processed buffers which caused us to | |
| 5074 * fall in this block of code. (Don't do anything.) | |
| 5075 */ | |
| 5076 if( (0 == ALmixer_Channel_List[i].almixer_data->current_buffer) || (1 == is_out_of_sync) ) | |
| 5077 { | |
| 5078 if(my_queue_size > 0) | |
| 5079 { | |
| 5080 current_buffer_id = CircularQueueUnsignedInt_Front( | |
| 5081 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5082 if(0 == current_buffer_id) | |
| 5083 { | |
| 5084 fprintf(stderr, "53a Internal Error, current_buffer_id=0 when it shouldn't be 0\n"); | |
| 5085 } | |
| 5086 /* | |
| 5087 else | |
| 5088 { | |
| 5089 fprintf(stderr, "Queue in processed check, after pop\n"); | |
| 5090 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5091 } | |
| 5092 */ | |
| 5093 ALmixer_Channel_List[i].almixer_data->current_buffer | |
| 5094 = (ALuint)current_buffer_id; | |
| 5095 | |
| 5096 #if 0 | |
| 5097 /* Remove me...only for checking...doesn't work on Nvidia */ | |
| 5098 { | |
| 5099 ALuint real_id; | |
| 5100 alGetSourcei( | |
| 5101 ALmixer_Channel_List[i].alsource, | |
| 5102 AL_BUFFER, &real_id | |
| 5103 ); | |
| 5104 alGetError(); | |
| 5105 fprintf(stderr, "Callback fired on data buffer=%d, real_id shoud be=%d\n", current_buffer_id, real_id); | |
| 5106 } | |
| 5107 #endif | |
| 5108 Invoke_Streamed_Channel_Data_Callback(i, ALmixer_Channel_List[i].almixer_data, current_buffer_id); | |
| 5109 } | |
| 5110 else | |
| 5111 { | |
| 1 | 5112 /* |
| 0 | 5113 fprintf(stderr, "53b, Notice/Warning:, OpenAL queue has been depleted.\n"); |
| 1 | 5114 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 5115 */ | |
| 0 | 5116 /* In this case, we might either be in an underrun or finished with playback */ |
| 5117 ALmixer_Channel_List[i].almixer_data->current_buffer = 0; | |
| 5118 } | |
| 5119 } | |
| 5120 } | |
| 5121 #endif | |
| 5122 | |
| 5123 | |
| 5124 | |
| 5125 /* Just a test - remove | |
| 5126 if( ALmixer_Channel_List[i].loops > 0) | |
| 5127 { | |
| 5128 fprintf(stderr, ">>>>>>>>>>>>>>>Loops = %d\n", | |
| 5129 ALmixer_Channel_List[i].loops); | |
| 5130 } | |
| 5131 */ | |
| 5132 #if 0 | |
| 5133 fprintf(stderr, "Buffers processed = %d\n", buffers_processed); | |
| 5134 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
| 5135 #endif | |
| 5136 /* We've used up a buffer so we need to unqueue and replace */ | |
| 5137 /* Okay, it gets more complicated here: | |
| 5138 * We need to Queue more data | |
| 5139 * if buffers_processed > 0 or | |
| 5140 * if num_of_buffers_in_use < NUMBER_OF_QUEUE_BUFFERS | |
| 5141 * but we don't do this if at EOF, | |
| 5142 * except when there is looping | |
| 5143 */ | |
| 5144 /* For this to work, we must rely on EVERYTHING | |
| 5145 * else to unset the EOF if there is looping. | |
| 5146 * Remember, even Play() must do this | |
| 5147 */ | |
| 5148 | |
| 5149 /* If not EOF, then we are still playing. | |
| 5150 * Inside, we might find num_of_buffers < NUM...QUEUE_BUF.. | |
| 5151 * or buffers_process > 0 | |
| 5152 * in which case we queue up. | |
| 5153 * We also might find no buffers we need to fill, | |
| 5154 * in which case we just keep going | |
| 5155 */ | |
| 5156 if( ! ALmixer_Channel_List[i].almixer_data->eof) | |
| 5157 { | |
|
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5158 ALuint bytes_returned; |
| 0 | 5159 /* We have a priority. We first must assign |
| 5160 * unused buffers in reserve. If there is nothing | |
| 5161 * left, then we may unqueue buffers. We can't | |
| 5162 * do it the other way around because we will | |
| 5163 * lose the pointer to the unqueued buffer | |
| 5164 */ | |
| 5165 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5166 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5167 { | |
| 1 | 5168 #if 0 |
| 0 | 5169 fprintf(stderr, "Getting more data in NOT_EOF and num_buffers_in_use (%d) < max_queue (%d)\n", |
| 5170 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
| 5171 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 1 | 5172 #endif |
| 0 | 5173 /* Going to add an unused packet. |
| 5174 * Grab next packet */ | |
| 5175 bytes_returned = GetMoreData( | |
| 5176 ALmixer_Channel_List[i].almixer_data, | |
| 5177 ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5178 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5179 ); | |
| 5180 } | |
| 5181 /* For processed > 0 */ | |
| 5182 else if(buffers_processed > 0) | |
| 5183 { | |
| 5184 /* Unqueue only 1 buffer for now. | |
| 5185 * If there are more than one, | |
| 5186 * let the next Update pass deal with it | |
| 5187 * so we don't stall the program for too long. | |
| 5188 */ | |
| 5189 #if 0 | |
| 5190 fprintf(stderr, "About to Unqueue, Buffers processed = %d\n", buffers_processed); | |
| 5191 fprintf(stderr, "Buffers queued= %d\n", buffers_still_queued); | |
| 5192 fprintf(stderr, "Unqueuing a buffer\n"); | |
| 5193 #endif | |
| 5194 alSourceUnqueueBuffers( | |
| 5195 ALmixer_Channel_List[i].alsource, | |
| 5196 1, &unqueued_buffer_id | |
| 5197 ); | |
| 5198 if((error = alGetError()) != AL_NO_ERROR) | |
| 5199 { | |
| 5200 fprintf(stderr, "Error with unqueue: %s", | |
|
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5201 alGetString(error)); |
| 0 | 5202 ALmixer_SetError("Unqueue buffer failed: %s", |
|
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5203 alGetString(error) ); |
| 0 | 5204 error_flag--; |
| 5205 } | |
| 5206 /* | |
| 5207 fprintf(stderr, "Right after unqueue..."); | |
| 5208 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5209 fprintf(stderr, "Getting more data for NOT_EOF, max_buffers filled\n"); | |
| 5210 */ | |
| 5211 /* Grab unqueued packet */ | |
| 5212 bytes_returned = GetMoreData( | |
| 5213 ALmixer_Channel_List[i].almixer_data, | |
| 5214 unqueued_buffer_id); | |
| 5215 } | |
| 5216 /* We are still streaming, but currently | |
| 5217 * don't need to fill any buffers */ | |
| 5218 else | |
| 5219 { | |
| 5220 /* Might want to check state */ | |
| 5221 /* In case the playback stopped, | |
|
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5222 * we need to resume |
|
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5223 * a.k.a. buffer underrun |
|
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|
5224 */ |
| 1 | 5225 #if 1 |
| 5226 /* Try not refetching the state here because I'm getting a duplicate | |
| 5227 buffer playback (hiccup) */ | |
| 0 | 5228 alGetSourcei( |
| 5229 ALmixer_Channel_List[i].alsource, | |
| 5230 AL_SOURCE_STATE, &state | |
| 5231 ); | |
| 1 | 5232 if((error = alGetError()) != AL_NO_ERROR) |
| 5233 { | |
| 5234 fprintf(stderr, "54bTesting error: %s\n", | |
|
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|
5235 alGetString(error)); |
| 1 | 5236 } |
| 5237 /* Get the number of buffers processed | |
| 5238 */ | |
| 5239 alGetSourcei( | |
| 5240 ALmixer_Channel_List[i].alsource, | |
| 5241 AL_BUFFERS_PROCESSED, | |
| 5242 &buffers_processed | |
| 5243 ); | |
| 5244 if((error = alGetError()) != AL_NO_ERROR) | |
| 5245 { | |
| 5246 fprintf(stderr, "54cError, Can't get buffers_processed: %s\n", | |
|
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|
5247 alGetString(error)); |
| 1 | 5248 } |
| 5249 #endif | |
| 0 | 5250 if(AL_STOPPED == state) |
| 5251 { | |
| 5252 /* Resuming in not eof, but nothing to buffer */ | |
| 1 | 5253 |
| 5254 /* Okay, here's another lately discovered problem: | |
| 5255 * I can't find it in the spec, but for at least some of the | |
| 5256 * implementations, if I call play on a stopped source that | |
| 5257 * has processed buffers, all those buffers get marked as unprocessed | |
| 5258 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
| 5259 * processed, on resume, the earlier 24 buffers will get replayed, | |
| 5260 * causing a "hiccup" like sound in the playback. | |
| 5261 * To avoid this, I must unqueue all processed buffers before | |
| 5262 * calling play. But to complicate things, I need to worry about resyncing | |
| 5263 * the circular queue with this since I designed this thing | |
| 5264 * with some correlation between the two. However, I might | |
| 5265 * have already handled this, so I will try writing this code without | |
| 5266 * syncing for now. | |
| 5267 * There is currently an assumption that a buffer | |
| 5268 * was queued above so I actually have something | |
| 5269 * to play. | |
| 5270 */ | |
| 5271 ALint temp_count; | |
|
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5272 #if 0 |
|
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5273 fprintf(stderr, "STOPPED1, need to clear processed=%d, status is:\n", buffers_processed); |
| 1 | 5274 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
|
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|
5275 #endif |
| 1 | 5276 for(temp_count=0; temp_count<buffers_processed; temp_count++) |
| 5277 { | |
| 5278 alSourceUnqueueBuffers( | |
| 5279 ALmixer_Channel_List[i].alsource, | |
| 5280 1, &unqueued_buffer_id | |
| 5281 ); | |
| 5282 if((error = alGetError()) != AL_NO_ERROR) | |
| 5283 { | |
| 5284 fprintf(stderr, "55aTesting error: %s\n", | |
|
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5285 alGetString(error)); |
| 1 | 5286 error_flag--; |
| 5287 } | |
| 5288 } | |
|
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5289 #if 0 |
| 1 | 5290 fprintf(stderr, "After unqueue clear...:\n"); |
| 5291 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
|
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5292 #endif |
|
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|
5293 /* My assertion: We are STOPPED but not EOF. |
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5294 * This means we have a buffer underrun. |
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5295 * We just cleared out the unqueued buffers. |
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5296 * So we need to reset the mixer_data to reflect we have |
|
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5297 * no buffers in queue. |
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5298 * We need to GetMoreData and then queue up the data. |
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5299 * Then we need to resume playing. |
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|
5300 */ |
|
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5301 #if 0 |
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|
5302 int buffers_queued; |
|
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|
5303 alGetSourcei( |
|
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|
5304 ALmixer_Channel_List[i].alsource, |
|
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|
5305 AL_BUFFERS_QUEUED, |
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5306 &buffers_queued |
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|
5307 ); |
|
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|
5308 |
| 1 | 5309 if((error = alGetError()) != AL_NO_ERROR) |
| 5310 { | |
|
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|
5311 fprintf(stderr, "Error in PrintQueueStatus, Can't get buffers_queued: %s\n", |
|
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5312 alGetString(error)); |
|
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5313 } |
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5314 assert(buffers_queued == 0); |
|
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5315 fprintf(stderr, "buffer underrun: buffers_queued:%d\n", buffers_queued); |
|
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5316 #endif |
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5317 |
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|
5318 /* Reset the number of buffers in use to 0 */ |
|
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|
5319 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; |
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5320 |
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|
5321 /* Get more data and put it in the first buffer */ |
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5322 bytes_returned = GetMoreData( |
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5323 ALmixer_Channel_List[i].almixer_data, |
|
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5324 ALmixer_Channel_List[i].almixer_data->buffer[0] |
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5325 ); |
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|
5326 /* NOTE: We might want to look for EOF and handle it here. |
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5327 * Currently, I just let the next loop handle it which seems to be working. |
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5328 */ |
|
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5329 if(bytes_returned > 0) |
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5330 { |
|
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5331 /* Queue up the new data */ |
|
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5332 alSourceQueueBuffers( |
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5333 ALmixer_Channel_List[i].alsource, |
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5334 1, |
|
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5335 &ALmixer_Channel_List[i].almixer_data->buffer[0] |
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5336 ); |
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5337 if((error = alGetError()) != AL_NO_ERROR) |
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5338 { |
|
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|
5339 fprintf(stderr, "56e alSourceQueueBuffers error: %s\n", |
|
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5340 alGetString(error)); |
|
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|
5341 } |
|
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5342 /* Increment the number of buffers in use */ |
|
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5343 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; |
|
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5344 |
|
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5345 |
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|
5346 /* We need to empty and update the circular buffer queue if it is in use */ |
|
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|
5347 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) |
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5348 { |
|
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|
5349 ALuint queue_ret_flag; |
|
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|
5350 CircularQueueUnsignedInt_Clear(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); |
|
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|
5351 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
|
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|
5352 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, |
|
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|
5353 ALmixer_Channel_List[i].almixer_data->buffer[0] |
|
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5354 ); |
|
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|
5355 if(0 == queue_ret_flag) |
|
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|
5356 { |
|
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|
5357 fprintf(stderr, "56fSerious internal error: CircularQueue could not push into queue.\n"); |
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5358 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); |
|
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5359 } |
|
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5360 } |
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5361 |
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5362 |
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5363 |
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5364 |
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5365 /* Resume playback from underrun */ |
|
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|
5366 alSourcePlay(ALmixer_Channel_List[i].alsource); |
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|
5367 if((error = alGetError()) != AL_NO_ERROR) |
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|
5368 { |
|
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|
5369 fprintf(stderr, "55Tbesting error: %s\n", |
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|
5370 alGetString(error)); |
|
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|
5371 } |
| 1 | 5372 } |
|
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5373 |
| 0 | 5374 } |
| 5375 /* Let's escape to the next loop. | |
| 5376 * All code below this point is for queuing up | |
| 5377 */ | |
| 5378 /* | |
| 1 | 5379 fprintf(stderr, "Entry: Nothing to do...continue\n\n"); |
| 5380 */ | |
| 0 | 5381 continue; |
| 5382 } | |
| 5383 /* We now know we have to fill an available | |
| 5384 * buffer. | |
| 5385 */ | |
| 5386 | |
| 5387 /* In the previous branch, we just grabbed more data. | |
| 5388 * Let's check it to make sure it's okay, | |
| 5389 * and then queue it up | |
| 5390 */ | |
|
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5391 /* This check doesn't work anymore because it is now ALuint */ |
| 0 | 5392 #if 0 |
| 5393 if(-1 == bytes_returned) | |
| 5394 { | |
| 5395 /* Problem occurred...not sure what to do */ | |
| 5396 /* Go to next loop? */ | |
| 5397 error_flag--; | |
| 5398 /* Set the eof flag to force a quit so | |
| 5399 * we don't get stuck in an infinite loop | |
| 5400 */ | |
| 5401 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
| 5402 continue; | |
| 5403 } | |
| 5404 #endif | |
| 5405 /* This is a special case where we've run | |
| 5406 * out of data. We should check for loops | |
| 5407 * and get more data. If there is no loop, | |
| 5408 * then do nothing and wait for future | |
| 5409 * update passes to handle the EOF. | |
| 5410 * The advantage of handling the loop here | |
| 5411 * instead of waiting for play to stop is | |
| 5412 * that we should be able to keep the buffer | |
| 5413 * filled. | |
| 5414 */ | |
| 5415 #if 0 | |
| 5416 else if(0 == bytes_returned) | |
| 5417 #endif | |
| 5418 if(0 == bytes_returned) | |
| 5419 { | |
|
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5420 /* |
| 0 | 5421 fprintf(stderr, "We got 0 bytes from reading. Checking for loops\n"); |
|
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|
5422 */ |
| 0 | 5423 /* Check for loops */ |
| 5424 if( ALmixer_Channel_List[i].loops != 0 ) | |
| 5425 { | |
| 5426 /* We have to loop, so rewind | |
| 5427 * and fetch more data | |
| 5428 */ | |
|
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|
5429 /* |
| 0 | 5430 fprintf(stderr, "Rewinding data\n"); |
|
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|
5431 */ |
| 0 | 5432 if(0 == Sound_Rewind( |
| 5433 ALmixer_Channel_List[i].almixer_data->sample)) | |
| 5434 { | |
| 5435 fprintf(stderr, "Rewinding failed\n"); | |
| 5436 ALmixer_SetError( Sound_GetError() ); | |
| 5437 ALmixer_Channel_List[i].loops = 0; | |
| 5438 error_flag--; | |
| 5439 /* We'll continue on because we do have some valid data */ | |
| 5440 continue; | |
| 5441 } | |
| 5442 /* Remember to reset the data->eof flag */ | |
| 5443 ALmixer_Channel_List[i].almixer_data->eof = 0; | |
| 5444 if(ALmixer_Channel_List[i].loops > 0) | |
| 5445 { | |
| 5446 ALmixer_Channel_List[i].loops--; | |
| 5447 } | |
| 5448 /* Try grabbing another packet now. | |
| 5449 * Since we may have already unqueued a | |
| 5450 * buffer, we don't want to lose it. | |
| 5451 */ | |
| 5452 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5453 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5454 { | |
|
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|
5455 /* |
| 0 | 5456 fprintf(stderr, "We got %d bytes from reading loop. Filling unused packet\n", bytes_returned); |
|
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|
5457 */ |
| 0 | 5458 /* Grab next packet */ |
| 5459 bytes_returned = GetMoreData( | |
| 5460 ALmixer_Channel_List[i].almixer_data, | |
| 5461 ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5462 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5463 ); | |
|
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|
5464 /* |
| 0 | 5465 fprintf(stderr, "We reread %d bytes into unused packet\n", bytes_returned); |
|
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|
5466 */ |
| 0 | 5467 } |
| 5468 /* Refilling unqueued packet */ | |
| 5469 else | |
| 5470 { | |
|
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|
5471 /* |
| 0 | 5472 fprintf(stderr, "We got %d bytes from reading loop. Filling unqueued packet\n", bytes_returned); |
|
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|
5473 */ |
| 0 | 5474 /* Grab next packet */ |
| 5475 bytes_returned = GetMoreData( | |
| 5476 ALmixer_Channel_List[i].almixer_data, | |
| 5477 unqueued_buffer_id); | |
|
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|
5478 /* |
| 0 | 5479 fprintf(stderr, "We reread %d bytes into unqueued packet\n", bytes_returned); |
|
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|
5480 */ |
| 0 | 5481 } |
| 5482 /* Another error check */ | |
| 5483 /* | |
| 5484 if(bytes_returned <= 0) | |
| 5485 */ | |
| 5486 if(0 == bytes_returned) | |
| 5487 { | |
| 5488 fprintf(stderr, "??????????ERROR\n"); | |
| 5489 ALmixer_SetError("Could not loop because after rewind, no data could be retrieved"); | |
| 5490 /* Problem occurred...not sure what to do */ | |
| 5491 /* Go to next loop? */ | |
| 5492 error_flag--; | |
| 5493 /* Set the eof flag to force a quit so | |
| 5494 * we don't get stuck in an infinite loop | |
| 5495 */ | |
| 5496 ALmixer_Channel_List[i].almixer_data->eof = 1; | |
| 5497 continue; | |
| 5498 } | |
| 5499 /* We made it to the end. We still need | |
| 5500 * to BufferData, so let this branch | |
| 5501 * fall into the next piece of | |
| 5502 * code below which will handle that | |
| 5503 */ | |
| 5504 | |
| 5505 | |
| 5506 } /* END loop check */ | |
| 5507 else | |
| 5508 { | |
| 5509 /* No more loops to do. | |
| 5510 * EOF flag should be set. | |
| 5511 * Just go to next loop and | |
| 5512 * let things be handled correctly | |
| 5513 * in future update calls | |
| 5514 */ | |
| 1 | 5515 /* |
| 0 | 5516 fprintf(stderr, "SHOULD BE EOF\n"); |
| 5517 | |
| 5518 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 1 | 5519 */ |
| 0 | 5520 continue; |
| 5521 } | |
| 5522 } /* END if bytes_returned == 0 */ | |
| 5523 /********* Possible trouble point. I might be queueing empty buffers on the mac. | |
| 5524 * This check doesn't say if the buffer is valid. Only the EOF assumption is a clue at this point | |
| 5525 */ | |
| 5526 /* Fall here */ | |
| 5527 /* Everything is normal. We aren't | |
| 5528 * at an EOF, but need to simply | |
| 5529 * queue more data. The data is already checked for good, | |
| 5530 * so queue it up */ | |
| 5531 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5532 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5533 { | |
| 5534 /* Keep count of how many buffers we have | |
| 5535 * to queue so we can return the value | |
| 5536 */ | |
| 5537 retval++; | |
| 1 | 5538 /* |
| 0 | 5539 fprintf(stderr, "NOT_EOF???, about to Queue more data for num_buffers (%d) < max_queue (%d)\n", |
| 5540 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use, | |
| 5541 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 1 | 5542 */ |
| 0 | 5543 alSourceQueueBuffers( |
| 5544 ALmixer_Channel_List[i].alsource, | |
| 5545 1, | |
| 5546 &ALmixer_Channel_List[i].almixer_data->buffer[ | |
| 5547 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5548 ); | |
| 5549 if((error = alGetError()) != AL_NO_ERROR) | |
| 5550 { | |
| 5551 fprintf(stderr, "56Testing error: %s\n", | |
|
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|
5552 alGetString(error)); |
| 0 | 5553 } |
| 5554 /* This is part of the hideous Nvidia workaround. In order to figure out | |
| 5555 * which buffer to show during callbacks (for things like | |
| 5556 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
| 5557 * data structure. This code will be called only if | |
| 5558 * "access_data" was set, indicated by whether the queue is NULL. | |
| 5559 */ | |
| 5560 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5561 { | |
|
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|
5562 ALuint queue_ret_flag; |
|
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|
5563 // fprintf(stderr, "56d: CircularQueue_PushBack.\n"); |
| 0 | 5564 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
| 5565 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
| 5566 ALmixer_Channel_List[i].almixer_data->buffer[ALmixer_Channel_List[i].almixer_data->num_buffers_in_use] | |
| 5567 ); | |
| 5568 if(0 == queue_ret_flag) | |
| 5569 { | |
| 5570 fprintf(stderr, "56aSerious internal error: CircularQueue could not push into queue.\n"); | |
| 5571 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
| 5572 } | |
| 5573 /* | |
| 5574 else | |
| 5575 { | |
| 5576 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5577 } | |
| 5578 */ | |
| 5579 } | |
| 5580 } | |
| 5581 /* for processed > 0 */ | |
| 5582 else | |
| 5583 { | |
| 5584 /* Keep count of how many buffers we have | |
| 5585 * to queue so we can return the value | |
| 5586 */ | |
| 5587 retval++; | |
| 5588 /* | |
| 5589 fprintf(stderr, "NOT_EOF, about to Queue more data for filled max_queue (%d)\n", | |
| 5590 ALmixer_Channel_List[i].almixer_data->max_queue_buffers); | |
| 5591 */ | |
| 5592 alSourceQueueBuffers( | |
| 5593 ALmixer_Channel_List[i].alsource, | |
| 5594 1, &unqueued_buffer_id); | |
| 5595 if((error = alGetError()) != AL_NO_ERROR) | |
| 5596 { | |
| 5597 ALmixer_SetError("Could not QueueBuffer: %s", | |
|
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|
5598 alGetString(error) ); |
| 0 | 5599 error_flag--; |
| 5600 continue; | |
| 5601 } | |
| 5602 /* This is part of the hideous Nvidia workaround. In order to figure out | |
| 5603 * which buffer to show during callbacks (for things like | |
| 5604 * o-scopes), I must keep a copy of the buffers that are queued in my own | |
| 5605 * data structure. This code will be called only if | |
| 5606 * "access_data" was set, indicated by whether the queue is NULL. | |
| 5607 */ | |
| 5608 if(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue != NULL) | |
| 5609 { | |
|
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|
5610 ALuint queue_ret_flag; |
|
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|
5611 // fprintf(stderr, "56e: CircularQueue_PushBack.\n"); |
| 0 | 5612 queue_ret_flag = CircularQueueUnsignedInt_PushBack( |
| 5613 ALmixer_Channel_List[i].almixer_data->circular_buffer_queue, | |
| 5614 unqueued_buffer_id | |
| 5615 ); | |
| 5616 if(0 == queue_ret_flag) | |
| 5617 { | |
| 5618 fprintf(stderr, "56bSerious internal error: CircularQueue could not push into queue.\n"); | |
| 5619 ALmixer_SetError("Serious internal error: CircularQueue failed to push into queue"); | |
| 5620 } | |
|
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|
5621 #if 0 |
| 0 | 5622 else |
| 5623 { | |
| 5624 CircularQueueUnsignedInt_Print(ALmixer_Channel_List[i].almixer_data->circular_buffer_queue); | |
| 5625 } | |
|
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|
5626 #endif |
| 0 | 5627 } |
| 5628 } | |
| 5629 /* If we used an available buffer queue, | |
| 5630 * then we need to update the number of them in use | |
| 5631 */ | |
| 5632 if(ALmixer_Channel_List[i].almixer_data->num_buffers_in_use | |
| 5633 < ALmixer_Channel_List[i].almixer_data->max_queue_buffers) | |
| 5634 { | |
| 5635 /* Increment the number of buffers in use */ | |
| 5636 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use++; | |
| 5637 } | |
| 5638 /* Might want to check state */ | |
| 5639 /* In case the playback stopped, | |
| 5640 * we need to resume */ | |
| 1 | 5641 #if 1 |
| 5642 /* Try not refetching the state here because I'm getting a duplicate | |
| 5643 buffer playback (hiccup) */ | |
| 0 | 5644 alGetSourcei( |
| 5645 ALmixer_Channel_List[i].alsource, | |
| 5646 AL_SOURCE_STATE, &state | |
| 5647 ); | |
| 1 | 5648 if((error = alGetError()) != AL_NO_ERROR) |
| 5649 { | |
| 5650 fprintf(stderr, "57bTesting error: %s\n", | |
|
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|
5651 alGetString(error)); |
| 1 | 5652 } |
| 5653 /* Get the number of buffers processed | |
| 5654 */ | |
| 5655 alGetSourcei( | |
| 5656 ALmixer_Channel_List[i].alsource, | |
| 5657 AL_BUFFERS_PROCESSED, | |
| 5658 &buffers_processed | |
| 5659 ); | |
| 5660 if((error = alGetError()) != AL_NO_ERROR) | |
| 5661 { | |
| 5662 fprintf(stderr, "57cError, Can't get buffers_processed: %s\n", | |
|
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|
5663 alGetString(error)); |
| 1 | 5664 } |
| 5665 #endif | |
| 0 | 5666 if(AL_STOPPED == state) |
| 5667 { | |
| 1 | 5668 /* |
| 0 | 5669 fprintf(stderr, "Resuming in not eof\n"); |
| 1 | 5670 */ |
| 5671 /* Okay, here's another lately discovered problem: | |
| 5672 * I can't find it in the spec, but for at least some of the | |
| 5673 * implementations, if I call play on a stopped source that | |
| 5674 * has processed buffers, all those buffers get marked as unprocessed | |
| 5675 * on alSourcePlay. So if I had a queue of 25 with 24 of the buffers | |
| 5676 * processed, on resume, the earlier 24 buffers will get replayed, | |
| 5677 * causing a "hiccup" like sound in the playback. | |
| 5678 * To avoid this, I must unqueue all processed buffers before | |
| 5679 * calling play. But to complicate things, I need to worry about resyncing | |
| 5680 * the circular queue with this since I designed this thing | |
| 5681 * with some correlation between the two. However, I might | |
| 5682 * have already handled this, so I will try writing this code without | |
| 5683 * syncing for now. | |
| 5684 * There is currently an assumption that a buffer | |
| 5685 * was queued above so I actually have something | |
| 5686 * to play. | |
| 5687 */ | |
| 5688 ALint temp_count; | |
| 5689 /* | |
| 5690 fprintf(stderr, "STOPPED2, need to clear processed, status is:\n"); | |
| 5691 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5692 */ | |
| 5693 | |
| 5694 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
| 5695 { | |
| 5696 alSourceUnqueueBuffers( | |
| 5697 ALmixer_Channel_List[i].alsource, | |
| 5698 1, &unqueued_buffer_id | |
| 5699 ); | |
| 5700 if((error = alGetError()) != AL_NO_ERROR) | |
| 5701 { | |
| 5702 fprintf(stderr, "58aTesting error: %s\n", | |
|
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|
5703 alGetString(error)); |
| 1 | 5704 error_flag--; |
| 5705 } | |
| 5706 } | |
| 5707 /* | |
| 5708 fprintf(stderr, "After unqueue clear...:\n"); | |
| 5709 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5710 */ | |
| 5711 | |
| 5712 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 5713 if((error = alGetError()) != AL_NO_ERROR) | |
| 5714 { | |
| 5715 fprintf(stderr, "55Tbesting 8rror: %s\n", | |
|
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|
5716 alGetString(error)); |
| 1 | 5717 } |
| 0 | 5718 } |
| 5719 continue; | |
| 5720 } /* END if( ! eof) */ | |
| 5721 /* We have hit EOF in the SDL_Sound sample and there | |
| 5722 * are no more loops. However, there may still be | |
| 5723 * buffers in the OpenAL queue which still need to | |
| 5724 * be played out. The following body of code will | |
| 5725 * determine if play is still happening or | |
| 5726 * initiate the stop/cleanup sequenece. | |
| 5727 */ | |
| 5728 else | |
| 5729 { | |
| 5730 /* Let's continue to remove the used up | |
| 5731 * buffers as they come in. */ | |
| 5732 if(buffers_processed > 0) | |
| 5733 { | |
|
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|
5734 ALint temp_count; |
| 0 | 5735 /* Do as a for-loop because I don't want |
| 5736 * to have to create an array for the | |
| 5737 * unqueued_buffer_id's | |
| 5738 */ | |
| 5739 for(temp_count=0; temp_count<buffers_processed; temp_count++) | |
| 5740 { | |
|
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|
5741 /* |
| 0 | 5742 fprintf(stderr, "unqueuing remainder, %d\n", temp_count); |
|
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|
5743 */ |
| 0 | 5744 alSourceUnqueueBuffers( |
| 5745 ALmixer_Channel_List[i].alsource, | |
| 5746 1, &unqueued_buffer_id | |
| 5747 ); | |
| 5748 if((error = alGetError()) != AL_NO_ERROR) | |
| 5749 { | |
| 5750 fprintf(stderr, "59Testing error: %s\n", | |
|
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|
5751 alGetString(error)); |
| 0 | 5752 } |
| 5753 } | |
|
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|
5754 /* |
| 0 | 5755 fprintf(stderr, "done unqueuing remainder for this loop, %d\n", temp_count); |
|
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|
5756 */ |
|
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|
5757 |
| 0 | 5758 /* Need to update counts since we removed everything. |
| 5759 * If we don't update the counts here, we end up in the | |
| 5760 * "Shouldn't be here section, but maybe it's okay due to race conditions" | |
| 5761 */ | |
| 5762 alGetSourcei( | |
| 5763 ALmixer_Channel_List[i].alsource, | |
| 5764 AL_BUFFERS_QUEUED, &buffers_still_queued | |
| 5765 ); | |
| 5766 if((error = alGetError()) != AL_NO_ERROR) | |
| 5767 { | |
| 5768 fprintf(stderr, "5100Testing error: %s\n", | |
|
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|
5769 alGetString(error)); |
| 0 | 5770 } |
| 5771 /* Get the number of buffers processed | |
| 5772 * so we know if we need to refill | |
| 5773 */ | |
| 5774 alGetSourcei( | |
| 5775 ALmixer_Channel_List[i].alsource, | |
| 5776 AL_BUFFERS_PROCESSED, &buffers_processed | |
| 5777 ); | |
| 5778 if((error = alGetError()) != AL_NO_ERROR) | |
| 5779 { | |
| 5780 fprintf(stderr, "5200Testing error: %s\n", | |
|
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|
5781 alGetString(error)); |
| 0 | 5782 } |
| 5783 } | |
| 5784 | |
| 5785 | |
| 5786 /* Else if buffers_processed == 0 | |
| 5787 * and buffers_still_queued == 0. | |
| 5788 * then we check to see if the source | |
| 5789 * is still playing. Quit if stopped | |
| 5790 * We shouldn't need to worry about | |
| 5791 * looping because that should have | |
| 5792 * been handled above. | |
| 5793 */ | |
| 5794 if(0 == buffers_still_queued) | |
| 5795 { | |
| 5796 /* Make sure playback has stopped before | |
| 5797 * we shutdown. | |
| 5798 */ | |
| 5799 alGetSourcei( | |
| 5800 ALmixer_Channel_List[i].alsource, | |
| 5801 AL_SOURCE_STATE, &state | |
| 5802 ); | |
| 5803 if((error = alGetError()) != AL_NO_ERROR) | |
| 5804 { | |
| 5805 fprintf(stderr, "60Testing error: %s\n", | |
|
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|
5806 alGetString(error)); |
| 0 | 5807 } |
| 5808 if(AL_STOPPED == state) | |
| 5809 { | |
| 5810 ALmixer_Channel_List[i].almixer_data->num_buffers_in_use = 0; | |
| 5811 /* Playback has ended. | |
| 5812 * Loop if necessary, or launch callback | |
| 5813 * and clear channel (or clear channel and | |
| 5814 * then launch callback?) | |
|
10
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Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
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|
5815 * Update: Need to do callback first because I reference the mixer_data and source |
| 0 | 5816 */ |
|
10
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Bug fix: Moved Invoke_Callback before CleanChannel because I was trying to get the ALmixer_Data pointer, but it was cleared:
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6
diff
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|
5817 |
|
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|
5818 /* Launch callback */ |
|
c808684660a7
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diff
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|
5819 Invoke_Channel_Done_Callback(i, AL_TRUE); |
|
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diff
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|
5820 |
| 0 | 5821 Clean_Channel(i); |
| 5822 /* Subtract counter */ | |
| 5823 Is_Playing_global--; | |
| 5824 | |
| 5825 | |
| 5826 /* We're done for this loop. | |
| 5827 * Go to next channel | |
| 5828 */ | |
| 5829 continue; | |
| 5830 } | |
| 5831 } /* End end-playback */ | |
| 5832 else | |
| 5833 { | |
| 5834 /* Need to run out buffer */ | |
| 5835 #if 1 | |
| 5836 /* Might want to check state */ | |
| 5837 /* In case the playback stopped, | |
| 5838 * we need to resume */ | |
| 5839 alGetSourcei( | |
| 5840 ALmixer_Channel_List[i].alsource, | |
| 5841 AL_SOURCE_STATE, &state | |
| 5842 ); | |
| 5843 if((error = alGetError()) != AL_NO_ERROR) | |
| 5844 { | |
| 5845 fprintf(stderr, "61Testing error: %s\n", | |
|
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|
5846 alGetString(error)); |
| 0 | 5847 } |
| 5848 if(AL_STOPPED == state) | |
| 5849 { | |
|
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diff
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|
5850 /* |
| 0 | 5851 fprintf(stderr, "Shouldn't be here. %d Buffers still in queue, but play stopped. This might be correct though because race conditions could have caused the STOP to happen right after our other tests...Checking queue status...\n", buffers_still_queued); |
|
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|
5852 */ |
| 1 | 5853 /* |
| 0 | 5854 PrintQueueStatus(ALmixer_Channel_List[i].alsource); |
| 1 | 5855 */ |
| 0 | 5856 /* Rather than force unqueuing the buffer, let's see if |
| 5857 * setting the buffer to none works (the OpenAL 1.0 | |
| 5858 * Reference Annotation suggests this should work). | |
| 5859 */ | |
| 5860 alSourcei(ALmixer_Channel_List[i].alsource, | |
| 5861 AL_BUFFER, AL_NONE); | |
| 5862 /* | |
| 5863 PrintQueueStatus(ALmixer_Channel_List[i].alsource); | |
| 5864 */ | |
| 5865 /* This doesn't work because in some cases, I think | |
| 5866 * it causes the sound to be replayed | |
| 5867 */ | |
| 5868 /* | |
| 5869 fprintf(stderr, "Resuming in eof (trying to run out buffers\n"); | |
| 5870 alSourcePlay(ALmixer_Channel_List[i].alsource); | |
| 5871 */ | |
| 5872 } | |
| 5873 #endif | |
| 5874 } /* End trap section */ | |
| 5875 } /* End POST-EOF use-up buffer section */ | |
| 5876 } /* END Streamed section */ | |
| 5877 } /* END channel in use */ | |
| 5878 } /* END for-loop for each channel */ | |
| 5879 | |
| 5880 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 5881 alcProcessContext(alcGetCurrentContext()); | |
| 5882 if((error = alGetError()) != AL_NO_ERROR) | |
| 5883 { | |
| 5884 fprintf(stderr, "62Testing error: %s\n", | |
|
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|
5885 alGetString(error)); |
|
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|
5886 } |
|
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|
5887 #endif |
|
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|
5888 |
|
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|
5889 #ifdef ENABLE_ALMIXER_THREADS |
|
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|
5890 SDL_UnlockMutex(s_simpleLock); |
|
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|
5891 #endif |
| 0 | 5892 /* Return the number of errors */ |
| 5893 if(error_flag < 0) | |
| 5894 { | |
| 5895 return error_flag; | |
| 5896 } | |
| 5897 /* Return the number of buffers that were queued */ | |
| 5898 return retval; | |
| 5899 } | |
| 5900 | |
| 5901 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 5902 /* This is only here so we can call SDL_OpenAudio() */ | |
|
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|
5903 static void my_dummy_audio_callback(void* userdata, ALbyte* stream, int len) |
| 0 | 5904 { |
| 5905 } | |
| 5906 #endif | |
| 5907 | |
| 5908 | |
| 5909 | |
| 5910 | |
| 5911 #ifdef ENABLE_ALMIXER_THREADS | |
| 5912 /* We might need threads. We | |
| 5913 * must constantly poll OpenAL to find out | |
| 5914 * if sound is being streamed, if play has | |
| 5915 * ended, etc. Without threads, this must | |
| 5916 * be explicitly done by the user. | |
| 5917 * We could try to do it for them if we | |
| 5918 * finish the threads. | |
| 5919 */ | |
| 5920 | |
| 5921 static int Stream_Data_Thread_Callback(void* data) | |
| 5922 { | |
|
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|
5923 ALint retval; |
| 0 | 5924 |
| 5925 while(ALmixer_Initialized) | |
| 5926 { | |
| 5927 retval = Update_ALmixer(data); | |
| 5928 /* 0 means that nothing needed updating and | |
| 5929 * the function returned quickly | |
| 5930 */ | |
| 5931 if(0 == retval) | |
| 5932 { | |
| 5933 /* Let's be nice and make the thread sleep since | |
| 5934 * little work was done in update | |
| 5935 */ | |
| 5936 /* Make sure times are multiples of 10 | |
| 5937 * for optimal performance and accuracy in Linux | |
| 5938 */ | |
|
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diff
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|
5939 ALmixer_Delay(10); |
| 0 | 5940 } |
| 5941 else | |
| 5942 { | |
| 5943 /* should I also be sleeping/yielding here? */ | |
|
2
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|
5944 ALmixer_Delay(0); |
| 0 | 5945 } |
| 5946 } | |
|
6
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3
diff
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|
5947 /* |
| 0 | 5948 fprintf(stderr, "Thread is closing\n"); |
|
6
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3
diff
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|
5949 */ |
| 0 | 5950 return 0; |
| 5951 } | |
| 5952 #endif /* End of ENABLE_ALMIXER_THREADS */ | |
| 5953 | |
| 5954 | |
|
2
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diff
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|
5955 /* SDL/SDL_mixer returns -1 on error and 0 on success. |
|
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diff
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|
5956 * I actually prefer false/true conventions (SDL_Sound/OpenAL/GL) |
|
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|
5957 * so SDL_mixer porting people beware. |
| 0 | 5958 * Warning: SDL_QuitSubSystem(SDL_INIT_AUDIO) is called which |
| 5959 * means the SDL audio system will be disabled. It will not | |
| 5960 * be restored (in case SDL is not actually being used) so | |
| 5961 * the user will need to restart it if they need it after | |
| 5962 * OpenAL shuts down. | |
| 5963 */ | |
|
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|
5964 ALboolean ALmixer_Init(ALuint frequency, ALint num_sources, ALuint refresh) |
| 0 | 5965 { |
| 5966 ALCdevice* dev; | |
| 5967 ALCcontext* context; | |
|
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|
5968 ALint i; |
| 0 | 5969 ALenum error; |
| 5970 ALuint* source; | |
| 5971 | |
| 5972 #ifdef USING_LOKI_AL_DIST | |
| 5973 /* The Loki dist requires that I set both the | |
| 5974 * device and context frequency values separately | |
| 5975 */ | |
| 5976 /* Hope this won't overflow */ | |
| 5977 char device_string[256]; | |
| 5978 #endif | |
| 5979 | |
| 5980 /* (Venting frustration) Damn it! Nobody bothered | |
| 5981 * documenting how you're supposed to use an attribute | |
| 5982 * list. In fact, the not even the Loki test program | |
| 5983 * writers seem to know because they use it inconsistently. | |
| 5984 * For example, how do you terminate that attribute list? | |
| 5985 * The Loki test code does it 3 different ways. They | |
| 5986 * set the last value to 0, or they set it to ALC_INVALID, | |
| 5987 * or they set two final values: ALC_INVALID, 0 | |
| 5988 * In Loki, 0 and ALC_INVALID happen to be the same, | |
| 5989 * but with Creative Labs ALC_INVALID is -1. | |
| 5990 * So something's going to break. Loki's source | |
| 5991 * code says to terminate with ALC_INVALID. But I | |
| 5992 * don't know if that's really true, or it happens | |
| 5993 * to be a coinicidence because it's defined to 0. | |
| 5994 * Creative provides no source code, so I can't look at how | |
| 5995 * they terminate it. | |
| 5996 * So this is really, really ticking me off... | |
| 5997 * For now, I'm going to use ALC_INVALID. | |
| 5998 * (Update...after further review of the API spec, | |
| 5999 * it seems that a NULL terminated string is the correct | |
| 6000 * termination value to use, so 0 it is.) | |
| 6001 */ | |
| 6002 #if 0 | |
| 6003 ALint attrlist[] = { | |
| 6004 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
| 6005 /* Don't know anything about these values. | |
| 6006 * Trust defaults? */ | |
| 6007 /* Supposed to be the refresh rate in Hz. | |
| 6008 * I think 15-120 are supposed to be good | |
| 6009 * values. Though I haven't gotten any effect except | |
| 6010 * for one strange instance on a Mac. But it was | |
| 6011 * unrepeatable. | |
| 6012 */ | |
| 6013 #if 0 | |
| 6014 ALC_REFRESH, 15, | |
| 6015 #endif | |
| 6016 /* Sync requires a alcProcessContext() call | |
| 6017 * for every cycle. By default, this is | |
| 6018 * not used and the value is AL_FALSE | |
| 6019 * because it will probably perform | |
| 6020 * pretty badly for me. | |
| 6021 */ | |
| 6022 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6023 ALC_SYNC, AL_TRUE, | |
| 6024 #else | |
| 6025 ALC_SYNC, AL_FALSE, | |
| 6026 #endif | |
| 6027 /* Looking at the API spec, it implies | |
| 6028 * that the list be a NULL terminated string | |
| 6029 * so it's probably not safe to use ALC_INVALID | |
| 6030 */ | |
| 6031 /* | |
| 6032 ALC_INVALID }; | |
| 6033 */ | |
| 6034 '\0'}; | |
| 6035 #endif | |
| 6036 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
| 6037 * However, if no value is specified, I don't | |
| 6038 * want it in the list so I can get the OpenAL defaults | |
| 6039 */ | |
| 6040 ALint attrlist[7]; | |
|
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|
6041 ALsizei current_attrlist_index = 0; |
| 0 | 6042 |
| 6043 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6044 /* More problems: I'm getting bit by endian/signedness issues on | |
| 6045 * different platforms. I can find the endianess easily enough, | |
| 6046 * but I don't know how to determine what the correct signedness | |
| 6047 * is (if such a thing exists). I do know that if I try using | |
| 6048 * unsigned on OSX with an originally signed sample, I get | |
| 6049 * distortion. However, I don't have any native unsigned samples | |
| 6050 * to test. But I'm assuming that the platform must be in the | |
| 6051 * correct signedness no matter what. | |
| 6052 * I can either assume everybody is signed, or I can try to | |
| 6053 * determine the value. If I try to determine the values, | |
| 6054 * I think my only ability to figure it out will be to open | |
| 6055 * SDL_Audio, and read what the obtained settings were. | |
| 6056 * Then shutdown everything. However, I don't even know how | |
| 6057 * reliable this is. | |
| 6058 * Update: I think I resolved the issues...forgot to update | |
| 6059 * these comments when it happened. I should check the revision control | |
| 6060 * log... Anyway, I think the issue was partly related to me not | |
| 6061 * doing something correctly with the AudioInfo or some kind | |
| 6062 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
| 6063 * revision control, I think I might have assumed that SDL_Sound would | |
| 6064 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
| 6065 * and had to fill one out myself. | |
| 6066 */ | |
| 6067 SDL_AudioSpec desired; | |
| 6068 SDL_AudioSpec obtained; | |
| 6069 #endif | |
| 6070 | |
| 6071 | |
| 6072 /* Make sure ALmixer isn't already initialized */ | |
| 6073 if(ALmixer_Initialized) | |
| 6074 { | |
|
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|
6075 return AL_FALSE; |
| 0 | 6076 } |
| 6077 | |
|
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|
6078 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
|
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|
6079 ALmixer_InitTime(); |
|
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|
6080 |
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|
6081 /* Note: The pool may have been created on previous Init's */ |
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|
6082 /* I leave the pool allocated allocated in case the user wants |
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|
6083 * to read the pool in case of a failure (such as in this function). |
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|
6084 * This is not actually a leak. |
|
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|
6085 */ |
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|
6086 if(NULL == s_ALmixerErrorPool) |
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|
6087 { |
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|
6088 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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|
6089 } |
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|
6090 if(NULL == s_ALmixerErrorPool) |
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6091 { |
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|
6092 return AL_FALSE; |
|
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|
6093 } |
|
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|
6094 /* |
|
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|
6095 fprintf(stderr, "tError Test0\n"); |
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|
6096 ALmixer_SetError("Initing (and testing SetError)"); |
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|
6097 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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|
6098 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
|
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6099 */ |
|
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6100 #endif |
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6101 |
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6102 |
| 0 | 6103 /* Set the defaults */ |
|
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6104 /* |
| 0 | 6105 attrlist[0] = ALC_FREQUENCY; |
| 6106 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
| 6107 attrlist[2] = ALC_SYNC; | |
| 6108 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6109 attrlist[3] = ALC_TRUE; | |
| 6110 #else | |
| 6111 attrlist[3] = ALC_FALSE; | |
| 6112 #endif | |
|
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6113 */ |
| 0 | 6114 /* Set frequency value if it is not 0 */ |
| 6115 if(0 != frequency) | |
| 6116 { | |
|
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|
6117 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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6118 current_attrlist_index++; |
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|
6119 attrlist[current_attrlist_index] = (ALint)frequency; |
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|
6120 current_attrlist_index++; |
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6121 } |
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6122 |
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|
6123 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6124 attrlist[current_attrlist_index] = ALC_SYNC; |
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|
6125 current_attrlist_index++; |
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6126 attrlist[current_attrlist_index] = ALC_TRUE; |
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6127 current_attrlist_index++; |
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|
6128 #endif |
| 0 | 6129 |
| 6130 /* If the user specifies a refresh value, | |
| 6131 * make room for it | |
| 6132 */ | |
| 6133 if(0 != refresh) | |
| 6134 { | |
|
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|
6135 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
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|
6136 current_attrlist_index++; |
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|
6137 attrlist[current_attrlist_index] = refresh; |
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|
6138 current_attrlist_index++; |
| 0 | 6139 } |
| 6140 | |
|
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6141 /* End attribute list */ |
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|
6142 attrlist[current_attrlist_index] = '\0'; |
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6143 |
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6144 |
| 0 | 6145 /* Initialize SDL_Sound */ |
| 6146 if(! Sound_Init() ) | |
| 6147 { | |
| 6148 ALmixer_SetError(Sound_GetError()); | |
|
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6149 return AL_FALSE; |
| 0 | 6150 } |
| 6151 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6152 /* Here is the paranoid check that opens | |
| 6153 * SDL audio in an attempt to find the correct | |
| 6154 * system values. | |
| 6155 */ | |
| 6156 /* Doesn't have to be the actual value I think | |
| 6157 * (as long as it doesn't influence format, in | |
| 6158 * which case I'm probably screwed anyway because OpenAL | |
| 6159 * may easily choose to do something else). | |
| 6160 */ | |
| 6161 desired.freq = 44100; | |
| 6162 desired.channels = 2; | |
| 6163 desired.format = AUDIO_S16SYS; | |
| 6164 desired.callback = my_dummy_audio_callback; | |
| 6165 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
| 6166 { | |
| 6167 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
| 6168 /* Now to get really paranoid, we should probably | |
| 6169 * also assume that the 8bit format is also the | |
| 6170 * same sign type and set that value | |
| 6171 */ | |
| 6172 if(AUDIO_S16SYS == obtained.format) | |
| 6173 { | |
| 6174 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
| 6175 } | |
| 6176 /* Should be AUDIO_U16SYS */ | |
| 6177 else | |
| 6178 { | |
| 6179 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
| 6180 } | |
| 6181 SDL_CloseAudio(); | |
| 6182 } | |
| 6183 else | |
| 6184 { | |
| 6185 /* Well, I guess I'm in trouble. I guess it's my best guess | |
| 6186 */ | |
| 6187 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
| 6188 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
| 6189 } | |
| 6190 #endif | |
|
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6191 |
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6192 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
| 0 | 6193 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
| 6194 * causes OpenAL and SMPEG to conflict. For some reason | |
| 6195 * if SDL_Init on audio is active, then all the SMPEG | |
| 6196 * decoded sound comes out silent. Unfortunately, | |
| 6197 * Sound_Init() invokes SDL_Init on audio. I'm | |
| 6198 * not sure why it actually needs it... | |
| 6199 * But we'll attempt to disable it here after the | |
| 6200 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
| 6201 */ | |
| 6202 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
|
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6203 #endif |
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6204 |
| 0 | 6205 /* I'm told NULL will call the default string |
| 6206 * and hopefully do the right thing for each platform | |
| 6207 */ | |
| 6208 /* | |
| 6209 dev = alcOpenDevice( NULL ); | |
| 6210 */ | |
| 6211 /* Now I'm told I need to set both the device and context | |
| 6212 * to have the same sampling rate, so I must pass a string | |
| 6213 * to OpenDevice(). I don't know how portable these strings are. | |
| 6214 * I don't even know if the format for strings is | |
| 6215 * compatible | |
| 6216 * From the testattrib.c in the Loki test section | |
| 6217 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
| 6218 */ | |
| 6219 | |
| 6220 #ifdef USING_LOKI_AL_DIST | |
| 6221 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
| 6222 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
| 6223 #else | |
| 6224 dev = alcOpenDevice( NULL ); | |
| 6225 #endif | |
| 6226 if(NULL == dev) | |
| 6227 { | |
| 6228 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
|
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|
6229 return AL_FALSE; |
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|
6230 } |
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|
6231 |
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|
6232 #ifdef __APPLE__ |
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|
6233 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
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|
6234 /* This extension must be called before the context is created. */ |
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|
6235 if(0 != frequency) |
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|
6236 { |
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|
6237 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
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|
6238 } |
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|
6239 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
|
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|
6240 /* |
|
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|
6241 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
|
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|
6242 */ |
|
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6243 #endif |
| 0 | 6244 |
| 6245 context = alcCreateContext(dev, attrlist); | |
| 6246 if(NULL == context) | |
| 6247 { | |
| 6248 ALmixer_SetError("Cannot create a context OpenAL"); | |
| 6249 alcCloseDevice(dev); | |
|
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|
6250 return AL_FALSE; |
| 0 | 6251 } |
| 6252 | |
| 6253 | |
| 6254 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6255 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6256 * According to Garin Hiebert, this is actually an inconsistency | |
| 6257 * in the Loki version. The function should return a boolean. | |
| 6258 * instead of ALC_NO_ERROR. Garin suggested I check via | |
| 6259 * alcGetError(). | |
| 6260 */ | |
| 6261 /* clear the error */ | |
| 6262 alcGetError(dev); | |
| 6263 alcMakeContextCurrent(context); | |
| 6264 | |
| 6265 error = alcGetError(dev); | |
| 6266 if( (ALC_NO_ERROR != error) ) | |
| 6267 { | |
| 6268 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6269 alcDestroyContext(context); | |
| 6270 alcCloseDevice(dev); | |
|
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6271 return AL_FALSE; |
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6272 } |
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|
6273 |
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|
6274 /* It looks like OpenAL won't let us ask it what |
|
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6275 * the set frequency is, so we need to save our |
|
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6276 * own copy. Yuck. |
|
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|
6277 * Update: J. Valenzuela just updated the Loki |
|
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6278 * dist (2003/01/02) to handle this. |
|
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6279 * The demo is in testattrib.c. |
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6280 */ |
|
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6281 /* |
|
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|
6282 ALmixer_Frequency_global = frequency; |
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6283 */ |
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6284 #ifndef __APPLE__ |
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6285 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
|
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6286 /* |
|
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|
6287 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
|
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6288 */ |
|
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|
6289 #endif |
|
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6290 |
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6291 |
| 0 | 6292 #if 0 |
| 6293 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
| 6294 if(alcGetCurrentContext() != context) | |
| 6295 { | |
| 6296 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6297 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6298 * I think this is a bug in the OpenAL implementation. | |
| 6299 */ | |
| 6300 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
| 6301 | |
| 6302 fprintf(stderr, "Making context current\n"); | |
| 6303 #ifndef __APPLE__ | |
| 6304 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
| 6305 #else | |
| 6306 if(!alcMakeContextCurrent(context)) | |
| 6307 #endif | |
| 6308 { | |
| 6309 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6310 alcDestroyContext(context); | |
| 6311 alcCloseDevice(dev); | |
|
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6312 return AL_FALSE; |
| 0 | 6313 } |
| 6314 } | |
| 6315 #endif | |
| 6316 | |
| 6317 | |
| 6318 /* #endif */ | |
| 6319 /* Saw this in the README with the OS X OpenAL distribution. | |
| 6320 * It looked interesting and simple, so I thought I might | |
| 6321 * try it out. | |
| 6322 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
| 6323 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
| 6324 * Audio format, the audio data passed to the | |
| 6325 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
| 6326 * usage by removing an audio data conversion | |
| 6327 * (per source) at render timem at the expense of a larger memory footprint. | |
| 6328 * | |
| 6329 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
| 6330 * setting will be applied to all subsequent | |
| 6331 * calls to alBufferData(). | |
| 6332 */ | |
| 6333 #ifdef __APPLE__ | |
|
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6334 /* |
|
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6335 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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6336 |
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6337 #else |
|
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6338 #endif |
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6339 */ |
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|
6340 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
|
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6341 /* |
|
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6342 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
|
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6343 */ |
|
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6344 if(0 != convert_data_enum) |
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6345 { |
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6346 alEnable(convert_data_enum); |
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6347 } |
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6348 if( (AL_NO_ERROR != alGetError()) ) |
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6349 { |
|
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6350 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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6351 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
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6352 } |
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6353 |
| 0 | 6354 #endif |
| 6355 | |
| 6356 | |
| 6357 | |
| 6358 | |
| 6359 ALmixer_Initialized = 1; | |
| 6360 | |
| 6361 if(num_sources <= 0) | |
| 6362 { | |
| 6363 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 6364 } | |
| 6365 else | |
| 6366 { | |
| 6367 Number_of_Channels_global = num_sources; | |
| 6368 } | |
| 6369 Number_of_Reserve_Channels_global = 0; | |
| 6370 Is_Playing_global = 0; | |
| 6371 /* Set to Null in case system quit and was reinitialized */ | |
| 6372 Channel_Done_Callback = NULL; | |
| 6373 Channel_Done_Callback_Userdata = NULL; | |
| 6374 Channel_Data_Callback = NULL; | |
| 1 | 6375 Channel_Data_Callback_Userdata = NULL; |
| 0 | 6376 |
| 6377 /* Allocate memory for the list of channels */ | |
| 6378 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
| 6379 if(NULL == ALmixer_Channel_List) | |
| 6380 { | |
| 6381 ALmixer_SetError("Out of Memory for Channel List"); | |
| 6382 alcDestroyContext(context); | |
| 6383 alcCloseDevice(dev); | |
| 6384 ALmixer_Initialized = 0; | |
| 6385 Number_of_Channels_global = 0; | |
|
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6386 return AL_FALSE; |
| 0 | 6387 } |
| 6388 | |
| 6389 /* Allocate memory for the list of sources that map to the channels */ | |
| 6390 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
| 6391 if(NULL == Source_Map_List) | |
| 6392 { | |
| 6393 ALmixer_SetError("Out of Memory for Source Map List"); | |
| 6394 free(ALmixer_Channel_List); | |
| 6395 alcDestroyContext(context); | |
| 6396 alcCloseDevice(dev); | |
| 6397 ALmixer_Initialized = 0; | |
| 6398 Number_of_Channels_global = 0; | |
|
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6399 return AL_FALSE; |
| 0 | 6400 } |
| 6401 | |
| 6402 /* Create array that will hold the sources */ | |
| 6403 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
| 6404 if(NULL == source) | |
| 6405 { | |
| 6406 ALmixer_SetError("Out of Memory for sources"); | |
| 6407 free(Source_Map_List); | |
| 6408 free(ALmixer_Channel_List); | |
| 6409 alcDestroyContext(context); | |
| 6410 alcCloseDevice(dev); | |
| 6411 ALmixer_Initialized = 0; | |
| 6412 Number_of_Channels_global = 0; | |
|
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6413 return AL_FALSE; |
| 0 | 6414 } |
| 6415 | |
| 6416 /* Clear the error state */ | |
| 6417 alGetError(); | |
| 6418 /* Generate the OpenAL sources */ | |
| 6419 alGenSources(Number_of_Channels_global, source); | |
| 6420 if( (error=alGetError()) != AL_NO_ERROR) | |
| 6421 { | |
|
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6422 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
| 0 | 6423 free(ALmixer_Channel_List); |
| 6424 free(Source_Map_List); | |
| 6425 alcDestroyContext(context); | |
| 6426 alcCloseDevice(dev); | |
| 6427 ALmixer_Initialized = 0; | |
| 6428 Number_of_Channels_global = 0; | |
|
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6429 return AL_FALSE; |
| 0 | 6430 } |
| 6431 | |
| 6432 /* Initialize each channel and associate one source to one channel */ | |
| 6433 for(i=0; i<Number_of_Channels_global; i++) | |
| 6434 { | |
| 6435 if(0 == source[i]) | |
| 6436 { | |
| 6437 fprintf(stderr, "SDL_ALmixer serious problem. This OpenAL implementation allowed 0 to be a valid source id which is in conflict with assumptions made in this library.\n"); | |
| 6438 } | |
| 6439 | |
| 6440 Init_Channel(i); | |
| 6441 /* Keeping the source allocation out of the Init function | |
| 6442 * in case I want to reuse the Init | |
| 6443 * function for resetting data | |
| 6444 */ | |
| 6445 ALmixer_Channel_List[i].alsource = source[i]; | |
| 6446 /* Now also keep a copy of the source to channel mapping | |
| 6447 * in case we need to look up a channel from the source | |
| 6448 * instead of a source from a channel | |
| 6449 */ | |
| 6450 Source_Map_List[i].source = source[i]; | |
| 6451 Source_Map_List[i].channel = i; | |
| 6452 /* Clean the channel because there are some things that need to | |
| 6453 * be done that can't happen until the source is set | |
| 6454 */ | |
| 6455 Clean_Channel(i); | |
| 6456 } | |
| 6457 | |
| 6458 /* The Source_Map_List must be sorted by source for binary searches | |
| 6459 */ | |
| 6460 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 6461 | |
|
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6462 /* |
|
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6463 ALmixer_OutputDecoders(); |
|
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6464 */ |
|
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6465 #ifdef ENABLE_ALMIXER_THREADS |
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6466 s_simpleLock = SDL_CreateMutex(); |
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6467 if(NULL == s_simpleLock) |
| 0 | 6468 { |
| 6469 /* SDL sets the error message already? */ | |
| 6470 free(source); | |
| 6471 free(ALmixer_Channel_List); | |
| 6472 free(Source_Map_List); | |
| 6473 alcDestroyContext(context); | |
| 6474 alcCloseDevice(dev); | |
| 6475 ALmixer_Initialized = 0; | |
| 6476 Number_of_Channels_global = 0; | |
|
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6477 return AL_FALSE; |
| 0 | 6478 } |
| 6479 | |
| 6480 | |
| 6481 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
| 6482 if(NULL == Stream_Thread_global) | |
| 6483 { | |
| 6484 /* SDL sets the error message already? */ | |
|
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|
6485 SDL_DestroyMutex(s_simpleLock); |
| 0 | 6486 free(source); |
| 6487 free(ALmixer_Channel_List); | |
| 6488 free(Source_Map_List); | |
| 6489 alcDestroyContext(context); | |
| 6490 alcCloseDevice(dev); | |
| 6491 ALmixer_Initialized = 0; | |
| 6492 Number_of_Channels_global = 0; | |
|
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6493 return AL_FALSE; |
| 0 | 6494 } |
| 6495 | |
|
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6496 /* |
| 0 | 6497 fprintf(stderr, "Using threads\n"); |
|
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6498 */ |
| 0 | 6499 #endif /* End of ENABLE_ALMIXER_THREADS */ |
| 6500 | |
| 6501 /* We don't need this array any more because all the sources | |
| 6502 * are connected to channels | |
| 6503 */ | |
| 6504 free(source); | |
|
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6505 return AL_TRUE; |
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6506 } |
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6507 |
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6508 |
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6509 ALboolean ALmixer_InitContext(ALuint frequency, ALuint refresh) |
| 0 | 6510 { |
| 6511 ALCdevice* dev; | |
| 6512 ALCcontext* context; | |
| 6513 ALCenum error; | |
| 6514 | |
| 6515 #ifdef USING_LOKI_AL_DIST | |
| 6516 /* The Loki dist requires that I set both the | |
| 6517 * device and context frequency values separately | |
| 6518 */ | |
| 6519 /* Hope this won't overflow */ | |
| 6520 char device_string[256]; | |
| 6521 #endif | |
| 6522 | |
| 6523 /* (Venting frustration) Damn it! Nobody bothered | |
| 6524 * documenting how you're supposed to use an attribute | |
| 6525 * list. In fact, the not even the Loki test program | |
| 6526 * writers seem to know because they use it inconsistently. | |
| 6527 * For example, how do you terminate that attribute list? | |
| 6528 * The Loki test code does it 3 different ways. They | |
| 6529 * set the last value to 0, or they set it to ALC_INVALID, | |
| 6530 * or they set two final values: ALC_INVALID, 0 | |
| 6531 * In Loki, 0 and ALC_INVALID happen to be the same, | |
| 6532 * but with Creative Labs ALC_INVALID is -1. | |
| 6533 * So something's going to break. Loki's source | |
| 6534 * code says to terminate with ALC_INVALID. But I | |
| 6535 * don't know if that's really true, or it happens | |
| 6536 * to be a coinicidence because it's defined to 0. | |
| 6537 * Creative provides no source code, so I can't look at how | |
| 6538 * they terminate it. | |
| 6539 * So this is really, really ticking me off... | |
| 6540 * For now, I'm going to use ALC_INVALID. | |
| 6541 * (Update...after further review of the API spec, | |
| 6542 * it seems that a NULL terminated string is the correct | |
| 6543 * termination value to use, so 0 it is.) | |
| 6544 */ | |
| 6545 #if 0 | |
| 6546 ALint attrlist[] = { | |
| 6547 ALC_FREQUENCY, ALMIXER_DEFAULT_FREQUENCY, | |
| 6548 /* Don't know anything about these values. | |
| 6549 * Trust defaults? */ | |
| 6550 /* Supposed to be the refresh rate in Hz. | |
| 6551 * I think 15-120 are supposed to be good | |
| 6552 * values. Though I haven't gotten any effect except | |
| 6553 * for one strange instance on a Mac. But it was | |
| 6554 * unrepeatable. | |
| 6555 */ | |
| 6556 #if 0 | |
| 6557 ALC_REFRESH, 15, | |
| 6558 #endif | |
| 6559 /* Sync requires a alcProcessContext() call | |
| 6560 * for every cycle. By default, this is | |
| 6561 * not used and the value is AL_FALSE | |
| 6562 * because it will probably perform | |
| 6563 * pretty badly for me. | |
| 6564 */ | |
| 6565 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6566 ALC_SYNC, AL_TRUE, | |
| 6567 #else | |
| 6568 ALC_SYNC, AL_FALSE, | |
| 6569 #endif | |
| 6570 /* Looking at the API spec, it implies | |
| 6571 * that the list be a NULL terminated string | |
| 6572 * so it's probably not safe to use ALC_INVALID | |
| 6573 */ | |
| 6574 /* | |
| 6575 ALC_INVALID }; | |
| 6576 */ | |
| 6577 '\0'}; | |
| 6578 #endif | |
| 6579 /* Redo: I'm going to allow ALC_REFRESH to be set. | |
| 6580 * However, if no value is specified, I don't | |
| 6581 * want it in the list so I can get the OpenAL defaults | |
| 6582 */ | |
| 6583 ALint attrlist[7]; | |
|
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6584 ALsizei current_attrlist_index = 0; |
| 0 | 6585 |
| 6586 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6587 /* More problems: I'm getting bit by endian/signedness issues on | |
| 6588 * different platforms. I can find the endianess easily enough, | |
| 6589 * but I don't know how to determine what the correct signedness | |
| 6590 * is (if such a thing exists). I do know that if I try using | |
| 6591 * unsigned on OSX with an originally signed sample, I get | |
| 6592 * distortion. However, I don't have any native unsigned samples | |
| 6593 * to test. But I'm assuming that the platform must be in the | |
| 6594 * correct signedness no matter what. | |
| 6595 * I can either assume everybody is signed, or I can try to | |
| 6596 * determine the value. If I try to determine the values, | |
| 6597 * I think my only ability to figure it out will be to open | |
| 6598 * SDL_Audio, and read what the obtained settings were. | |
| 6599 * Then shutdown everything. However, I don't even know how | |
| 6600 * reliable this is. | |
| 6601 * Update: I think I resolved the issues...forgot to update | |
| 6602 * these comments when it happened. I should check the revision control | |
| 6603 * log... Anyway, I think the issue was partly related to me not | |
| 6604 * doing something correctly with the AudioInfo or some kind | |
| 6605 * of stupid endian bug in my code, and weirdness ensued. Looking at the | |
| 6606 * revision control, I think I might have assumed that SDL_Sound would | |
| 6607 * do the right thing with a NULL AudioInfo, but I was incorrect, | |
| 6608 * and had to fill one out myself. | |
| 6609 */ | |
| 6610 SDL_AudioSpec desired; | |
| 6611 SDL_AudioSpec obtained; | |
| 6612 #endif | |
| 6613 | |
| 6614 | |
| 6615 | |
| 6616 | |
| 6617 /* Make sure ALmixer isn't already initialized */ | |
| 6618 if(ALmixer_Initialized) | |
| 6619 { | |
|
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6620 return AL_FALSE; |
| 0 | 6621 } |
| 6622 | |
| 6623 /* Set the defaults */ | |
| 6624 attrlist[0] = ALC_FREQUENCY; | |
| 6625 attrlist[1] = ALMIXER_DEFAULT_FREQUENCY; | |
| 6626 attrlist[2] = ALC_SYNC; | |
| 6627 #ifdef ENABLE_ALMIXER_ALC_SYNC | |
| 6628 attrlist[3] = ALC_TRUE; | |
| 6629 #else | |
| 6630 attrlist[3] = ALC_FALSE; | |
| 6631 #endif | |
| 6632 /* Set frequency value if it is not 0 */ | |
| 6633 if(0 != frequency) | |
| 6634 { | |
|
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6635 attrlist[current_attrlist_index] = ALC_FREQUENCY; |
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6636 current_attrlist_index++; |
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6637 attrlist[current_attrlist_index] = (ALint)frequency; |
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6638 current_attrlist_index++; |
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|
6639 } |
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6640 |
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|
6641 #ifdef ENABLE_ALMIXER_ALC_SYNC |
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6642 attrlist[current_attrlist_index] = ALC_SYNC; |
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|
6643 current_attrlist_index++; |
|
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|
6644 attrlist[current_attrlist_index] = ALC_TRUE; |
|
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|
6645 current_attrlist_index++; |
|
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|
6646 #endif |
| 0 | 6647 |
| 6648 /* If the user specifies a refresh value, | |
| 6649 * make room for it | |
| 6650 */ | |
| 6651 if(0 != refresh) | |
| 6652 { | |
|
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|
6653 attrlist[current_attrlist_index] = (ALint)ALC_REFRESH; |
|
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|
6654 current_attrlist_index++; |
|
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|
6655 attrlist[current_attrlist_index] = refresh; |
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|
6656 current_attrlist_index++; |
| 0 | 6657 } |
| 6658 | |
|
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|
6659 /* End attribute list */ |
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|
6660 attrlist[current_attrlist_index] = '\0'; |
|
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|
6661 |
|
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|
6662 |
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6663 |
| 0 | 6664 /* Initialize SDL_Sound */ |
| 6665 if(! Sound_Init() ) | |
| 6666 { | |
| 6667 ALmixer_SetError(Sound_GetError()); | |
|
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|
6668 return AL_FALSE; |
| 0 | 6669 } |
| 6670 #ifdef ENABLE_PARANOID_SIGNEDNESS_CHECK | |
| 6671 /* Here is the paranoid check that opens | |
| 6672 * SDL audio in an attempt to find the correct | |
| 6673 * system values. | |
| 6674 */ | |
| 6675 /* Doesn't have to be the actual value I think | |
| 6676 * (as long as it doesn't influence format, in | |
| 6677 * which case I'm probably screwed anyway because OpenAL | |
| 6678 * may easily choose to do something else). | |
| 6679 */ | |
| 6680 desired.freq = 44100; | |
| 6681 desired.channels = 2; | |
| 6682 desired.format = AUDIO_S16SYS; | |
| 6683 desired.callback = my_dummy_audio_callback; | |
| 6684 if(SDL_OpenAudio(&desired, &obtained) >= 0) | |
| 6685 { | |
| 6686 SIGN_TYPE_16BIT_FORMAT = obtained.format; | |
| 6687 /* Now to get really paranoid, we should probably | |
| 6688 * also assume that the 8bit format is also the | |
| 6689 * same sign type and set that value | |
| 6690 */ | |
| 6691 if(AUDIO_S16SYS == obtained.format) | |
| 6692 { | |
| 6693 SIGN_TYPE_8BIT_FORMAT = AUDIO_S8; | |
| 6694 } | |
| 6695 /* Should be AUDIO_U16SYS */ | |
| 6696 else | |
| 6697 { | |
| 6698 SIGN_TYPE_8BIT_FORMAT = AUDIO_U8; | |
| 6699 } | |
| 6700 SDL_CloseAudio(); | |
| 6701 } | |
| 6702 else | |
| 6703 { | |
| 6704 /* Well, I guess I'm in trouble. I guess it's my best guess | |
| 6705 */ | |
| 6706 SIGN_TYPE_16_BIT_FORMAT = AUDIO_S16SYS; | |
| 6707 SIGN_TYPE_8_BIT_FORMAT = AUDIO_S8; | |
| 6708 } | |
| 6709 #endif | |
|
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6710 |
|
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|
6711 #ifndef ALMIXER_COMPILE_WITHOUT_SDL |
| 0 | 6712 /* Weirdness: It seems that SDL_Init(SDL_INIT_AUDIO) |
| 6713 * causes OpenAL and SMPEG to conflict. For some reason | |
| 6714 * if SDL_Init on audio is active, then all the SMPEG | |
| 6715 * decoded sound comes out silent. Unfortunately, | |
| 6716 * Sound_Init() invokes SDL_Init on audio. I'm | |
| 6717 * not sure why it actually needs it... | |
| 6718 * But we'll attempt to disable it here after the | |
| 6719 * SDL_Sound::Init call and hope it doesn't break SDL_Sound. | |
| 6720 */ | |
| 6721 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
|
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|
6722 #endif |
|
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|
6723 |
| 0 | 6724 /* I'm told NULL will call the default string |
| 6725 * and hopefully do the right thing for each platform | |
| 6726 */ | |
| 6727 /* | |
| 6728 dev = alcOpenDevice( NULL ); | |
| 6729 */ | |
| 6730 /* Now I'm told I need to set both the device and context | |
| 6731 * to have the same sampling rate, so I must pass a string | |
| 6732 * to OpenDevice(). I don't know how portable these strings are. | |
| 6733 * I don't even know if the format for strings is | |
| 6734 * compatible | |
| 6735 * From the testattrib.c in the Loki test section | |
| 6736 * dev = alcOpenDevice( (const ALubyte *) "'((sampling-rate 22050))" ); | |
| 6737 */ | |
| 6738 | |
| 6739 #ifdef USING_LOKI_AL_DIST | |
| 6740 sprintf(device_string, "'((sampling-rate %d))", attrlist[1]); | |
| 6741 dev = alcOpenDevice( (const ALubyte *) device_string ); | |
| 6742 #else | |
| 6743 dev = alcOpenDevice( NULL ); | |
| 6744 #endif | |
| 6745 if(NULL == dev) | |
| 6746 { | |
| 6747 ALmixer_SetError("Cannot open sound device for OpenAL"); | |
|
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|
6748 return AL_FALSE; |
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|
6749 } |
|
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|
6750 |
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|
6751 #ifdef __APPLE__ |
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|
6752 /* The ALC_FREQUENCY attribute is ignored with Apple's implementation. */ |
|
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|
6753 /* This extension must be called before the context is created. */ |
|
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|
6754 if(0 != frequency) |
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|
6755 { |
|
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|
6756 Internal_alcMacOSXMixerOutputRate((ALdouble)frequency); |
|
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|
6757 } |
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|
6758 ALmixer_Frequency_global = (ALuint)Internal_alcMacOSXGetMixerOutputRate(); |
|
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|
6759 /* |
|
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|
6760 fprintf(stderr, "Internal_alcMacOSXMixerOutputRate is: %lf", Internal_alcMacOSXGetMixerOutputRate()); |
|
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|
6761 */ |
|
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|
6762 #endif |
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|
6763 |
| 0 | 6764 |
| 6765 context = alcCreateContext(dev, attrlist); | |
| 6766 if(NULL == context) | |
| 6767 { | |
| 6768 ALmixer_SetError("Cannot create a context OpenAL"); | |
| 6769 alcCloseDevice(dev); | |
|
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|
6770 return AL_FALSE; |
| 0 | 6771 } |
| 6772 | |
| 6773 | |
| 6774 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6775 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6776 * According to Garin Hiebert, this is actually an inconsistency | |
| 6777 * in the Loki version. The function should return a boolean. | |
| 6778 * instead of ALC_NO_ERROR. Garin suggested I check via | |
| 6779 * alcGetError(). | |
| 6780 */ | |
| 6781 /* clear the error */ | |
| 6782 alcGetError(dev); | |
| 6783 alcMakeContextCurrent(context); | |
| 6784 | |
| 6785 error = alcGetError(dev); | |
| 6786 if( (ALC_NO_ERROR != error) ) | |
| 6787 { | |
| 6788 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6789 alcDestroyContext(context); | |
| 6790 alcCloseDevice(dev); | |
|
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|
6791 return AL_FALSE; |
| 0 | 6792 } |
| 6793 | |
| 6794 | |
| 6795 #if 0 | |
| 6796 /* OSX is failing on alcMakeContextCurrent(). Try checking it first? */ | |
| 6797 if(alcGetCurrentContext() != context) | |
| 6798 { | |
| 6799 /* Hmmm, OSX is returning 1 on alcMakeContextCurrent, | |
| 6800 * but ALC_NO_ERROR is defined to ALC_FALSE. | |
| 6801 * I think this is a bug in the OpenAL implementation. | |
| 6802 */ | |
| 6803 fprintf(stderr,"alcMakeContextCurrent returns %d\n", alcMakeContextCurrent(context)); | |
| 6804 | |
| 6805 fprintf(stderr, "Making context current\n"); | |
| 6806 #ifndef __APPLE__ | |
| 6807 if(alcMakeContextCurrent(context) != ALC_NO_ERROR) | |
| 6808 #else | |
| 6809 if(!alcMakeContextCurrent(context)) | |
| 6810 #endif | |
| 6811 { | |
| 6812 ALmixer_SetError("Could not MakeContextCurrent"); | |
| 6813 alcDestroyContext(context); | |
| 6814 alcCloseDevice(dev); | |
|
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|
6815 return AL_FALSE; |
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6816 } |
|
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|
6817 |
|
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|
6818 } |
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|
6819 #endif |
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6820 |
|
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|
6821 /* It looks like OpenAL won't let us ask it what |
|
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|
6822 * the set frequency is, so we need to save our |
|
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|
6823 * own copy. Yuck. |
|
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|
6824 * Update: J. Valenzuela just updated the Loki |
|
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|
6825 * dist (2003/01/02) to handle this. |
|
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|
6826 * The demo is in testattrib.c. |
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|
6827 */ |
|
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|
6828 #ifndef __APPLE__ |
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|
6829 alcGetIntegerv(dev, ALC_FREQUENCY, 1, &ALmixer_Frequency_global); |
|
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|
6830 /* |
|
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|
6831 fprintf(stderr, "alcGetIntegerv ALC_FREQUENCY is: %d", ALmixer_Frequency_global); |
|
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6832 */ |
|
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|
6833 #endif |
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6834 |
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6835 |
| 0 | 6836 |
| 6837 /* Saw this in the README with the OS X OpenAL distribution. | |
| 6838 * It looked interesting and simple, so I thought I might | |
| 6839 * try it out. | |
| 6840 * ***** ALC_CONVERT_DATA_UPON_LOADING | |
| 6841 * This extension allows the caller to tell OpenAL to preconvert to the native Core | |
| 6842 * Audio format, the audio data passed to the | |
| 6843 * library with the alBufferData() call. Preconverting the audio data, reduces CPU | |
| 6844 * usage by removing an audio data conversion | |
| 6845 * (per source) at render timem at the expense of a larger memory footprint. | |
| 6846 * | |
| 6847 * This feature is toggled on/off by using the alDisable() & alEnable() APIs. This | |
| 6848 * setting will be applied to all subsequent | |
| 6849 * calls to alBufferData(). | |
| 6850 */ | |
| 6851 #ifdef __APPLE__ | |
|
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|
6852 /* |
|
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|
6853 #if (TARGET_OS_IPHONE == 1) || (TARGET_IPHONE_SIMULATOR == 1) |
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|
6854 |
|
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|
6855 #else |
|
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|
6856 #endif |
|
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|
6857 */ |
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|
6858 ALenum convert_data_enum = alcGetEnumValue(dev, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING"); |
|
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|
6859 /* |
|
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|
6860 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING=0x%x", convert_data_enum); |
|
6
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|
6861 */ |
|
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|
6862 if(0 != convert_data_enum) |
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|
6863 { |
|
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|
6864 alEnable(convert_data_enum); |
|
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6865 } |
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|
6866 if( (AL_NO_ERROR != alGetError()) ) |
|
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|
6867 { |
|
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|
6868 fprintf(stderr, "ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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|
6869 ALmixer_SetError("ALC_MAC_OSX_CONVERT_DATA_UPON_LOADING attempted but failed"); |
|
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|
6870 } |
|
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|
6871 #endif |
|
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|
6872 |
|
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|
6873 return AL_TRUE; |
|
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|
6874 } |
|
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|
6875 |
|
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|
6876 |
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|
6877 ALboolean ALmixer_InitMixer(ALint num_sources) |
| 0 | 6878 { |
|
2
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|
6879 ALint i; |
| 0 | 6880 ALenum error; |
| 6881 ALuint* source; | |
| 6882 | |
| 6883 | |
| 6884 ALmixer_Initialized = 1; | |
| 6885 | |
|
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6886 |
|
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6887 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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6888 ALmixer_InitTime(); |
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6889 |
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6890 /* Note: The pool may have been created on previous Init's */ |
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6891 /* I leave the pool allocated allocated in case the user wants |
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6892 * to read the pool in case of a failure (such as in this function). |
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6893 * This is not actually a leak. |
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6894 */ |
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6895 if(NULL == s_ALmixerErrorPool) |
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6896 { |
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6897 s_ALmixerErrorPool = TError_CreateErrorPool(); |
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6898 } |
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6899 if(NULL == s_ALmixerErrorPool) |
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6900 { |
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6901 return AL_FALSE; |
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6902 } |
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6903 /* |
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6904 fprintf(stderr, "tError Test0\n"); |
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6905 ALmixer_SetError("Initing (and testing SetError)"); |
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6906 fprintf(stderr, "tError Test1: %s\n", ALmixer_GetError()); |
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6907 fprintf(stderr, "tError Test2: %s\n", ALmixer_GetError()); |
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6908 */ |
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6909 #endif |
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6910 |
| 0 | 6911 if(num_sources <= 0) |
| 6912 { | |
| 6913 Number_of_Channels_global = ALMIXER_DEFAULT_NUM_CHANNELS; | |
| 6914 } | |
| 6915 else | |
| 6916 { | |
| 6917 Number_of_Channels_global = num_sources; | |
| 6918 } | |
| 6919 Number_of_Reserve_Channels_global = 0; | |
| 6920 Is_Playing_global = 0; | |
| 6921 /* Set to Null in case system quit and was reinitialized */ | |
| 6922 Channel_Done_Callback = NULL; | |
| 6923 Channel_Done_Callback_Userdata = NULL; | |
| 6924 Channel_Data_Callback = NULL; | |
| 1 | 6925 Channel_Data_Callback_Userdata = NULL; |
| 0 | 6926 |
| 6927 /* Allocate memory for the list of channels */ | |
| 6928 ALmixer_Channel_List = (struct ALmixer_Channel*) malloc(Number_of_Channels_global * sizeof(struct ALmixer_Channel)); | |
| 6929 if(NULL == ALmixer_Channel_List) | |
| 6930 { | |
| 6931 ALmixer_SetError("Out of Memory for Channel List"); | |
| 6932 ALmixer_Initialized = 0; | |
| 6933 Number_of_Channels_global = 0; | |
|
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6934 return AL_FALSE; |
| 0 | 6935 } |
| 6936 | |
| 6937 /* Allocate memory for the list of sources that map to the channels */ | |
| 6938 Source_Map_List = (Source_Map*) malloc(Number_of_Channels_global * sizeof(Source_Map)); | |
| 6939 if(NULL == Source_Map_List) | |
| 6940 { | |
| 6941 ALmixer_SetError("Out of Memory for Source Map List"); | |
| 6942 free(ALmixer_Channel_List); | |
| 6943 ALmixer_Initialized = 0; | |
| 6944 Number_of_Channels_global = 0; | |
|
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6945 return AL_FALSE; |
| 0 | 6946 } |
| 6947 | |
| 6948 /* Create array that will hold the sources */ | |
| 6949 source = (ALuint*)malloc(Number_of_Channels_global * sizeof(ALuint)); | |
| 6950 if(NULL == source) | |
| 6951 { | |
| 6952 ALmixer_SetError("Out of Memory for sources"); | |
| 6953 free(Source_Map_List); | |
| 6954 free(ALmixer_Channel_List); | |
| 6955 ALmixer_Initialized = 0; | |
| 6956 Number_of_Channels_global = 0; | |
|
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6957 return AL_FALSE; |
| 0 | 6958 } |
| 6959 | |
| 6960 /* Clear the error state */ | |
| 6961 alGetError(); | |
| 6962 /* Generate the OpenAL sources */ | |
| 6963 alGenSources(Number_of_Channels_global, source); | |
| 6964 if( (error=alGetError()) != AL_NO_ERROR) | |
| 6965 { | |
|
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6966 ALmixer_SetError("Couldn't generate sources: %s\n", alGetString(error)); |
| 0 | 6967 free(ALmixer_Channel_List); |
| 6968 free(Source_Map_List); | |
| 6969 ALmixer_Initialized = 0; | |
| 6970 Number_of_Channels_global = 0; | |
|
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6971 return AL_FALSE; |
| 0 | 6972 } |
| 6973 | |
| 6974 /* Initialize each channel and associate one source to one channel */ | |
| 6975 for(i=0; i<Number_of_Channels_global; i++) | |
| 6976 { | |
| 6977 Init_Channel(i); | |
| 6978 /* Keeping the source allocation out of the Init function | |
| 6979 * in case I want to reuse the Init | |
| 6980 * function for resetting data | |
| 6981 */ | |
| 6982 ALmixer_Channel_List[i].alsource = source[i]; | |
| 6983 /* Now also keep a copy of the source to channel mapping | |
| 6984 * in case we need to look up a channel from the source | |
| 6985 * instead of a source from a channel | |
| 6986 */ | |
| 6987 Source_Map_List[i].source = source[i]; | |
| 6988 Source_Map_List[i].channel = i; | |
| 6989 /* Clean the channel because there are some things that need to | |
| 6990 * be done that can't happen until the source is set | |
| 6991 */ | |
| 6992 Clean_Channel(i); | |
| 6993 } | |
| 6994 | |
| 6995 /* The Source_Map_List must be sorted by source for binary searches | |
| 6996 */ | |
| 6997 qsort(Source_Map_List, Number_of_Channels_global, sizeof(Source_Map), Compare_Source_Map); | |
| 6998 | |
|
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6999 |
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7000 #ifdef ENABLE_ALMIXER_THREADS |
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7001 s_simpleLock = SDL_CreateMutex(); |
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7002 if(NULL == s_simpleLock) |
| 0 | 7003 { |
| 7004 /* SDL sets the error message already? */ | |
| 7005 free(source); | |
| 7006 free(ALmixer_Channel_List); | |
| 7007 free(Source_Map_List); | |
| 7008 ALmixer_Initialized = 0; | |
| 7009 Number_of_Channels_global = 0; | |
|
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7010 return AL_FALSE; |
| 0 | 7011 } |
| 7012 | |
| 7013 | |
| 7014 Stream_Thread_global = SDL_CreateThread(Stream_Data_Thread_Callback, NULL); | |
| 7015 if(NULL == Stream_Thread_global) | |
| 7016 { | |
| 7017 /* SDL sets the error message already? */ | |
|
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7018 SDL_DestroyMutex(s_simpleLock); |
| 0 | 7019 free(source); |
| 7020 free(ALmixer_Channel_List); | |
| 7021 free(Source_Map_List); | |
| 7022 ALmixer_Initialized = 0; | |
| 7023 Number_of_Channels_global = 0; | |
|
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7024 return AL_FALSE; |
| 0 | 7025 } |
| 7026 | |
|
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7027 /* |
| 0 | 7028 fprintf(stderr, "Using threads\n"); |
|
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7029 */ |
| 0 | 7030 #endif /* End of ENABLE_ALMIXER_THREADS */ |
| 7031 | |
| 7032 /* We don't need this array any more because all the sources | |
| 7033 * are connected to channels | |
| 7034 */ | |
| 7035 free(source); | |
|
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7036 return AL_TRUE; |
| 0 | 7037 } |
| 7038 | |
| 7039 | |
| 7040 | |
| 7041 /* Keep the return value void to allow easy use with | |
| 7042 * atexit() | |
| 7043 */ | |
| 7044 void ALmixer_Quit() | |
| 7045 { | |
| 7046 ALCcontext* context; | |
| 7047 ALCdevice* dev; | |
|
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7048 ALint i; |
| 0 | 7049 |
| 7050 if( ! ALmixer_Initialized) | |
| 7051 { | |
| 7052 return; | |
| 7053 } | |
|
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7054 #ifdef ENABLE_ALMIXER_THREADS |
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7055 SDL_LockMutex(s_simpleLock); |
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7056 #endif |
| 0 | 7057 /* Shutdown everything before closing context */ |
|
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7058 Internal_HaltChannel(-1, AL_FALSE); |
| 0 | 7059 |
| 7060 /* This flag will cause the thread to terminate */ | |
| 7061 ALmixer_Initialized = 0; | |
| 7062 #ifdef ENABLE_ALMIXER_THREADS | |
|
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7063 SDL_UnlockMutex(s_simpleLock); |
| 0 | 7064 SDL_WaitThread(Stream_Thread_global, NULL); |
| 7065 | |
|
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7066 SDL_DestroyMutex(s_simpleLock); |
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7067 #endif |
| 0 | 7068 |
| 7069 /* Delete all the OpenAL sources */ | |
| 7070 for(i=0; i<Number_of_Channels_global; i++) | |
| 7071 { | |
| 7072 alDeleteSources(1, &ALmixer_Channel_List[i].alsource); | |
| 7073 } | |
| 7074 /* Delete all the channels */ | |
| 7075 free(ALmixer_Channel_List); | |
| 7076 free(Source_Map_List); | |
| 7077 | |
| 7078 /* Reset the Number_of_Channels just in case somebody | |
| 7079 * tries using a ALmixer function. | |
| 7080 * I probably should put "Initialized" checks everywhere, | |
| 7081 * but I'm too lazy at the moment. | |
| 7082 */ | |
| 7083 Number_of_Channels_global = 0; | |
| 7084 | |
| 7085 context = alcGetCurrentContext(); | |
| 7086 if(NULL == context) | |
| 7087 { | |
| 7088 return; | |
| 7089 } | |
| 7090 /* Need to get the device before I close the context */ | |
| 7091 dev = alcGetContextsDevice(context); | |
| 7092 alcDestroyContext(context); | |
| 7093 | |
| 7094 if(NULL == dev) | |
| 7095 { | |
| 7096 return; | |
| 7097 } | |
| 7098 alcCloseDevice(dev); | |
| 7099 | |
| 7100 Sound_Quit(); | |
| 7101 | |
|
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7102 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7103 /* Remember: ALmixer_SetError/GetError calls will not work while this is gone. */ |
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7104 TError_FreeErrorPool(s_ALmixerErrorPool); |
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7105 s_ALmixerErrorPool = NULL; |
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7106 #endif |
| 0 | 7107 return; |
| 7108 } | |
| 7109 | |
|
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7110 ALboolean ALmixer_IsInitialized() |
| 0 | 7111 { |
| 7112 return ALmixer_Initialized; | |
| 7113 } | |
| 7114 | |
|
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7115 ALuint ALmixer_GetFrequency() |
| 0 | 7116 { |
| 7117 return ALmixer_Frequency_global; | |
| 7118 } | |
| 7119 | |
|
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7120 const ALmixer_version* ALmixer_GetLinkedVersion() |
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7121 { |
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7122 static ALmixer_version linked_mixver; |
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7123 ALMIXER_GET_COMPILED_VERSION(&linked_mixver); |
| 0 | 7124 return(&linked_mixver); |
| 7125 } | |
| 7126 | |
|
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7127 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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7128 |
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7129 const char* ALmixer_GetError() |
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7130 { |
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7131 const char* error_string = NULL; |
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7132 if(NULL == s_ALmixerErrorPool) |
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7133 { |
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7134 return "Error: You should not call ALmixer_GetError while ALmixer is not initialized"; |
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7135 } |
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|
7136 error_string = TError_GetLastErrorStr(s_ALmixerErrorPool); |
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7137 /* SDL returns empty strings instead of NULL */ |
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7138 if(NULL == error_string) |
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|
7139 { |
|
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7140 return ""; |
|
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7141 } |
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7142 else |
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7143 { |
|
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7144 return error_string; |
|
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7145 } |
|
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7146 } |
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7147 |
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7148 void ALmixer_SetError(const char* err_str, ...) |
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7149 { |
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7150 if(NULL == s_ALmixerErrorPool) |
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7151 { |
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7152 fprintf(stderr, "Error: You should not call ALmixer_SetError while ALmixer is not initialized\n"); |
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7153 return; |
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7154 } |
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7155 va_list argp; |
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7156 va_start(argp, err_str); |
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7157 // SDL_SetError which I'm emulating has no number parameter. |
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7158 TError_SetErrorv(s_ALmixerErrorPool, 1, err_str, argp); |
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7159 va_end(argp); |
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7160 } |
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7161 |
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7162 #endif |
| 0 | 7163 |
| 7164 | |
| 7165 | |
| 7166 | |
| 7167 #if 0 | |
|
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7168 void ALmixer_OutputAttributes() |
| 0 | 7169 { |
| 7170 ALint num_flags = 0; | |
| 7171 ALint* flags = 0; | |
| 7172 int i; | |
| 7173 ALCdevice* dev = alcGetContextsDevice( alcGetCurrentContext() ); | |
| 7174 | |
| 7175 | |
| 7176 printf("custom context\n"); | |
| 7177 | |
| 7178 alcGetIntegerv(dev, ALC_ATTRIBUTES_SIZE, | |
| 7179 sizeof num_flags, &num_flags ); | |
| 7180 | |
| 7181 printf("Number of Flags: %d\n", num_flags); | |
| 7182 | |
| 7183 if(num_flags) | |
| 7184 { | |
| 7185 flags = malloc(sizeof(num_flags) * sizeof(ALint)); | |
| 7186 | |
| 7187 alcGetIntegerv(dev, ALC_ALL_ATTRIBUTES, | |
| 7188 sizeof num_flags * sizeof(ALint), | |
| 7189 flags ); | |
| 7190 } | |
| 7191 for(i = 0; i < num_flags-1; i += 2) | |
| 7192 { | |
| 7193 printf("key 0x%x : value %d\n", | |
| 7194 flags[i], flags[i+1]); | |
| 7195 } | |
| 7196 free(flags); | |
| 7197 } | |
| 7198 #endif | |
| 7199 | |
| 7200 | |
|
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7201 void ALmixer_OutputDecoders() |
| 0 | 7202 { |
| 7203 Sound_Version sound_compile_version; | |
| 7204 Sound_Version sound_link_version; | |
| 7205 | |
| 7206 const Sound_DecoderInfo **rc = Sound_AvailableDecoders(); | |
| 7207 const Sound_DecoderInfo **i; | |
| 7208 const char **ext; | |
| 7209 FILE* stream = stdout; | |
| 7210 | |
| 7211 | |
| 7212 fprintf(stream, "SDL_sound Information:\n"); | |
| 7213 | |
| 7214 SOUND_VERSION(&sound_compile_version); | |
| 7215 fprintf(stream, "\tCompiled with SDL_sound version: %d.%d.%d\n", | |
| 7216 sound_compile_version.major, | |
| 7217 sound_compile_version.minor, | |
| 7218 sound_compile_version.patch); | |
| 7219 | |
| 7220 Sound_GetLinkedVersion(&sound_link_version); | |
| 7221 fprintf(stream, "\tRunning (linked) with SDL_sound version: %d.%d.%d\n", | |
| 7222 sound_link_version.major, | |
| 7223 sound_link_version.minor, | |
| 7224 sound_link_version.patch); | |
| 7225 | |
| 7226 fprintf(stream, "Supported sound formats:\n"); | |
| 7227 if (rc == NULL) | |
| 7228 fprintf(stream, " * Apparently, NONE!\n"); | |
| 7229 else | |
| 7230 { | |
| 7231 for (i = rc; *i != NULL; i++) | |
| 7232 { | |
| 7233 fprintf(stream, " * %s\n", (*i)->description); | |
| 7234 | |
| 7235 for (ext = (*i)->extensions; *ext != NULL; ext++) | |
| 7236 fprintf(stream, " File extension \"%s\"\n", *ext); | |
| 7237 | |
| 7238 fprintf(stream, " Written by %s.\n %s\n\n", | |
| 7239 (*i)->author, (*i)->url); | |
| 7240 } /* for */ | |
| 7241 } /* else */ | |
| 7242 | |
| 7243 fprintf(stream, "\n"); | |
| 7244 } | |
| 7245 | |
|
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7246 void ALmixer_OutputOpenALInfo() |
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7247 { |
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7248 ALmixer_version mixer_compile_version; |
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7249 const ALmixer_version * mixer_link_version=ALmixer_GetLinkedVersion(); |
| 0 | 7250 FILE* stream = stdout; |
| 7251 | |
| 7252 fprintf(stream, "OpenAL Information:\n"); | |
| 7253 fprintf(stream, "\tAL_VENDOR: %s\n", alGetString( AL_VENDOR ) ); | |
| 7254 fprintf(stream, "\tAL_VERSION: %s\n", alGetString( AL_VERSION ) ); | |
| 7255 fprintf(stream, "\tAL_RENDERER: %s\n", alGetString( AL_RENDERER ) ); | |
| 7256 fprintf(stream, "\tAL_EXTENSIONS: %s\n", alGetString( AL_EXTENSIONS ) ); | |
| 7257 | |
|
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7258 ALMIXER_GET_COMPILED_VERSION(&mixer_compile_version); |
| 0 | 7259 fprintf(stream, "\nSDL_ALmixer Information:\n"); |
| 7260 fprintf(stream, "\tCompiled with SDL_ALmixer version: %d.%d.%d\n", | |
| 7261 mixer_compile_version.major, | |
| 7262 mixer_compile_version.minor, | |
| 7263 mixer_compile_version.patch); | |
| 7264 | |
| 7265 fprintf(stream, "\tRunning (linked) with SDL_ALmixer version: %d.%d.%d\n", | |
| 7266 mixer_link_version->major, | |
| 7267 mixer_link_version->minor, | |
| 7268 mixer_link_version->patch); | |
| 7269 | |
| 7270 fprintf(stream, "\tCompile flags: "); | |
| 7271 #ifdef ENABLE_LOKI_QUEUE_FIX_HACK | |
| 7272 fprintf(stream, "ENABLE_LOKI_QUEUE_FIX_HACK "); | |
| 7273 #endif | |
| 7274 #ifdef ENABLE_ALMIXER_THREADS | |
| 7275 fprintf(stream, "ENABLE_ALMIXER_THREADS "); | |
| 7276 #endif | |
| 7277 #ifdef ENABLE_ALC_SYNC | |
| 7278 fprintf(stream, "ENABLE_ALC_SYNC "); | |
| 7279 #endif | |
| 7280 fprintf(stream, "\n"); | |
| 7281 } | |
| 7282 | |
| 7283 | |
|
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7284 ALint ALmixer_AllocateChannels(ALint numchans) |
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7285 { |
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7286 ALint retval; |
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7287 #ifdef ENABLE_ALMIXER_THREADS |
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7288 SDL_LockMutex(s_simpleLock); |
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7289 #endif |
| 0 | 7290 retval = Internal_AllocateChannels(numchans); |
|
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7291 #ifdef ENABLE_ALMIXER_THREADS |
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7292 SDL_UnlockMutex(s_simpleLock); |
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7293 #endif |
| 0 | 7294 return retval; |
| 7295 } | |
| 7296 | |
| 7297 | |
|
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7298 ALint ALmixer_ReserveChannels(ALint num) |
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7299 { |
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7300 ALint retval; |
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7301 #ifdef ENABLE_ALMIXER_THREADS |
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7302 SDL_LockMutex(s_simpleLock); |
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7303 #endif |
| 0 | 7304 retval = Internal_ReserveChannels(num); |
|
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7305 #ifdef ENABLE_ALMIXER_THREADS |
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7306 SDL_UnlockMutex(s_simpleLock); |
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7307 #endif |
| 0 | 7308 return retval; |
| 7309 } | |
| 7310 | |
| 7311 | |
| 7312 | |
| 7313 | |
|
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7314 static ALmixer_Data* DoLoad(Sound_Sample* sample, ALuint buffersize, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data) |
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7315 { |
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|
7316 ALuint bytes_decoded; |
| 0 | 7317 ALmixer_Data* ret_data; |
| 7318 ALenum error; | |
| 7319 | |
| 7320 /* Allocate memory */ | |
| 7321 ret_data = (ALmixer_Data *)malloc(sizeof(ALmixer_Data)); | |
| 7322 if (NULL == ret_data) | |
| 7323 { | |
| 7324 ALmixer_SetError("Out of memory"); | |
| 7325 return(NULL); | |
| 7326 } | |
| 7327 | |
| 7328 /* Initialize the data fields */ | |
| 7329 | |
| 7330 /* Set the Sound_Sample pointer */ | |
| 7331 ret_data->sample = sample; | |
| 7332 | |
| 7333 /* Flag the data to note that it is not in use */ | |
| 7334 ret_data->in_use = 0; | |
| 7335 | |
| 7336 /* Initialize remaining flags */ | |
| 7337 ret_data->total_time = -1; | |
| 7338 ret_data->eof = 0; | |
| 7339 | |
| 7340 /* Just initialize */ | |
| 7341 ret_data->num_buffers_in_use = 0; | |
| 7342 | |
| 7343 /* Just initialize */ | |
| 7344 ret_data->total_bytes = 0; | |
| 7345 | |
| 7346 /* Just initialize */ | |
| 7347 ret_data->loaded_bytes = 0; | |
| 7348 | |
| 7349 /* Set the max queue buffers (minimum must be 2) */ | |
| 7350 if(max_queue_buffers < 2) | |
| 7351 { | |
| 7352 max_queue_buffers = ALMIXER_DEFAULT_QUEUE_BUFFERS; | |
| 7353 } | |
| 7354 ret_data->max_queue_buffers = max_queue_buffers; | |
| 7355 /* Set up the start up buffers */ | |
| 7356 if(0 == num_startup_buffers) | |
| 7357 { | |
| 7358 num_startup_buffers = ALMIXER_DEFAULT_STARTUP_BUFFERS; | |
| 7359 } | |
| 7360 /* Make sure start up buffers is less or equal to max_queue_buffers */ | |
| 7361 if(num_startup_buffers > max_queue_buffers) | |
| 7362 { | |
| 7363 num_startup_buffers = max_queue_buffers; | |
| 7364 } | |
| 7365 ret_data->num_startup_buffers = num_startup_buffers; | |
| 7366 | |
| 7367 ret_data->buffer_map_list = NULL; | |
| 7368 ret_data->current_buffer = 0; | |
| 7369 | |
| 7370 ret_data->circular_buffer_queue = NULL; | |
| 7371 | |
| 7372 /* Now decode and load the data into a data chunk */ | |
| 7373 /* Different cases for Streamed and Predecoded | |
| 7374 * Streamed might turn into a predecoded if buffersize | |
| 7375 * is large enough */ | |
|
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|
7376 if(AL_FALSE == decode_mode_is_predecoded) |
| 0 | 7377 { |
| 7378 bytes_decoded = Sound_Decode(sample); | |
| 7379 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7380 { | |
| 7381 ALmixer_SetError(Sound_GetError()); | |
| 7382 Sound_FreeSample(sample); | |
| 7383 free(ret_data); | |
| 7384 return NULL; | |
| 7385 } | |
| 7386 | |
| 7387 /* If no data, return an error */ | |
| 7388 if(0 == bytes_decoded) | |
| 7389 { | |
| 7390 ALmixer_SetError("File has no data"); | |
| 7391 Sound_FreeSample(sample); | |
| 7392 free(ret_data); | |
| 7393 return NULL; | |
| 7394 } | |
| 7395 | |
| 7396 /* Note, currently, my Ogg conservative modifications | |
| 7397 * prevent EOF from being detected in the first read | |
| 7398 * because of the weird packet behavior of ov_read(). | |
| 7399 * The EAGAIN will get set, but not the EOF. | |
| 7400 * I don't know the best way to handle this, | |
| 7401 * so for now, Ogg's can only be explicitly | |
| 7402 * predecoded. | |
| 7403 */ | |
| 7404 | |
| 7405 /* Correction: Since we no longer actually keep the | |
| 7406 * streamed data we read here (we rewind and throw | |
| 7407 * it away, and start over on Play), it is | |
| 7408 * safe to read another chunk to see if we've hit EOF | |
| 7409 */ | |
| 7410 if(sample->flags & SOUND_SAMPLEFLAG_EAGAIN) | |
| 7411 { | |
| 7412 bytes_decoded = Sound_Decode(sample); | |
| 7413 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7414 { | |
| 7415 ALmixer_SetError(Sound_GetError()); | |
| 7416 Sound_FreeSample(sample); | |
| 7417 free(ret_data); | |
| 7418 return NULL; | |
| 7419 } | |
| 7420 } | |
| 7421 | |
| 7422 | |
| 7423 /* If we found an EOF, the entire file was | |
| 7424 * decoded, so we can treat it like one. | |
| 7425 */ | |
| 7426 | |
| 7427 if(sample->flags & SOUND_SAMPLEFLAG_EOF) | |
| 7428 { | |
|
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|
7429 /* |
| 0 | 7430 fprintf(stderr, "We got LUCKY! File is predecoded even though STREAM was requested\n"); |
|
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|
7431 */ |
| 0 | 7432 ret_data->decoded_all = 1; |
| 7433 /* Need to keep this information around for | |
| 7434 * seek and rewind abilities. | |
| 7435 */ | |
| 7436 ret_data->total_bytes = bytes_decoded; | |
| 7437 /* For now, the loaded bytes is the same as total bytes, but | |
| 7438 * this could change during a seek operation | |
| 7439 */ | |
| 7440 ret_data->loaded_bytes = bytes_decoded; | |
| 7441 | |
| 7442 /* Let's compute the total playing time | |
| 7443 * SDL_sound does not yet provide this (we're working on | |
| 7444 * that at the moment...) | |
| 7445 */ | |
| 7446 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
| 7447 | |
| 7448 /* Create one element in the buffer array for data for OpanAL */ | |
| 7449 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
| 7450 if(NULL == ret_data->buffer) | |
| 7451 { | |
| 7452 ALmixer_SetError("Out of Memory"); | |
| 7453 Sound_FreeSample(sample); | |
| 7454 free(ret_data); | |
| 7455 return NULL; | |
| 7456 } | |
| 7457 /* Clear the error code */ | |
| 7458 alGetError(); | |
| 7459 /* Now generate an OpenAL buffer using that first element */ | |
| 7460 alGenBuffers(1, ret_data->buffer); | |
| 7461 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7462 { | |
|
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7463 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7464 Sound_FreeSample(sample); |
| 7465 free(ret_data->buffer); | |
| 7466 free(ret_data); | |
| 7467 return NULL; | |
| 7468 } | |
| 7469 | |
| 7470 | |
| 7471 /* Now copy the data to the OpenAL buffer */ | |
| 7472 /* We can't just set a pointer because the API needs | |
| 7473 * its own copy to assist hardware acceleration */ | |
| 7474 alBufferData(ret_data->buffer[0], | |
| 7475 TranslateFormat(&sample->desired), | |
| 7476 sample->buffer, | |
| 7477 bytes_decoded, | |
| 7478 sample->desired.rate | |
| 7479 ); | |
| 7480 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7481 { | |
|
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7482 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 7483 Sound_FreeSample(sample); |
| 7484 alDeleteBuffers(1, ret_data->buffer); | |
| 7485 free(ret_data->buffer); | |
| 7486 free(ret_data); | |
| 7487 return NULL; | |
| 7488 } | |
| 7489 | |
| 7490 /* We should be done with the sample since it's all | |
| 7491 * predecoded. So we can free the memory */ | |
| 7492 | |
| 7493 /* Additional notes: | |
| 7494 * We need to keep data around in case Seek() is needed | |
| 7495 * or other Sound_AudioInfo is needed. | |
| 7496 * This can either be done by not deleting the sample, | |
| 7497 * or it can be done by dynamically recreating it | |
| 7498 * when we need it. | |
| 7499 */ | |
| 7500 /* Since OpenAL won't let us retrieve it | |
| 7501 * (aka dynamically), we have to keep the Sample | |
| 7502 * around because since the user requested | |
| 7503 * streamed and we offered predecoded, | |
| 7504 * we don't want to mess up the user who | |
| 7505 * was expecting seek support | |
| 7506 * So Don't Do anything | |
| 7507 */ | |
| 7508 /* | |
| 7509 if(0 == access_data) | |
| 7510 { | |
| 7511 Sound_FreeSample(sample); | |
| 7512 ret_data->sample = NULL; | |
| 7513 } | |
| 7514 */ | |
| 7515 /* Else, We keep a copy of the sample around. | |
| 7516 * so don't do anything. | |
| 7517 */ | |
| 7518 | |
| 7519 #if 0 | |
| 7520 #if defined(DISABLE_PREDECODED_SEEK) | |
| 7521 Sound_FreeSample(sample); | |
| 7522 ret_data->sample = NULL; | |
| 7523 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 7524 Sound_FreeSample(sample); | |
| 7525 ret_data->sample = NULL; | |
| 7526 #else | |
| 7527 /* We keep a copy of the sample around. | |
| 7528 * so don't do anything. | |
| 7529 */ | |
| 7530 #endif | |
| 7531 #endif | |
| 7532 /* okay we're done here */ | |
| 7533 | |
| 7534 } | |
| 7535 /* Else, we need to stream the data, so we'll | |
| 7536 * create multple buffers for queuing */ | |
| 7537 else | |
| 7538 { | |
|
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|
7539 /* |
| 0 | 7540 fprintf(stderr, "Loading streamed data (not lucky)\n"); |
|
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|
7541 */ |
| 0 | 7542 ret_data->decoded_all = 0; |
| 7543 | |
| 7544 /* This information is for predecoded. | |
| 7545 * Set to 0, since we don't know. | |
| 7546 */ | |
| 7547 ret_data->total_bytes = 0; | |
| 7548 | |
| 7549 /* Create buffers for data | |
| 7550 */ | |
| 7551 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) * max_queue_buffers); | |
| 7552 if(NULL == ret_data->buffer) | |
| 7553 { | |
| 7554 ALmixer_SetError("Out of Memory"); | |
| 7555 Sound_FreeSample(sample); | |
| 7556 free(ret_data); | |
| 7557 return NULL; | |
| 7558 } | |
| 7559 | |
| 7560 /* Clear the error code */ | |
| 7561 alGetError(); | |
| 7562 /* Now generate an OpenAL buffer using that first element */ | |
| 7563 alGenBuffers(max_queue_buffers, ret_data->buffer); | |
| 7564 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7565 { | |
|
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|
7566 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7567 Sound_FreeSample(sample); |
| 7568 free(ret_data->buffer); | |
| 7569 free(ret_data); | |
| 7570 return NULL; | |
| 7571 } | |
| 7572 | |
| 7573 /* Redesign: Okay, because of the unqueuing problems and such, | |
| 7574 * I've decided to redesign where and how queuing is handled. | |
| 7575 * Before, everything was queued up here. However, this | |
| 7576 * placed a penalty on load and made performance inconsistent | |
| 7577 * when samples had to be rewound. It did make things easier | |
| 7578 * to queue because I could let OpenAL decide which buffer | |
| 7579 * needed to be queued next. | |
| 7580 * Now, I'm going to push off the queuing to the actual | |
| 7581 * Play() command. I'm going to add some book keeping, | |
| 7582 * and allow for additional buffers to be filled at later | |
| 7583 * times. | |
| 7584 */ | |
| 7585 | |
| 7586 | |
| 7587 /* So first of all, because of I already decoded the sample | |
| 7588 * for testing, I need to decide what to do with it. | |
| 7589 * The best thing would be be to alBufferData() it. | |
| 7590 * The problem is it may conflict with the rest of | |
| 7591 * the system because everything now assumes buffers | |
| 7592 * are entirely stripped (because of the unqueing | |
| 7593 * problem). | |
| 7594 * So it looks like I have to do the crappy thing | |
| 7595 * and throw away the data, and rewind. | |
| 7596 */ | |
| 7597 | |
| 7598 if(0 == Sound_Rewind(ret_data->sample)) | |
| 7599 { | |
| 7600 ALmixer_SetError("Cannot use sample for streamed data because it must be rewindable: %s", Sound_GetError() ); | |
| 7601 Sound_FreeSample(sample); | |
| 7602 free(ret_data->buffer); | |
| 7603 free(ret_data); | |
| 7604 return NULL; | |
| 7605 } | |
| 7606 | |
| 7607 | |
| 7608 /* If the user has selected access_data, we need to | |
| 7609 * keep copies of the queuing buffers around because | |
| 7610 * OpenAL won't let us access the data. | |
| 7611 * Allocate the memory for the buffers here | |
| 7612 * and initialize the albuffer-index map | |
| 7613 */ | |
| 7614 if(access_data) | |
| 7615 { | |
|
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|
7616 ALuint j; |
| 0 | 7617 /* Create buffers for data access |
| 7618 * Should be the same number as the number of queue buffers | |
| 7619 */ | |
| 1 | 7620 ret_data->buffer_map_list = (ALmixer_Buffer_Map*)malloc( sizeof(ALmixer_Buffer_Map) * max_queue_buffers); |
| 0 | 7621 if(NULL == ret_data->buffer_map_list) |
| 7622 { | |
| 7623 ALmixer_SetError("Out of Memory"); | |
| 7624 Sound_FreeSample(sample); | |
| 7625 free(ret_data->buffer); | |
| 7626 free(ret_data); | |
| 7627 return NULL; | |
| 7628 } | |
| 7629 | |
| 7630 ret_data->circular_buffer_queue = CircularQueueUnsignedInt_CreateQueue(max_queue_buffers); | |
| 7631 if(NULL == ret_data->circular_buffer_queue) | |
| 7632 { | |
| 7633 ALmixer_SetError("Out of Memory"); | |
| 7634 free(ret_data->buffer_map_list); | |
| 7635 Sound_FreeSample(sample); | |
| 7636 free(ret_data->buffer); | |
| 7637 free(ret_data); | |
| 7638 return NULL; | |
| 7639 } | |
| 7640 | |
| 7641 | |
| 7642 for(j=0; j<max_queue_buffers; j++) | |
| 7643 { | |
| 7644 ret_data->buffer_map_list[j].albuffer = ret_data->buffer[j]; | |
| 7645 ret_data->buffer_map_list[j].index = j; | |
| 7646 ret_data->buffer_map_list[j].num_bytes = 0; | |
|
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|
7647 ret_data->buffer_map_list[j].data = (ALbyte*)malloc( sizeof(ALbyte) * buffersize); |
| 0 | 7648 if(NULL == ret_data->buffer_map_list[j].data) |
| 7649 { | |
| 7650 ALmixer_SetError("Out of Memory"); | |
| 7651 break; | |
| 7652 } | |
| 7653 } | |
| 7654 /* If an error happened, we have to clean up the memory */ | |
| 7655 if(j < max_queue_buffers) | |
| 7656 { | |
| 7657 fprintf(stderr, "################## Buffer allocation failed\n"); | |
| 7658 for( ; j>=0; j--) | |
| 7659 { | |
| 7660 free(ret_data->buffer_map_list[j].data); | |
| 7661 } | |
| 7662 free(ret_data->buffer_map_list); | |
| 7663 CircularQueueUnsignedInt_FreeQueue(ret_data->circular_buffer_queue); | |
| 7664 Sound_FreeSample(sample); | |
| 7665 free(ret_data->buffer); | |
| 7666 free(ret_data); | |
| 7667 return NULL; | |
| 7668 } | |
| 7669 | |
| 7670 /* The Buffer_Map_List must be sorted by albuffer for binary searches | |
| 7671 */ | |
| 1 | 7672 qsort(ret_data->buffer_map_list, max_queue_buffers, sizeof(ALmixer_Buffer_Map), Compare_Buffer_Map); |
| 0 | 7673 } /* End if access_data==true */ |
| 7674 | |
| 7675 | |
| 7676 } /* End of do stream */ | |
| 7677 } /* end of DECODE_STREAM */ | |
| 7678 /* User requested decode all (easy, nothing to figure out) */ | |
|
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|
7679 else if(AL_TRUE == decode_mode_is_predecoded) |
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|
7680 { |
|
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|
7681 #ifdef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION |
|
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|
7682 /* SDL_sound (behind the scenes) seems to loop on buffer_size chunks |
|
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|
7683 * until the buffer is filled. It seems like we can |
|
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|
7684 * do much better and precompute the size of the buffer |
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|
7685 * so looping isn't needed. |
|
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|
7686 * WARNING: Due to the way SDL_sound is currently implemented, |
|
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|
7687 * this may waste a lot of memory up front. |
|
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|
7688 * SDL_sound seems to pre-create a buffer of the requested size, |
|
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|
7689 * but on DecodeAll, an entirely new buffer is created and |
|
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|
7690 * everything is memcpy'd into the new buffer in read chunks |
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|
7691 * of the buffer_size. This means we need roughly twice the memory |
|
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|
7692 * to load a file. |
|
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|
7693 */ |
|
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|
7694 ALint sound_duration = Sound_GetDuration(sample); |
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|
7695 if(sound_duration > 0) |
|
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|
7696 { |
|
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|
7697 size_t total_bytes = Compute_Total_Bytes_With_Frame_Padding(&sample->desired, (ALuint)sound_duration); |
|
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|
7698 int buffer_resize_succeeded = Sound_SetBufferSize(sample, total_bytes); |
|
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|
7699 if(0 == buffer_resize_succeeded) |
|
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|
7700 { |
|
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|
7701 ALmixer_SetError(Sound_GetError()); |
|
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|
7702 Sound_FreeSample(sample); |
|
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|
7703 free(ret_data); |
|
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|
7704 return NULL; |
|
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|
7705 } |
|
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|
7706 } |
|
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|
7707 #endif /* ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION */ |
| 0 | 7708 bytes_decoded = Sound_DecodeAll(sample); |
| 7709 if(sample->flags & SOUND_SAMPLEFLAG_ERROR) | |
| 7710 { | |
| 7711 ALmixer_SetError(Sound_GetError()); | |
| 7712 Sound_FreeSample(sample); | |
| 7713 free(ret_data); | |
| 7714 return NULL; | |
| 7715 } | |
| 7716 | |
| 7717 /* If no data, return an error */ | |
| 7718 if(0 == bytes_decoded) | |
| 7719 { | |
| 7720 ALmixer_SetError("File has no data"); | |
| 7721 Sound_FreeSample(sample); | |
| 7722 free(ret_data); | |
| 7723 return NULL; | |
| 7724 } | |
| 7725 | |
| 7726 | |
| 7727 ret_data->decoded_all = 1; | |
| 7728 /* Need to keep this information around for | |
| 7729 * seek and rewind abilities. | |
| 7730 */ | |
| 7731 ret_data->total_bytes = bytes_decoded; | |
| 7732 /* For now, the loaded bytes is the same as total bytes, but | |
| 7733 * this could change during a seek operation | |
| 7734 */ | |
| 7735 ret_data->loaded_bytes = bytes_decoded; | |
| 7736 | |
| 7737 /* Let's compute the total playing time | |
| 7738 * SDL_sound does not yet provide this (we're working on | |
| 7739 * that at the moment...) | |
| 7740 */ | |
| 7741 ret_data->total_time = Compute_Total_Time(&sample->desired, bytes_decoded); | |
| 7742 | |
| 7743 /* Create one element in the buffer array for data for OpanAL */ | |
| 7744 ret_data->buffer = (ALuint*)malloc( sizeof(ALuint) ); | |
| 7745 if(NULL == ret_data->buffer) | |
| 7746 { | |
| 7747 ALmixer_SetError("Out of Memory"); | |
| 7748 Sound_FreeSample(sample); | |
| 7749 free(ret_data); | |
| 7750 return NULL; | |
| 7751 } | |
| 7752 /* Clear the error code */ | |
| 7753 alGetError(); | |
| 7754 /* Now generate an OpenAL buffer using that first element */ | |
| 7755 alGenBuffers(1, ret_data->buffer); | |
| 7756 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7757 { | |
|
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|
7758 ALmixer_SetError("alGenBuffers failed: %s\n", alGetString(error)); |
| 0 | 7759 Sound_FreeSample(sample); |
| 7760 free(ret_data->buffer); | |
| 7761 free(ret_data); | |
| 7762 return NULL; | |
| 7763 } | |
|
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|
7764 /* |
| 0 | 7765 fprintf(stderr, "Actual rate=%d, desired=%d\n", sample->actual.rate, sample->desired.rate); |
|
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|
7766 */ |
| 0 | 7767 /* Now copy the data to the OpenAL buffer */ |
| 7768 /* We can't just set a pointer because the API needs | |
| 7769 * its own copy to assist hardware acceleration */ | |
| 7770 alBufferData(ret_data->buffer[0], | |
| 7771 TranslateFormat(&sample->desired), | |
| 7772 sample->buffer, | |
| 7773 bytes_decoded, | |
| 7774 sample->desired.rate | |
| 7775 ); | |
| 7776 if( (error = alGetError()) != AL_NO_ERROR) | |
| 7777 { | |
|
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|
7778 ALmixer_SetError("alBufferData failed: %s\n", alGetString(error)); |
| 0 | 7779 Sound_FreeSample(sample); |
| 7780 alDeleteBuffers(1, ret_data->buffer); | |
| 7781 free(ret_data->buffer); | |
| 7782 free(ret_data); | |
| 7783 return NULL; | |
| 7784 } | |
| 7785 | |
| 7786 /* We should be done with the sample since it's all | |
| 7787 * predecoded. So we can free the memory */ | |
| 7788 /* Need to keep around because Seek() needs it */ | |
| 7789 | |
| 7790 /* Additional notes: | |
| 7791 * We need to keep data around in case Seek() is needed | |
| 7792 * or other Sound_AudioInfo is needed. | |
| 7793 * This can either be done by not deleting the sample, | |
| 7794 * or it can be done by dynamically recreating it | |
| 7795 * when we need it. | |
| 7796 * Update: I think now it's up to the user by passing the | |
| 7797 * access_data flag. If they set the flag, then they get | |
| 7798 * data callbacks and seek support. If not, then they can | |
| 7799 * get all that stuff at the expense of keeping extra memory | |
| 7800 * around. | |
| 7801 */ | |
| 7802 if(0 == access_data) | |
| 7803 { | |
| 7804 Sound_FreeSample(sample); | |
| 7805 ret_data->sample = NULL; | |
| 7806 } | |
| 7807 | |
| 7808 /* Else, We keep a copy of the sample around. | |
| 7809 * so don't do anything. | |
| 7810 */ | |
| 7811 #if 0 | |
| 7812 #if defined(DISABLE_PREDECODED_SEEK) | |
| 7813 Sound_FreeSample(sample); | |
| 7814 ret_data->sample = NULL; | |
| 7815 #elif !defined(DISABLE_SEEK_MEMORY_OPTIMIZATION) | |
| 7816 Sound_FreeSample(sample); | |
| 7817 ret_data->sample = NULL; | |
| 7818 #else | |
| 7819 /* We keep a copy of the sample around. | |
| 7820 * so don't do anything. | |
| 7821 */ | |
| 7822 #endif | |
| 7823 #endif | |
| 7824 | |
| 7825 /* okay we're done here */ | |
| 7826 } | |
| 7827 else | |
| 7828 { | |
| 7829 /* Shouldn't get here */ | |
| 7830 ALmixer_SetError("Unknown decode mode"); | |
| 7831 Sound_FreeSample(sample); | |
| 7832 free(ret_data); | |
| 7833 return NULL; | |
| 7834 } | |
| 7835 | |
| 7836 return ret_data; | |
| 7837 } | |
| 7838 | |
| 7839 | |
| 7840 /* This will load a sample for us. Most of the uglyness is | |
| 7841 * error checking and the fact that streamed/predecoded files | |
| 7842 * must be treated differently. | |
| 7843 * I don't like the AudioInfo parameter. I removed it once, | |
| 7844 * but the system will fail on RAW samples because the user | |
| 7845 * must specify it, so I had to bring it back. | |
| 7846 * Remember I must close the rwops if there is an error before NewSample() | |
| 7847 */ | |
|
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|
7848 ALmixer_Data* ALmixer_LoadSample_RW(ALmixer_RWops* rwops, const char* fileext, ALuint buffersize, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data) |
| 0 | 7849 { |
| 7850 Sound_Sample* sample = NULL; | |
| 7851 Sound_AudioInfo target; | |
| 7852 | |
| 7853 /* Initialize target values to defaults | |
| 7854 * 0 tells SDL_sound to use the "actual" values | |
| 7855 */ | |
| 7856 target.channels = 0; | |
| 7857 target.rate = 0; | |
| 7858 #if 0 | |
| 7859 /* This requires my new additions to SDL_sound. It will | |
| 7860 * convert the sample to the proper endian order. | |
| 7861 * If the actual is 8-bit, it will do unsigned, if | |
| 7862 * the actual is 16-bit, it will do signed. | |
| 7863 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
| 7864 * in this way. | |
| 7865 */ | |
| 7866 target.format = AUDIO_U8S16SYS; | |
| 7867 #else | |
| 7868 target.format = AUDIO_S16SYS; | |
| 7869 #endif | |
| 7870 | |
| 7871 /* Set a default buffersize if needed */ | |
| 7872 if(0 == buffersize) | |
| 7873 { | |
| 7874 buffersize = ALMIXER_DEFAULT_BUFFERSIZE; | |
| 7875 } | |
| 7876 | |
| 7877 sample = Sound_NewSample(rwops, fileext, &target, buffersize); | |
| 7878 if(NULL == sample) | |
| 7879 { | |
| 7880 ALmixer_SetError(Sound_GetError()); | |
| 7881 return NULL; | |
| 7882 } | |
| 7883 | |
| 1 | 7884 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 7885 } |
| 7886 | |
| 7887 | |
| 7888 | |
| 7889 /* This will load a sample for us from | |
| 7890 * a file (instead of RWops). Most of the uglyness is | |
| 7891 * error checking and the fact that streamed/predecoded files | |
| 7892 * must be treated differently. | |
| 7893 */ | |
|
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|
7894 ALmixer_Data* ALmixer_LoadSample(const char* filename, ALuint buffersize, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data) |
| 0 | 7895 { |
| 7896 Sound_Sample* sample = NULL; | |
| 7897 Sound_AudioInfo target; | |
| 7898 | |
| 7899 /* Initialize target values to defaults | |
| 7900 * 0 tells SDL_sound to use the "actual" values | |
| 7901 */ | |
| 7902 target.channels = 0; | |
| 7903 target.rate = 0; | |
| 7904 | |
| 7905 #if 0 | |
| 7906 /* This requires my new additions to SDL_sound. It will | |
| 7907 * convert the sample to the proper endian order. | |
| 7908 * If the actual is 8-bit, it will do unsigned, if | |
| 7909 * the actual is 16-bit, it will do signed. | |
| 7910 * I'm told by Ryan Gordon that OpenAL prefers the signedness | |
| 7911 * in this way. | |
| 7912 */ | |
| 7913 target.format = AUDIO_U8S16SYS; | |
| 7914 #else | |
| 7915 target.format = AUDIO_S16SYS; | |
| 7916 #endif | |
| 7917 | |
| 7918 #if 0 | |
| 7919 /* Okay, here's a messy hack. The problem is that we need | |
| 7920 * to convert the sample to have the correct bitdepth, | |
| 7921 * endian order, and signedness values. | |
| 7922 * The bit depth is 8 or 16. | |
| 7923 * The endian order is the native order of the system. | |
| 7924 * The signedness depends on what the original value | |
| 7925 * of the sample. Unfortunately, we can't specify these | |
| 7926 * values until we after we already know what the original | |
| 7927 * values were for bitdepth and signedness. | |
| 7928 * So we must open the file once to get the values, | |
| 7929 * then close it, and then reopen it with the | |
| 7930 * correct desired target values. | |
| 7931 * I tried changing the sample->desired field after | |
| 7932 * the NewSample call, but it had no effect, so | |
| 7933 * it looks like it must be set on open. | |
| 7934 */ | |
| 7935 /* Pick a small buffersize for the first open to not | |
| 7936 * waste much time allocating memory */ | |
| 7937 sample = Sound_NewSampleFromFile(filename, NULL, 512); | |
| 7938 if(NULL == sample) | |
| 7939 { | |
| 7940 ALmixer_SetError(Sound_GetError()); | |
| 7941 return NULL; | |
| 7942 } | |
| 7943 | |
| 7944 bit_depth = GetBitDepth(sample->actual.format); | |
| 7945 signedness_value = GetSignednessValue(sample->actual.format); | |
| 7946 if(8 == bit_depth) | |
| 7947 { | |
| 7948 /* If 8 bit, then we don't have to worry about | |
| 7949 * endian issues. We can just use the actual format | |
| 7950 * value and it should do the right thing | |
| 7951 */ | |
| 7952 target.format = sample->actual.format; | |
| 7953 } | |
| 7954 else | |
| 7955 { | |
| 7956 /* We'll assume it's 16-bit, and if it's not | |
| 7957 * hopefully SDL_sound will return an error, | |
| 7958 * or let us convert to 16-bit | |
| 7959 */ | |
| 7960 /* Now we need to get the correct signedness */ | |
| 7961 if(ALMIXER_UNSIGNED_VALUE == signedness_value) | |
| 7962 { | |
| 7963 /* Set to Unsigned 16-bit, system endian order */ | |
| 7964 target.format = AUDIO_U16SYS; | |
| 7965 } | |
| 7966 else | |
| 7967 { | |
| 7968 /* Again, we'll assume it's Signed 16-bit system order | |
| 7969 * or force the conversion and hope it works out | |
| 7970 */ | |
| 7971 target.format = AUDIO_S16SYS; | |
| 7972 } | |
| 7973 } | |
| 7974 | |
| 7975 /* Now we have the correct info. We need to close and reopen */ | |
| 7976 Sound_FreeSample(sample); | |
| 7977 #endif | |
| 7978 | |
| 7979 sample = Sound_NewSampleFromFile(filename, &target, buffersize); | |
| 7980 if(NULL == sample) | |
| 7981 { | |
| 7982 ALmixer_SetError(Sound_GetError()); | |
| 7983 return NULL; | |
| 7984 } | |
| 7985 | |
|
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|
7986 /* |
| 0 | 7987 fprintf(stderr, "Correction test: Actual rate=%d, desired=%d, actual format=%d, desired format=%d\n", sample->actual.rate, sample->desired.rate, sample->actual.format, sample->desired.format); |
|
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|
7988 */ |
| 1 | 7989 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 7990 } |
| 7991 | |
| 7992 | |
| 7993 /* This is a back door for RAW samples or if you need the | |
| 7994 * AudioInfo field. Use at your own risk. | |
| 7995 */ | |
|
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|
7996 ALmixer_Data* ALmixer_LoadSample_RAW_RW(ALmixer_RWops* rwops, const char* fileext, ALmixer_AudioInfo* desired, ALuint buffersize, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data) |
| 0 | 7997 { |
| 7998 Sound_Sample* sample = NULL; | |
| 1 | 7999 Sound_AudioInfo sound_desired; |
| 8000 /* Rather than copying the data from struct to struct, I could just | |
| 8001 * cast the thing since the structs are meant to be identical. | |
| 8002 * But if SDL_sound changes it's implementation, bad things | |
| 8003 * will probably happen. (Or if I change my implementation and | |
| 8004 * forget about the cast, same bad scenario.) Since this is a load | |
| 8005 * function, performance of this is negligible. | |
| 8006 */ | |
| 8007 if(NULL == desired) | |
| 8008 { | |
| 8009 sample = Sound_NewSample(rwops, fileext, NULL, buffersize); | |
| 8010 } | |
| 8011 else | |
| 8012 { | |
| 8013 sound_desired.format = desired->format; | |
| 8014 sound_desired.channels = desired->channels; | |
| 8015 sound_desired.rate = desired->rate; | |
| 8016 sample = Sound_NewSample(rwops, fileext, &sound_desired, buffersize); | |
| 8017 } | |
| 0 | 8018 if(NULL == sample) |
| 8019 { | |
| 8020 ALmixer_SetError(Sound_GetError()); | |
| 8021 return NULL; | |
| 8022 } | |
| 1 | 8023 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 8024 } |
| 8025 | |
| 8026 | |
| 8027 | |
| 8028 | |
| 8029 /* This is a back door for RAW samples or if you need the | |
| 8030 * AudioInfo field. Use at your own risk. | |
| 8031 */ | |
|
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|
8032 ALmixer_Data* ALmixer_LoadSample_RAW(const char* filename, ALmixer_AudioInfo* desired, ALuint buffersize, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data) |
| 0 | 8033 { |
| 8034 Sound_Sample* sample = NULL; | |
| 1 | 8035 Sound_AudioInfo sound_desired; |
| 8036 /* Rather than copying the data from struct to struct, I could just | |
| 8037 * cast the thing since the structs are meant to be identical. | |
| 8038 * But if SDL_sound changes it's implementation, bad things | |
| 8039 * will probably happen. (Or if I change my implementation and | |
| 8040 * forget about the cast, same bad scenario.) Since this is a load | |
| 8041 * function, performance of this is negligible. | |
| 8042 */ | |
| 8043 if(NULL == desired) | |
| 8044 { | |
| 8045 sample = Sound_NewSampleFromFile(filename, NULL, buffersize); | |
| 8046 } | |
| 8047 else | |
| 8048 { | |
| 8049 sound_desired.format = desired->format; | |
| 8050 sound_desired.channels = desired->channels; | |
| 8051 sound_desired.rate = desired->rate; | |
| 8052 sample = Sound_NewSampleFromFile(filename, &sound_desired, buffersize); | |
| 8053 } | |
| 8054 | |
| 0 | 8055 if(NULL == sample) |
| 8056 { | |
| 8057 ALmixer_SetError(Sound_GetError()); | |
| 8058 return NULL; | |
| 8059 } | |
| 1 | 8060 return( DoLoad(sample, buffersize, decode_mode_is_predecoded, max_queue_buffers, num_startup_buffers, access_data)); |
| 0 | 8061 } |
| 8062 | |
| 8063 | |
| 8064 | |
| 8065 | |
| 8066 void ALmixer_FreeData(ALmixer_Data* data) | |
| 8067 { | |
| 8068 ALenum error; | |
| 8069 if(NULL == data) | |
| 8070 { | |
| 8071 return; | |
| 8072 } | |
| 8073 | |
| 8074 if(data->decoded_all) | |
| 8075 { | |
| 8076 /* If access_data was enabled, then the Sound_Sample* | |
| 8077 * still exists. We need to free it | |
| 8078 */ | |
| 8079 if(data->sample != NULL) | |
| 8080 { | |
| 8081 Sound_FreeSample(data->sample); | |
| 8082 } | |
| 8083 alDeleteBuffers(1, data->buffer); | |
| 8084 if((error = alGetError()) != AL_NO_ERROR) | |
| 8085 { | |
| 8086 fprintf(stderr, "70Testing error: %s\n", | |
|
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|
8087 alGetString(error)); |
| 0 | 8088 } |
| 8089 | |
| 8090 } | |
| 8091 else | |
| 8092 { | |
|
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|
8093 ALuint i; |
| 0 | 8094 |
| 8095 /* Delete buffer copies if access_data was enabled */ | |
| 8096 if(data->buffer_map_list != NULL) | |
| 8097 { | |
| 8098 for(i=0; i<data->max_queue_buffers; i++) | |
| 8099 { | |
| 8100 free(data->buffer_map_list[i].data); | |
| 8101 } | |
| 8102 free(data->buffer_map_list); | |
| 8103 } | |
| 8104 if(data->circular_buffer_queue != NULL) | |
| 8105 { | |
| 8106 CircularQueueUnsignedInt_FreeQueue(data->circular_buffer_queue); | |
| 8107 } | |
| 8108 | |
| 8109 Sound_FreeSample(data->sample); | |
| 8110 alDeleteBuffers(data->max_queue_buffers, data->buffer); | |
| 8111 if((error = alGetError()) != AL_NO_ERROR) | |
| 8112 { | |
| 8113 fprintf(stderr, "71Testing error: %s\n", | |
|
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|
8114 alGetString(error)); |
| 0 | 8115 } |
| 8116 } | |
| 8117 free(data->buffer); | |
| 8118 free(data); | |
| 8119 } | |
| 8120 | |
|
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|
8121 ALint ALmixer_GetTotalTime(ALmixer_Data* data) |
|
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8122 { |
|
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|
8123 if(NULL == data) |
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8124 { |
|
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8125 return -1; |
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|
8126 } |
| 0 | 8127 return data->total_time; |
| 8128 } | |
| 8129 | |
| 8130 /* This function will look up the source for the corresponding channel */ | |
| 8131 /* Must return 0 on error instead of -1 because of unsigned int */ | |
|
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|
8132 ALuint ALmixer_GetSource(ALint channel) |
| 0 | 8133 { |
| 8134 ALuint retval; | |
|
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8135 #ifdef ENABLE_ALMIXER_THREADS |
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|
8136 SDL_LockMutex(s_simpleLock); |
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8137 #endif |
| 0 | 8138 retval = Internal_GetSource(channel); |
|
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8139 #ifdef ENABLE_ALMIXER_THREADS |
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8140 SDL_UnlockMutex(s_simpleLock); |
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|
8141 #endif |
| 0 | 8142 return retval; |
| 8143 } | |
| 8144 | |
| 8145 /* This function will look up the channel for the corresponding source */ | |
|
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|
8146 ALint ALmixer_GetChannel(ALuint source) |
|
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8147 { |
|
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|
8148 ALint retval; |
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8149 #ifdef ENABLE_ALMIXER_THREADS |
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8150 SDL_LockMutex(s_simpleLock); |
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|
8151 #endif |
| 0 | 8152 retval = Internal_GetChannel(source); |
|
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8153 #ifdef ENABLE_ALMIXER_THREADS |
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8154 SDL_UnlockMutex(s_simpleLock); |
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8155 #endif |
| 0 | 8156 return retval; |
| 8157 } | |
| 8158 | |
|
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|
8159 ALint ALmixer_FindFreeChannel(ALint start_channel) |
|
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|
8160 { |
|
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|
8161 ALint retval; |
|
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|
8162 #ifdef ENABLE_ALMIXER_THREADS |
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|
8163 SDL_LockMutex(s_simpleLock); |
|
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|
8164 #endif |
| 0 | 8165 retval = Internal_FindFreeChannel(start_channel); |
|
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8166 #ifdef ENABLE_ALMIXER_THREADS |
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|
8167 SDL_UnlockMutex(s_simpleLock); |
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|
8168 #endif |
| 0 | 8169 return retval; |
| 8170 } | |
| 8171 | |
| 8172 | |
| 8173 | |
| 8174 /* API update function. | |
| 8175 * It should return the number of buffers that were | |
| 8176 * queued during the call. The value might be | |
| 8177 * used to guage how long you might wait to | |
| 8178 * call the next update loop in case you are worried | |
| 8179 * about preserving CPU cycles. The idea is that | |
| 8180 * when a buffer is queued, there was probably some | |
| 8181 * CPU intensive looping which took awhile. | |
| 8182 * It's mainly provided as a convenience. | |
|
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8183 * Timing the call with ALmixer_GetTicks() would produce |
| 0 | 8184 * more accurate information. |
| 8185 * Returns a negative value if there was an error, | |
| 8186 * the value being the number of errors. | |
| 8187 */ | |
|
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8188 ALint ALmixer_Update() |
| 0 | 8189 { |
| 8190 #ifdef ENABLE_ALMIXER_THREADS | |
| 8191 /* The thread will handle all updates by itself. | |
| 8192 * Don't allow the user to explicitly call update. | |
| 8193 */ | |
| 8194 return 0; | |
| 8195 #else | |
| 8196 return( Update_ALmixer(NULL) ); | |
| 8197 #endif | |
| 8198 } | |
| 8199 | |
| 8200 | |
| 8201 | |
|
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|
8202 void ALmixer_SetPlaybackFinishedCallback(void (*playback_finished_callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data), void* user_data) |
|
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8203 { |
|
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8204 #ifdef ENABLE_ALMIXER_THREADS |
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8205 SDL_LockMutex(s_simpleLock); |
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8206 #endif |
|
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|
8207 Channel_Done_Callback = playback_finished_callback; |
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|
8208 Channel_Done_Callback_Userdata = user_data; |
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8209 #ifdef ENABLE_ALMIXER_THREADS |
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|
8210 SDL_UnlockMutex(s_simpleLock); |
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|
8211 #endif |
|
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|
8212 } |
|
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|
8213 |
|
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8214 |
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|
8215 void ALmixer_SetPlaybackDataCallback(void (*playback_data_callback)(ALint which_chan, ALuint al_source, ALbyte* data, ALuint num_bytes, ALuint frequency, ALubyte channels, ALubyte bit_depth, ALboolean is_unsigned, ALboolean decode_mode_is_predecoded, ALuint length_in_msec, void* user_data), void* user_data) |
|
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|
8216 { |
|
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|
8217 #ifdef ENABLE_ALMIXER_THREADS |
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|
8218 SDL_LockMutex(s_simpleLock); |
|
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|
8219 #endif |
|
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|
8220 Channel_Data_Callback = playback_data_callback; |
| 1 | 8221 Channel_Data_Callback_Userdata = user_data; |
|
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8222 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8223 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8224 #endif |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8225 } |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8226 |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8227 |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8228 |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8229 |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8230 |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8231 ALint ALmixer_PlayChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALint ticks) |
|
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8232 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8233 ALint retval; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8234 #ifdef ENABLE_ALMIXER_THREADS |
|
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parents:
1
diff
changeset
|
8235 SDL_LockMutex(s_simpleLock); |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8236 #endif |
| 0 | 8237 retval = Internal_PlayChannelTimed(channel, data, loops, ticks); |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8238 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8239 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8240 #endif |
| 0 | 8241 return retval; |
| 8242 } | |
| 8243 | |
| 8244 | |
| 8245 /* In case the user wants to specify a source instead of a channel, | |
| 8246 * they may use this function. This function will look up the | |
| 8247 * source-to-channel map, and convert the call into a | |
| 8248 * PlayChannelTimed() function call. | |
| 8249 * Returns the channel it's being played on. | |
| 8250 * Note: If you are prefer this method, then you need to be careful | |
| 8251 * about using PlayChannel, particularly if you request the | |
| 8252 * first available channels because source and channels have | |
| 8253 * a one-to-one mapping in this API. It is quite easy for | |
| 8254 * a channel/source to already be in use because of this. | |
| 8255 * In this event, an error message will be returned to you. | |
| 8256 */ | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8257 ALuint ALmixer_PlaySourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALint ticks) |
| 0 | 8258 { |
| 8259 ALuint retval; | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8260 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8261 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8262 #endif |
| 0 | 8263 retval = Internal_PlaySourceTimed(source, data, loops, ticks); |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8264 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8265 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8266 #endif |
| 0 | 8267 return retval; |
| 8268 } | |
| 8269 | |
| 8270 | |
| 8271 /* Will return the number of channels halted | |
| 8272 * or 0 for error | |
| 8273 */ | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8274 ALint ALmixer_HaltChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8275 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8276 ALint retval; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8277 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8278 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8279 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8280 retval = Internal_HaltChannel(channel, AL_FALSE); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8281 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8282 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8283 #endif |
| 0 | 8284 return retval; |
| 8285 } | |
| 8286 | |
| 8287 /* Will return the number of channels halted | |
| 8288 * or 0 for error | |
| 8289 */ | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8290 ALint ALmixer_HaltSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8291 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8292 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8293 #ifdef ENABLE_ALMIXER_THREADS |
|
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8294 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8295 #endif |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8296 retval = Internal_HaltSource(source, AL_FALSE); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8297 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8298 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8299 #endif |
| 0 | 8300 return retval; |
| 8301 } | |
| 8302 | |
| 8303 | |
| 8304 /* This will rewind the SDL_Sound sample for streamed | |
| 8305 * samples and start buffering up the data for the next | |
| 8306 * playback. This may require samples to be halted | |
| 8307 */ | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8308 ALint ALmixer_RewindData(ALmixer_Data* data) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8309 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8310 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8311 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8312 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8313 #endif |
| 0 | 8314 retval = Internal_RewindData(data); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8315 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8316 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8317 #endif |
| 0 | 8318 return retval; |
| 8319 } | |
| 8320 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8321 ALint ALmixer_RewindChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8322 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8323 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8324 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8325 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8326 #endif |
| 0 | 8327 retval = Internal_RewindChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8328 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8329 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8330 #endif |
| 0 | 8331 return retval; |
| 8332 } | |
| 8333 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8334 ALint ALmixer_RewindSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8335 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8336 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8337 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8338 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8339 #endif |
| 0 | 8340 retval = Internal_RewindSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8341 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8342 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8343 #endif |
| 0 | 8344 return retval; |
| 8345 } | |
| 8346 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8347 ALint ALmixer_PauseChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8348 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8349 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8350 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8351 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8352 #endif |
| 0 | 8353 retval = Internal_PauseChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8354 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8355 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8356 #endif |
| 0 | 8357 return retval; |
| 8358 } | |
| 8359 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8360 ALint ALmixer_PauseSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8361 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8362 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8363 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8364 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8365 #endif |
| 0 | 8366 retval = Internal_PauseSource(source); |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8367 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8368 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8369 #endif |
| 0 | 8370 return retval; |
| 8371 } | |
| 8372 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8373 ALint ALmixer_ResumeChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8374 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8375 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8376 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8377 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8378 #endif |
| 0 | 8379 retval = Internal_ResumeChannel(channel); |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
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1
diff
changeset
|
8380 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8381 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8382 #endif |
| 0 | 8383 return retval; |
| 8384 } | |
| 8385 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8386 ALint ALmixer_ResumeSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8387 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8388 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8389 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8390 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8391 #endif |
| 0 | 8392 retval = Internal_ResumeSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8393 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8394 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8395 #endif |
| 0 | 8396 return retval; |
| 8397 } | |
| 8398 | |
| 8399 /* Might consider setting eof to 0 as a "feature" | |
| 8400 * This will allow seek to end to stay there because | |
| 8401 * Play automatically rewinds if at the end */ | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8402 ALint ALmixer_SeekData(ALmixer_Data* data, ALuint msec) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8403 { |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8404 ALint retval; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8405 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8406 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8407 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8408 retval = Internal_SeekData(data, msec); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8409 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8410 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8411 #endif |
| 0 | 8412 return retval; |
| 8413 } | |
| 8414 | |
|
2
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8415 ALint ALmixer_FadeInChannelTimed(ALint channel, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8416 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8417 ALint retval; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8418 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8419 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8420 #endif |
| 0 | 8421 retval = Internal_FadeInChannelTimed(channel, data, loops, fade_ticks, expire_ticks); |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8422 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8423 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8424 #endif |
| 0 | 8425 return retval; |
| 8426 } | |
| 8427 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8428 ALuint ALmixer_FadeInSourceTimed(ALuint source, ALmixer_Data* data, ALint loops, ALuint fade_ticks, ALint expire_ticks) |
| 0 | 8429 { |
| 8430 ALuint retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8431 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8432 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8433 #endif |
| 0 | 8434 retval = Internal_FadeInSourceTimed(source, data, loops, fade_ticks, expire_ticks); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8435 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8436 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8437 #endif |
| 0 | 8438 return retval; |
| 8439 } | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8440 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8441 ALint ALmixer_FadeOutChannel(ALint channel, ALuint ticks) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8442 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8443 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8444 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8445 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8446 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8447 retval = Internal_FadeOutChannel(channel, ticks); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8448 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8449 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8450 #endif |
| 0 | 8451 return retval; |
| 8452 } | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8453 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8454 ALint ALmixer_FadeOutSource(ALuint source, ALuint ticks) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8455 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8456 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8457 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8458 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8459 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8460 retval = Internal_FadeOutSource(source, ticks); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8461 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8462 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8463 #endif |
| 0 | 8464 return retval; |
| 8465 } | |
| 8466 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8467 ALint ALmixer_FadeChannel(ALint channel, ALuint ticks, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8468 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8469 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8470 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8471 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8472 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8473 retval = Internal_FadeChannel(channel, ticks, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8474 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8475 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8476 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8477 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8478 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8479 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8480 ALint ALmixer_FadeSource(ALuint source, ALuint ticks, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8481 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8482 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8483 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8484 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8485 #endif |
| 0 | 8486 retval = Internal_FadeSource(source, ticks, volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8487 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8488 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8489 #endif |
| 0 | 8490 return retval; |
| 8491 } | |
| 8492 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8493 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8494 ALboolean ALmixer_SetVolumeChannel(ALint channel, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8495 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8496 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8497 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8498 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8499 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8500 retval = Internal_SetVolumeChannel(channel, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8501 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8502 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8503 #endif |
| 0 | 8504 return retval; |
| 8505 } | |
| 8506 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8507 ALboolean ALmixer_SetVolumeSource(ALuint source, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8508 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8509 ALboolean retval; |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8510 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8511 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8512 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8513 retval = Internal_SetVolumeSource(source, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8514 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8515 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8516 #endif |
| 0 | 8517 return retval; |
| 8518 } | |
| 8519 | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8520 ALfloat ALmixer_GetVolumeChannel(ALint channel) |
| 0 | 8521 { |
| 8522 ALfloat retval; | |
|
2
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8523 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8524 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8525 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8526 retval = Internal_GetVolumeChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8527 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8528 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8529 #endif |
|
279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8530 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8531 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8532 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8533 ALfloat ALmixer_GetVolumeSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8534 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8535 ALfloat retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8536 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8537 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8538 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8539 retval = Internal_GetVolumeSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8540 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8541 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8542 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8543 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8544 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8545 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8546 ALboolean ALmixer_SetMaxVolumeChannel(ALint channel, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8547 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8548 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8549 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8550 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8551 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8552 retval = Internal_SetMaxVolumeChannel(channel, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8553 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8554 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8555 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8556 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8557 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8558 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8559 ALboolean ALmixer_SetMaxVolumeSource(ALuint source, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8560 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8561 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8562 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8563 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8564 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8565 retval = Internal_SetMaxVolumeSource(source, volume); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8566 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8567 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8568 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8569 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8570 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8571 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8572 ALfloat ALmixer_GetMaxVolumeChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8573 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8574 ALfloat retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8575 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8576 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8577 #endif |
| 0 | 8578 retval = Internal_GetMaxVolumeChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8579 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8580 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8581 #endif |
| 0 | 8582 return retval; |
| 8583 } | |
| 8584 | |
| 8585 ALfloat ALmixer_GetMaxVolumeSource(ALuint source) | |
| 8586 { | |
| 8587 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8588 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8589 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8590 #endif |
| 0 | 8591 retval = Internal_GetMaxVolumeSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8592 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8593 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8594 #endif |
| 0 | 8595 return retval; |
| 8596 } | |
| 8597 | |
| 8598 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8599 ALboolean ALmixer_SetMinVolumeChannel(ALint channel, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8600 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8601 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8602 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8603 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8604 #endif |
| 0 | 8605 retval = Internal_SetMinVolumeChannel(channel, volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8606 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8607 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8608 #endif |
| 0 | 8609 return retval; |
| 8610 } | |
| 8611 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8612 ALboolean ALmixer_SetMinVolumeSource(ALuint source, ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8613 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8614 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8615 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8616 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8617 #endif |
| 0 | 8618 retval = Internal_SetMinVolumeSource(source, volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8619 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8620 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8621 #endif |
| 0 | 8622 return retval; |
| 8623 } | |
| 8624 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8625 ALfloat ALmixer_GetMinVolumeChannel(ALint channel) |
| 0 | 8626 { |
| 8627 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8628 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8629 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8630 #endif |
| 0 | 8631 retval = Internal_GetMinVolumeChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8632 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8633 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8634 #endif |
| 0 | 8635 return retval; |
| 8636 } | |
| 8637 | |
| 8638 ALfloat ALmixer_GetMinVolumeSource(ALuint source) | |
| 8639 { | |
| 8640 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8641 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8642 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8643 #endif |
| 0 | 8644 retval = Internal_GetMinVolumeSource(source); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8645 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8646 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8647 #endif |
| 0 | 8648 return retval; |
| 8649 } | |
| 8650 | |
| 8651 | |
| 8652 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8653 ALboolean ALmixer_SetMasterVolume(ALfloat volume) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8654 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8655 ALboolean retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8656 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8657 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8658 #endif |
| 0 | 8659 retval = Internal_SetMasterVolume(volume); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8660 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8661 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8662 #endif |
| 0 | 8663 return retval; |
| 8664 } | |
| 8665 | |
| 8666 ALfloat ALmixer_GetMasterVolume() | |
| 8667 { | |
| 8668 ALfloat retval; | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8669 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8670 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8671 #endif |
| 0 | 8672 retval = Internal_GetMasterVolume(); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8673 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8674 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8675 #endif |
| 0 | 8676 return retval; |
| 8677 } | |
| 8678 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8679 ALint ALmixer_ExpireChannel(ALint channel, ALint ticks) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8680 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8681 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8682 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8683 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8684 #endif |
| 0 | 8685 retval = Internal_ExpireChannel(channel, ticks); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8686 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8687 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8688 #endif |
| 0 | 8689 return retval; |
| 8690 } | |
| 8691 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8692 ALint ALmixer_ExpireSource(ALuint source, ALint ticks) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8693 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8694 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8695 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8696 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8697 #endif |
| 0 | 8698 retval = Internal_ExpireSource(source, ticks); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8699 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8700 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8701 #endif |
| 0 | 8702 return retval; |
| 8703 } | |
| 8704 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8705 ALint ALmixer_IsActiveChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8706 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8707 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8708 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8709 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8710 #endif |
| 0 | 8711 retval = Internal_QueryChannel(channel); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8712 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8713 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8714 #endif |
| 0 | 8715 return retval; |
| 8716 } | |
| 8717 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8718 ALint ALmixer_IsActiveSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8719 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8720 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8721 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8722 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8723 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8724 retval = Internal_QuerySource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8725 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8726 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8727 #endif |
| 0 | 8728 return retval; |
| 8729 } | |
| 8730 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8731 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8732 ALint ALmixer_IsPlayingChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8733 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8734 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8735 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8736 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8737 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8738 retval = Internal_PlayingChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8739 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8740 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8741 #endif |
| 0 | 8742 return retval; |
| 8743 } | |
| 8744 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8745 ALint ALmixer_IsPlayingSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8746 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8747 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8748 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8749 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8750 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8751 retval = Internal_PlayingSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8752 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8753 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8754 #endif |
| 0 | 8755 return retval; |
| 8756 } | |
| 8757 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8758 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8759 ALint ALmixer_IsPausedChannel(ALint channel) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8760 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8761 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8762 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8763 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8764 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8765 retval = Internal_PausedChannel(channel); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8766 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8767 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8768 #endif |
| 0 | 8769 return retval; |
| 8770 } | |
| 8771 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8772 ALint ALmixer_IsPausedSource(ALuint source) |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8773 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8774 ALint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8775 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8776 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8777 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8778 retval = Internal_PausedSource(source); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8779 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8780 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8781 #endif |
| 0 | 8782 return retval; |
| 8783 } | |
| 8784 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8785 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8786 ALuint ALmixer_CountAllFreeChannels() |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8787 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8788 ALuint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8789 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8790 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8791 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8792 retval = Internal_CountAllFreeChannels(); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8793 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8794 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8795 #endif |
| 0 | 8796 return retval; |
| 8797 } | |
| 8798 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8799 ALuint ALmixer_CountUnreservedFreeChannels() |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8800 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8801 ALuint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8802 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8803 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8804 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8805 retval = Internal_CountUnreservedFreeChannels(); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8806 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8807 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8808 #endif |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8809 return retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8810 } |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8811 |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8812 ALuint ALmixer_CountAllUsedChannels() |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8813 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8814 ALuint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8815 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8816 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8817 #endif |
| 0 | 8818 retval = Internal_CountAllUsedChannels(); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8819 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8820 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8821 #endif |
| 0 | 8822 return retval; |
| 8823 } | |
| 8824 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8825 ALuint ALmixer_CountUnreservedUsedChannels() |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8826 { |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8827 ALuint retval; |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8828 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8829 SDL_LockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8830 #endif |
| 0 | 8831 retval = Internal_CountUnreservedUsedChannels(); |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8832 #ifdef ENABLE_ALMIXER_THREADS |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8833 SDL_UnlockMutex(s_simpleLock); |
|
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8834 #endif |
| 0 | 8835 return retval; |
| 8836 } | |
| 8837 | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8838 ALboolean ALmixer_IsPredecoded(ALmixer_Data* data) |
| 1 | 8839 { |
| 8840 if(NULL == data) | |
| 8841 { | |
|
2
279d0427ef26
Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
changeset
|
8842 return AL_FALSE; |
| 1 | 8843 } |
| 8844 return data->decoded_all; | |
| 8845 } | |
| 8846 | |
|
2
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8847 ALboolean ALmixer_CompiledWithThreadBackend() |
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8848 { |
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parents:
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8849 #ifdef ENABLE_ALMIXER_THREADS |
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8850 return AL_TRUE; |
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8851 #else |
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8852 return AL_FALSE; |
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279d0427ef26
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
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diff
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8853 #endif |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
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diff
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8854 } |
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Eric Wing <ewing . public |-at-| gmail . com>
parents:
1
diff
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|
8855 |
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
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diff
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8856 |
|
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
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8857 |
|
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Overhaul prep for first public release.
Eric Wing <ewing . public |-at-| gmail . com>
parents:
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|
8858 |